From alexei.vasilyev at gmail.com Tue Jan 1 07:22:28 2019 From: alexei.vasilyev at gmail.com (Alexey Vasilyev) Date: Tue, 1 Jan 2019 13:22:28 +0100 Subject: [OpenSIPS-Users] usrloc restart persistency on seed node In-Reply-To: <002c01d4a12a$dcf2b920$96d82b60$@smartvox.co.uk> References: <002c01d4a12a$dcf2b920$96d82b60$@smartvox.co.uk> Message-ID: <6F8E96CF-7BB4-4DA7-B2E4-4C5216F0BF39@gmail.com> Hi John, Next is just my opinion. And I didn’t explore source code OpenSIPS for syncing data. The problem is little bit deeper. As we have cluster, we potentially have split-brain. We can disable seed node at all and just let nodes work after disaster/restart. But it means that we can’t guarantee consistency of data. So nodes must show this with «Not in sync» state. Usually clusters use quorum to trust on. But for OpenSIPS I think this approach is too expensive. And of course for quorum we need minimum 3 hosts. For 2 hosts after loosing/restoring interconnection it is impossible to say, which host has consistent data. That’s why OpenSIPS uses seed node as artificial trust point. I think «seed» node doesn’t solve syncing problems, but it simplifies total work. Let’s imagine 3 nodes A,B,C. A is Active. A and B lost interconnection. C is down. Then C is up and has 2 hosts for syncing. But A already has 200 phones re-registered for some reason. So we have 200 conflicts (on node B the same phones still in memory). Where to sync from? «Seed» host will answer this question in 2 cases (A or B). Of course if C is «seed» so it just will be happy from the start. And I actually don’t know what happens, if we now run «ul_cluster_sync» on C. Will it get all the contacts from A and B or not? We operate with specific data, which is temporary. So syncing policy can be more relaxed. May be it’s a good idea to connect somehow «seed» node with Active role in the cluster. But again, if Active node restarts and still Active - we will have a problem. ----- Alexey Vasilyev > 31 Dec 2018, в 18:04, John Quick написал(а): > > Hi Alexei, > > Many thanks for your reply to my query about syncing the seed node for > usrloc registrations. > I just tried the command you suggested and it does solve the problem. I also > read the other thread you pointed to. > > I do not really understand the need for the seed node, especially not for > the case of memory based registrations. > A seed node makes sense if that node has a superior knowledge of the > topology or the data than the other nodes. It's view of the universe is to > be trusted more than the view held by any other node. > However, in the case of a cluster topology that is pre-defined (no > auto-discovery) and for full-sharing of usrloc registration data held > exclusively in memory, then all the nodes are equal - there is no superior > knowledge that can exist in one node. The one with the most accurate view of > the world is the one that has been running the longest. > > I am wondering if there is a justifiable case for an option that would > disable the concept of the seed node and make it so that, on startup, every > instance will attempt to get the usrloc data from any other running instance > that has data available. In effect, I can mimic this behaviour by adding the > command line you suggested just after opensips has started: > opensipsctl fifo ul_cluster_sync > > Am I missing something here about the concept of the seed node? > It concerns me that this seed concept is at odds with the concept of true > horizontal scalability. > All nodes are equal, but some are more equal than others! > > John Quick > Smartvox Limited > Web: www.smartvox.co.uk > > From spanda at 3clogic.com Wed Jan 2 01:52:31 2019 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 2 Jan 2019 12:22:31 +0530 Subject: [OpenSIPS-Users] I deleted an entry from clusterer table , but still opensips try to ping that node . In-Reply-To: References: Message-ID: Hi Sammy, Yes , you are right . I need to reload the cluster data through MI command . After reloading its seems fine . I was not aware about the fact that the cluster data also get shared with all nodes when I am adding that in 1 node only . Thank you for your explanation . Its really helpful . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Mon, Dec 31, 2018 at 10:25 PM SamyGo wrote: > Hi, > Did you restart OpenSIPS process on both node1, and 2 simultaneously ? The > way I look at this is one of the two nodes kept the 3rd one in the memory > and restarting both nodes one at a time resulted in both sharing their node > structure and hence node3 stayed visible. > I think possible way to remove a node gracefully would be to disable the > node via the MI command and then remove from DB. I will try doing this on > my test setup as well. > > Regards, > Sammy > > > On Fri, Dec 28, 2018 at 6:40 AM Sasmita Panda wrote: > >> Hi All, >> >> I have a cluster of 2 nodes . Both in working condition . Then I added >> another node in the same cluster which is down . >> >> I restarted the opensips process , so it starts pinging the new node to >> check its status . As the new node is down , other nodes in the cluster >> wont get any reply for the ping . Then I remove the 3rd node from the >> cluster table and restart the opensips process . >> >> Now what I am getting in logs is , still the 2 working node in the >> cluster try to ping the 3rd node which is not in the DB . >> >> Is this an issue on the cluster module or I am doing something wrong ?? >> Please help me . >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Jan 2 05:10:22 2019 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 2 Jan 2019 15:40:22 +0530 Subject: [OpenSIPS-Users] I deleted an entry from clusterer table , but still opensips try to ping that node . In-Reply-To: References: Message-ID: Hi, I have another doubt . Please do help me . When I am reading usrloc module document , its saying in a cluster if we want to replicate the contacts across the cluster then we have to set a parameter as below . modparam("usrloc", "replicate_contacts_to", 1) The default value is 0 , where no cluster id is mentioned . I have not set this , I have a cluster having 2 node . While I am registering a user , the contact is getting replicated between 2 nodes . If I am trying to mention this parameter , then opensips is not getting started . Its saying *Parameter not found in module * *So , my question is , if this parameter is not set , still how contact replication is happening ? Is this the default behavior of cluster module ? * *May be my question is foolish ,it will be great if anybody will explain this . * *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Wed, Jan 2, 2019 at 12:22 PM Sasmita Panda wrote: > Hi Sammy, > > Yes , you are right . I need to reload the cluster data through MI command > . After reloading its seems fine . > > I was not aware about the fact that the cluster data also get shared with > all nodes when I am adding that in 1 node only . > > Thank you for your explanation . Its really helpful . > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Mon, Dec 31, 2018 at 10:25 PM SamyGo wrote: > >> Hi, >> Did you restart OpenSIPS process on both node1, and 2 simultaneously ? >> The way I look at this is one of the two nodes kept the 3rd one in the >> memory and restarting both nodes one at a time resulted in both sharing >> their node structure and hence node3 stayed visible. >> I think possible way to remove a node gracefully would be to disable the >> node via the MI command and then remove from DB. I will try doing this on >> my test setup as well. >> >> Regards, >> Sammy >> >> >> On Fri, Dec 28, 2018 at 6:40 AM Sasmita Panda wrote: >> >>> Hi All, >>> >>> I have a cluster of 2 nodes . Both in working condition . Then I >>> added another node in the same cluster which is down . >>> >>> I restarted the opensips process , so it starts pinging the new node to >>> check its status . As the new node is down , other nodes in the cluster >>> wont get any reply for the ping . Then I remove the 3rd node from the >>> cluster table and restart the opensips process . >>> >>> Now what I am getting in logs is , still the 2 working node in the >>> cluster try to ping the 3rd node which is not in the DB . >>> >>> Is this an issue on the cluster module or I am doing something wrong ?? >>> Please help me . >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Senior Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Wed Jan 2 05:24:41 2019 From: john.quick at smartvox.co.uk (John Quick) Date: Wed, 2 Jan 2019 10:24:41 -0000 Subject: [OpenSIPS-Users] usrloc restart persistency on seed node Message-ID: <000d01d4a285$5f9fb840$1edf28c0$@smartvox.co.uk> Alexey, Thanks for your feedback. I acknowledge that, in theory, a situation may arise where a node is brought online and all the previously running nodes were not fully synchronised so it is then a problem for the newly started node to know which data set to pull. In addition to the example you give - lost interconnection - I can also foresee difficulties when several nodes all start at the same time. However, I do not see how arbitrarily setting one node as "seed" will help to resolve either of these situations unless the seed node has more (or better) information than the others. I am trying to design a multi-node solution that is scalable. I want to be able to add and remove nodes according to current load. Also, to be able to take one node offline, do some maintenance, then bring it back online. For my scenario, the probability of any node being taken offline for maintenance during the year is 99.9% whereas I would say the probability of partial loss of LAN connectivity (causing the split-brain issue) is less than 0.01%. If possible, I would really like to see an option added to the usrloc module to override the "seed" node concept. Something that allows any node (including seed) to attempt to pull registration details from another node on startup. In my scenario, a newly started node with no usrloc data is a major problem - it could take it 40 minutes to get close to having a full set of registration data. I would prefer to take the risk of it pulling data from the wrong node rather than it not attempting to synchronise at all. Happy New Year to all. John Quick Smartvox Limited > Hi John, > > Next is just my opinion. And I didn't explore source code OpenSIPS for syncing data. > > The problem is little bit deeper. As we have cluster, we potentially have split-brain. > We can disable seed node at all and just let nodes work after disaster/restart. But it means that we can't guarantee consistency of data. So nodes must show this with state. > > Usually clusters use quorum to trust on. But for OpenSIPS I think this approach is too expensive. And of course for quorum we need minimum 3 hosts. > For 2 hosts after loosing/restoring interconnection it is impossible to say, which host has consistent data. That's why OpenSIPS uses seed node as artificial trust point. I think node doesn't solve syncing problems, but it simplifies total work. > > Let's imagine 3 nodes A,B,C. A is Active. A and B lost interconnection. C is down. Then C is up and has 2 hosts for syncing. But A already has 200 phones re-registered for some reason. So we have 200 conflicts (on node B the same phones still in memory). Where to sync from? host will answer this question in 2 cases (A or B). Of course if C is so it just will be happy from the start. And I actually don't know what happens, if we now run on C. Will it get all the contacts from A and B or not? > >We operate with specific data, which is temporary. So syncing policy can be more relaxed. May be it's a good idea to connect somehow node with Active role in the cluster. But again, if Active node restarts and still Active - we will have a problem. > > ----- > Alexey Vasilyev From vikash.tibrewal at aricent.com Wed Jan 2 07:41:55 2019 From: vikash.tibrewal at aricent.com (Vikash Tibrewal) Date: Wed, 2 Jan 2019 12:41:55 +0000 Subject: [OpenSIPS-Users] Integrate OpenSIPS with VG202 Audio Gateway Message-ID: Hello All, I am new to OpenSIPS and I want to integrate the OpenSIPS with VG202 audio gateway and connect that with hardphone. I have already installed OpenSIPS v 2.4 on Ubuntu 16.04 and its running fine and I have installed the OpenSIPS control panel and created the users in that. However, I am not getting any clue how to integrate OpenSIPS with Hardphone and make the calls? Can you please suggest what are the options available for integration? Regards, Vikash Tibrewal ===================================================== Please refer to http://www.aricent.com/email-disclaimer for important disclosures regarding this electronic communication. ===================================================== -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jan 3 05:13:02 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 12:13:02 +0200 Subject: [OpenSIPS-Users] DNS Cache module In-Reply-To: <1F2616F0-039C-4E50-B1B0-7AEED643C044@genesys.com> References: <667279fc-8dab-7cb7-7301-e10416234aea@opensips.org> <1F2616F0-039C-4E50-B1B0-7AEED643C044@genesys.com> Message-ID: <9ff60eb4-8730-7fcb-c596-abe74773f526@opensips.org> Hi Ben, What exact version / revision of OpenSIPS do you use ? maybe I can help adding some extra debug logs. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/07/2018 08:14 PM, Ben Newlin wrote: > > Bogdan, > > That had occurred to me as well. I have verified testing locally that > when successful the value is printed in the log. I have also verified > manually on the system with the error that the key does exist in the > cache, but the value in the cache is empty, just as in the log. That > is what I believe is causing the failure, the lookup from the cache is > successful (the key exists) but the value is empty, so OpenSIPS cannot > route. > > I have attempted increasing the logging on the system, but it appears > the dns_cache module does not log anything further of use. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Friday, December 7, 2018 at 12:21 PM > *To: *OpenSIPS users mailling list , Ben > Newlin > *Subject: *Re: [OpenSIPS-Users] DNS Cache module > > Hi Ben, > > IMO, the log itself is broken as the data to be cached is not > printable ....so the logs you see may be misleading. > > When you say "OpenSIPS appeared to not be able to resolve the domain", > you mean OpenSIPS is not doing any attempt to solve the FQDN, or you > mean OpenSIPS is loading from cash something wrong ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 2019 > https://www.opensips.org/events/Summit-2019Amsterdam/ > > On 12/06/2018 10:17 PM, Ben Newlin wrote: > > Hello, > > We use the DNS cache module to reduce the time spent querying DNS > records. We recently had a customer call failing and we traced the > failure to the customer using an FQDN in the Record-Route header. > On the ACK, OpenSIPS appeared to not be able to resolve the domain > even though it had been successfully resolved on the initial > request. I found the log for the DNS Cache module and noticed that > the value it was inserting was empty: > > INFO:dns_cache:put_dnscache_value: putting key > [dnscache_customer.domain.com_a] with value [] ttl = 60 > > This prompted me to examine all of our logs and I found that the > value for these DNS Cache logs is always empty, regardless of the > domain. It appears the records are not being serialized properly > into the cache. > > The DNS resolution must be succeeding or all of our requests using > DNS would be failing, but I have also verified the domains all can > be resolved manually on the same box: > > $ nslookup customer.domain.com > > Server: 10.27.0.2 > > Address: 10.27.0.2#53 > > Non-authoritative answer: > > Name: customer.domain.com > > Address: 10.27.172.132 > > Name: customer.domain.com > > Address: 10.27.192.211 > > Name: customer.domain.com > > Address: 10.27.255.53 > > Any thoughts? Is there more information I can obtain to determine > the cause? > > Ben Newlin > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Jan 3 05:33:50 2019 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 3 Jan 2019 12:33:50 +0200 Subject: [OpenSIPS-Users] usrloc restart persistency on seed node In-Reply-To: <000d01d4a285$5f9fb840$1edf28c0$@smartvox.co.uk> References: <000d01d4a285$5f9fb840$1edf28c0$@smartvox.co.uk> Message-ID: <10c4d65d-0052-cd7b-e9a9-e8f307568321@opensips.org> Happy New Year John, Alexey and everyone else! I just finished catching up with this thread, and I must admit that I now concur with John's distaste of the asymmetric nature of cluster node restarts! Although it is correct and gets the job done, the 2.4 "seed" mechanism forces the admin to conditionally add a "opensipsctl fifo ul_cluster_sync" command into the startup script of all "seed" nodes.  I think we can do better :) What if we kept the "seed" concept, but tweaked it such that instead of meaning: "following a restart, always start in 'synced' state, with an empty dataset" ... it would now mean: "following a restart or cluster sync command, fall back to a 'synced' state, with an empty dataset if and only if we are unable to find a suitable sync candidate within X seconds" This solution seems to fit all requirements that I've seen posted so far.  It is: * correct (a cluster with at least 1 "seed" node will still never deadlock) * symmetric (with the exception of cluster bootstrapping, all node restarts are identical) * autonomous (users need not even know about "ul_cluster_sync" anymore!  Not saying               this is necessarily good, but it brings down the learning curve) The only downside could be that any cluster bootstrap will now last at least X seconds. But that seems such a rare event (in production, at least) that we need not worry about it.  Furthermore, the X seconds will be configurable. What do you think? PS: by "cluster bootstrap" I mean (re)starting all nodes simultaneously. Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 02.01.2019 12:24, John Quick wrote: > Alexey, > > Thanks for your feedback. I acknowledge that, in theory, a situation may > arise where a node is brought online and all the previously running nodes > were not fully synchronised so it is then a problem for the newly started > node to know which data set to pull. In addition to the example you give - > lost interconnection - I can also foresee difficulties when several nodes > all start at the same time. However, I do not see how arbitrarily setting > one node as "seed" will help to resolve either of these situations unless > the seed node has more (or better) information than the others. > > I am trying to design a multi-node solution that is scalable. I want to be > able to add and remove nodes according to current load. Also, to be able to > take one node offline, do some maintenance, then bring it back online. For > my scenario, the probability of any node being taken offline for maintenance > during the year is 99.9% whereas I would say the probability of partial loss > of LAN connectivity (causing the split-brain issue) is less than 0.01%. > > If possible, I would really like to see an option added to the usrloc module > to override the "seed" node concept. Something that allows any node > (including seed) to attempt to pull registration details from another node > on startup. In my scenario, a newly started node with no usrloc data is a > major problem - it could take it 40 minutes to get close to having a full > set of registration data. I would prefer to take the risk of it pulling data > from the wrong node rather than it not attempting to synchronise at all. > > Happy New Year to all. > > John Quick > Smartvox Limited > > >> Hi John, >> >> Next is just my opinion. And I didn't explore source code OpenSIPS for > syncing data. >> The problem is little bit deeper. As we have cluster, we potentially have > split-brain. >> We can disable seed node at all and just let nodes work after > disaster/restart. But it means that we can't guarantee consistency of data. > So nodes must show this with state. >> Usually clusters use quorum to trust on. But for OpenSIPS I think this > approach is too expensive. And of course for quorum we need minimum 3 hosts. >> For 2 hosts after loosing/restoring interconnection it is impossible to > say, which host has consistent data. That's why OpenSIPS uses seed node as > artificial trust point. I think node doesn't solve syncing problems, > but it simplifies total work. >> Let's imagine 3 nodes A,B,C. A is Active. A and B lost interconnection. C > is down. Then C is up and has 2 hosts for syncing. But A already has 200 > phones re-registered for some reason. So we have 200 conflicts (on node B > the same phones still in memory). Where to sync from? host will > answer this question in 2 cases (A or B). Of course if C is so it > just will be happy from the start. And I actually don't know what happens, > if we now run on C. Will it get all the contacts from A > and B or not? >> We operate with specific data, which is temporary. So syncing policy can be > more relaxed. May be it's a good idea to connect somehow node with > Active role in the cluster. But again, if Active node restarts and still > Active - we will have a problem. >> ----- >> Alexey Vasilyev From john.quick at smartvox.co.uk Thu Jan 3 06:00:03 2019 From: john.quick at smartvox.co.uk (John Quick) Date: Thu, 3 Jan 2019 11:00:03 -0000 Subject: [OpenSIPS-Users] usrloc restart persistency on seed node Message-ID: <000201d4a353$7ac45fd0$704d1f70$@smartvox.co.uk> Hi Liviu, I like your suggestion. It seems like a pragmatic solution so I welcome this idea. The X second delay is probably unavoidable, but could there be a problem if new registration requests arrive during the delay period? I already have an X second delay because my current work-around is to launch a background script just before starting OpenSIPS. The background script has an X second delay then it runs "opensipsctl fifo ul_cluster_sync" and then exits. For backward compatibility, perhaps the default behaviour should be the same as it is now. John Quick Smartvox Limited > Happy New Year John, Alexey and everyone else! > > I just finished catching up with this thread, and I must admit that I now > concur with John's distaste of the asymmetric nature of cluster node restarts! > > Although it is correct and gets the job done, the 2.4 "seed" mechanism forces > the admin to conditionally add a "opensipsctl fifo ul_cluster_sync" command > into the startup script of all "seed" nodes. I think we can do better :) > > What if we kept the "seed" concept, but tweaked it such that instead of > meaning: > "following a restart, always start in 'synced' state, with an empty dataset" > > ... it would now mean: > "following a restart or cluster sync command, fall back to a 'synced' state, > with an empty dataset if and only if we are unable to find a suitable sync > candidate within X seconds" > > This solution seems to fit all requirements that I've seen posted so far. It is: > > * correct (a cluster with at least 1 "seed" node will still never deadlock) > * symmetric (with the exception of cluster bootstrapping, all node restarts are identical) > * autonomous (users need not even know about "ul_cluster_sync" anymore! > Not saying this is necessarily good, but it brings down the learning curve) > > The only downside could be that any cluster bootstrap will now last at > least X seconds. > But that seems such a rare event (in production, at least) that we need > not worry about it. Furthermore, the X seconds will be configurable. > > What do you think? > > PS: by "cluster bootstrap" I mean (re)starting all nodes simultaneously. > > Best regards, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com From alexei.vasilyev at gmail.com Thu Jan 3 06:23:32 2019 From: alexei.vasilyev at gmail.com (Alexey Vasilyev) Date: Thu, 3 Jan 2019 12:23:32 +0100 Subject: [OpenSIPS-Users] usrloc restart persistency on seed node In-Reply-To: <10c4d65d-0052-cd7b-e9a9-e8f307568321@opensips.org> References: <000d01d4a285$5f9fb840$1edf28c0$@smartvox.co.uk> <10c4d65d-0052-cd7b-e9a9-e8f307568321@opensips.org> Message-ID: Hi everybody, I like the approach, but here are some thoughts. I think that X seconds delay should not pause all the opensips work. Just to run asynchronously, allowing to process requests even before syncing data. For example, I use for syncyng from systemd "ExecStartPost" script. So it runs, when opensips already started. (And, by the way, John, be careful, don't run "ul_cluster_sync" when you are starting "seed" node first, without running any another node. It makes cluster "Not synced') Lets imagine, "seed" node starts and find 2 nodes (or more), which one to choose for syncing? And if they have different data (they were not synced between each other), what should it do? Thanks. чт, 3 янв. 2019 г. в 11:33, Liviu Chircu : > Happy New Year John, Alexey and everyone else! > > I just finished catching up with this thread, and I must admit that I now > concur with John's distaste of the asymmetric nature of cluster node > restarts! > > Although it is correct and gets the job done, the 2.4 "seed" mechanism > forces > the admin to conditionally add a "opensipsctl fifo ul_cluster_sync" command > into the startup script of all "seed" nodes. I think we can do better :) > > What if we kept the "seed" concept, but tweaked it such that instead of > meaning: > > "following a restart, always start in 'synced' state, with an empty > dataset" > > ... it would now mean: > > "following a restart or cluster sync command, fall back to a 'synced' > state, > with an empty dataset if and only if we are unable to find a suitable sync > candidate within X seconds" > > This solution seems to fit all requirements that I've seen posted so > far. It is: > > * correct (a cluster with at least 1 "seed" node will still never deadlock) > * symmetric (with the exception of cluster bootstrapping, all node > restarts are identical) > * autonomous (users need not even know about "ul_cluster_sync" anymore! > Not saying > this is necessarily good, but it brings down the learning > curve) > > The only downside could be that any cluster bootstrap will now last at > least X seconds. > But that seems such a rare event (in production, at least) that we need > not worry > about it. Furthermore, the X seconds will be configurable. > > What do you think? > > PS: by "cluster bootstrap" I mean (re)starting all nodes simultaneously. > > Best regards, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 02.01.2019 12:24, John Quick wrote: > > Alexey, > > > > Thanks for your feedback. I acknowledge that, in theory, a situation may > > arise where a node is brought online and all the previously running nodes > > were not fully synchronised so it is then a problem for the newly started > > node to know which data set to pull. In addition to the example you give > - > > lost interconnection - I can also foresee difficulties when several nodes > > all start at the same time. However, I do not see how arbitrarily setting > > one node as "seed" will help to resolve either of these situations unless > > the seed node has more (or better) information than the others. > > > > I am trying to design a multi-node solution that is scalable. I want to > be > > able to add and remove nodes according to current load. Also, to be able > to > > take one node offline, do some maintenance, then bring it back online. > For > > my scenario, the probability of any node being taken offline for > maintenance > > during the year is 99.9% whereas I would say the probability of partial > loss > > of LAN connectivity (causing the split-brain issue) is less than 0.01%. > > > > If possible, I would really like to see an option added to the usrloc > module > > to override the "seed" node concept. Something that allows any node > > (including seed) to attempt to pull registration details from another > node > > on startup. In my scenario, a newly started node with no usrloc data is a > > major problem - it could take it 40 minutes to get close to having a full > > set of registration data. I would prefer to take the risk of it pulling > data > > from the wrong node rather than it not attempting to synchronise at all. > > > > Happy New Year to all. > > > > John Quick > > Smartvox Limited > > > > > >> Hi John, > >> > >> Next is just my opinion. And I didn't explore source code OpenSIPS for > > syncing data. > >> The problem is little bit deeper. As we have cluster, we potentially > have > > split-brain. > >> We can disable seed node at all and just let nodes work after > > disaster/restart. But it means that we can't guarantee consistency of > data. > > So nodes must show this with state. > >> Usually clusters use quorum to trust on. But for OpenSIPS I think this > > approach is too expensive. And of course for quorum we need minimum 3 > hosts. > >> For 2 hosts after loosing/restoring interconnection it is impossible to > > say, which host has consistent data. That's why OpenSIPS uses seed node > as > > artificial trust point. I think node doesn't solve syncing > problems, > > but it simplifies total work. > >> Let's imagine 3 nodes A,B,C. A is Active. A and B lost interconnection. > C > > is down. Then C is up and has 2 hosts for syncing. But A already has 200 > > phones re-registered for some reason. So we have 200 conflicts (on node B > > the same phones still in memory). Where to sync from? host will > > answer this question in 2 cases (A or B). Of course if C is so it > > just will be happy from the start. And I actually don't know what > happens, > > if we now run on C. Will it get all the contacts from A > > and B or not? > >> We operate with specific data, which is temporary. So syncing policy > can be > > more relaxed. May be it's a good idea to connect somehow node with > > Active role in the cluster. But again, if Active node restarts and still > > Active - we will have a problem. > >> ----- > >> Alexey Vasilyev > -- Best regards Alexey Vasilyev -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jan 3 06:30:27 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 13:30:27 +0200 Subject: [OpenSIPS-Users] presence cluster In-Reply-To: <1463075272.3944.1545153039518.JavaMail.zimbra@skillsearch.ca> References: <1463075272.3944.1545153039518.JavaMail.zimbra@skillsearch.ca> Message-ID: Hi Volga, as it looks, the 'presence-pclean' timer task got stuck and no more triggering are possible. Why? maybe a postgres query issue, taking too long? Could you share the trap file ? Regards and A Happy New Year, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/18/2018 07:10 PM, Slava Bendersky wrote: > Hello Everyone, > Presence cluster getting execution delay messages > > Dec 18 11:03:01 aitossbc03 /usr/sbin/opensips[28251]: > WARNING:core:timer_ticker: timer task already > scheduled 99750 ms ago (now 1500130 ms), skipping execution > Dec 18 11:03:01 aitossbc03 /usr/sbin/opensips[28251]: > WARNING:core:timer_ticker: timer task already > scheduled 299330 ms ago (now 1500130 ms), delaying execution > Dec 18 11:03:02 aitossbc03 /usr/sbin/opensips[28251]: > WARNING:core:timer_ticker: timer task already > scheduled 300330 ms ago (now 1501130 ms), delaying execution > Dec 18 11:03:03 aitossbc03 /usr/sbin/opensips[28251]: > WARNING:core:timer_ticker: timer task already > scheduled 301330 ms ago (now 1502130 ms), delaying execution > Dec 18 11:03:04 aitossbc03 /usr/sbin/opensips[28251]: > WARNING:core:timer_ticker: timer task already > scheduled 302330 ms ago (now 1503130 ms), delaying execution > Dec 18 11:03:05 aitossbc03 /usr/sbin/opensips[28251]: > WARNING:core:timer_ticker: timer task already > scheduled 303330 ms ago (now 1504130 ms), delaying execution > Dec 18 11:03:06 aitossbc03 /usr/sbin/opensips[28251]: > WARNING:core:timer_ticker: timer task already > scheduled 304330 ms ago (now 1505130 ms), delaying execution > > > Current model is cluster of 3 opensips nodes with shared DB (PgSQL > BDR). Database connection is distributed by haproxy in front. Only > presence module generate messages. I checked configuration and switch > from dns to just ip, but no difference. I got trap file might be > helpful to understand. > > Any help thank you. > > volga629 > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jan 3 06:33:19 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 13:33:19 +0200 Subject: [OpenSIPS-Users] dispatcher, drouting, carrierroute and authorization In-Reply-To: References: Message-ID: <1f112d39-84b7-3029-fe22-5f1b337df797@opensips.org> Hi Michael, You can use any of those three, but in combination with the uac and uac_auth module (which provide user client authentication against another server). Use your preferred module (like drouting) to send the calls to the GW and arm a failure route. If you receive an 407 auth request, use the uac_auth() function (from uac module) to generate the auth response and do serial forking to the same GW. Regards and A Happy New Year, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/21/2018 03:54 AM, Michael Vale wrote: > > Hi, > > Which of the three, dispatcher, drouting or carrierroute should I use > if my gateway requires authorization and if it does, how do I > authorize? Do I use uac or registrar and if so how do I do so? Could > someone please give me an example? If I was to use drouting how do I > store the proxy auth credentials in a drouting entry’s attributes? > > Regards, > > Michael > > Sent from Mail for > Windows 10 > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jan 3 06:36:14 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 13:36:14 +0200 Subject: [OpenSIPS-Users] using LDAP attribute value's for aliases In-Reply-To: References: Message-ID: <3d63a161-75a8-b9ab-092e-73f44a1bc476@opensips.org> Hi Michael, The LDAP[1] module gives you the possibility to perform generic LDAP queries against an external LDAP server - so you can retrieve whatever fields form it and use them as you like (as aliases, as passwords, etc) [1] https://opensips.org/html/docs/modules/2.4.x/ldap.html Regards and A Happy New Year, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/21/2018 06:59 AM, Michael Vale wrote: > > Hi, > > How can I use an LDAP entry for aliases? Is it possible without > editing code? > > I can store the value as an AVP, can I then create a usrloc entry that > points to the user in hand instead of a dbalias? > > Sent from Mail for > Windows 10 > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jan 3 06:55:35 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 13:55:35 +0200 Subject: [OpenSIPS-Users] segfault on 2.4.4 In-Reply-To: References: Message-ID: <3a8c64e7-798c-9f1e-4659-114ee46e1e67@opensips.org> Hi Jennifer, Do you still have the core file for investigation with GDB ? Also, are you 100% sure your cluster is consistent when comes to the ACC settings, like *all* the nodes do have exactly the same "extra" and "legs" sets ? Best regards and A Happy New Year, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/21/2018 08:48 PM, Jennifer Hashimoto wrote: > Hi guys, I just updated and i’m getting segfault not sure why, maybe > to do with dialog replication? > > Here are the details, let me know if you could use more > information.https://opensips.org/html/docs/modules/2.4.x/ldap.html > > Thanks, > Jen > > > > version: opensips 2.4.4 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll, sigio_rt, select. > main.c compiled on with gcc 4.9.2 > > Dec 21 13:42:55 NFO:clusterer:handle_internal_msg: Node [2] is UP > Dec 21 13:43:11 13CB78-2323 64.86.243.114 BYE from caz5400 > sip:8198213679 at 64.86.243.116 -> sip:33232323434 at 209.58.46.142:5060 > bye_from=caz5400 64.86.243.116 > Dec 21 13:43:31 NFO:load_balancer:set_dst_state_from_rplcode: disable > destination 2 after 408 reply on probe > Dec 21 13:43:49 19C280-2672 64.86.243.114 BYE from caz5400 > sip:4036409658 at 64.86.243.116 -> sip:36205873156 at 209.58.46.142:5060 > bye_from=caz5400 64.86.243.116 > Dec 21 13:44:02 RITICAL:core:sig_usr: segfault in process pid: 3050, > id: 61 > Dec 21 13:44:03 RITICAL:core:handle_worker: dead child 61 (EOF > received), pid 3050 > Dec 21 13:44:03 RITICAL:core:handle_tcp_worker: dead tcp worker 0 (EOF > received), pid 3050 > Dec 21 13:44:05 NFO:core:handle_sigs: child process 3050 exited by a > signal 11 > Dec 21 13:44:05 NFO:core:handle_sigs: core was generated > Dec 21 13:44:05 NFO:core:handle_sigs: terminating due to SIGCHLD > Dec 21 13:44:05 NFO:core:sig_usr: signal 15 received > Dec 21 13:44:05 NFO:core:sig_usr: signal 15 received > Dec 21 13:44:05 NFO:core:sig_usr: signal 15 received > > sudo gdb /usr/sbin/opensips core > GNU gdb (Debian 7.7.1+dfsg-5) 7.7.1 > Copyright (C) 2014 Free Software Foundation, Inc. > License GPLv3+: GNU GPL version 3 or later > > This is free software: you are free to change and redistribute it. > There is NO WARRANTY, to the extent permitted by law. Type "show copying" > and "show warranty" for details. > This GDB was configured as "x86_64-linux-gnu". > Type "show configuration" for configuration details. > For bug reporting instructions, please see: > . > Find the GDB manual and other documentation resources online at: > . > For help, type "help". > Type "apropos word" to search for commands related to "word"... > Reading symbols from /usr/sbin/opensips...Reading symbols from > /usr/lib/debug/.build-id/4c/1b7823a23c3dbd2b5d0ee6392836a093740d2a.debug...done. > done. > [New LWP 3050] > [Thread debugging using libthread_db enabled] > Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". > Core was generated by `/usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg'. > Program terminated with signal SIGSEGV, Segmentation fault. > #0 0x00000000004bca7a in fm_free (qm=0x7f42afdc9000, > p=p at entry=0x3a6874676e654c2d) at mem/f_malloc.c:495 > 495mem/f_malloc.c: No such file or directory. > (gdb) bt full > #0 0x00000000004bca7a in fm_free (qm=0x7f42afdc9000, > p=p at entry=0x3a6874676e654c2d) at mem/f_malloc.c:495 > f = 0x3a6874676e654c15 > n = > #1 0x00007f42aad4bb8a in shm_free (_p=0x3a6874676e654c2d) > at ../../evi/../mem/shm_mem.h:588 > No locals. > #2 set_value_shm (pvt=pvt at entry=0x7ffe501b6e90, extra=0x7f42b2a94bd0) > at acc_vars.c:143 > s = > __FUNCTION__ = "set_value_shm" > #3 0x00007f42aad3a85b in restore_extra_from_str ( > tags_len=, extra_s=extra_s at entry=0x7ffe501b6f10, > extra_len=) at acc.c:1618 > i = > value = {rs = {s = 0x7f42bfe53f3f "oreB", len = 0}, > ri = -1297517200, flags = 1} > values = 0x7f42b2a94a68 > __FUNCTION__ = "restore_extra_from_str" > #4 0x00007f42aad42691 in restore_extra ( > type_str=0x7f42aaf560e0 , ctx=0x7f42b2a87ad0, > ---Type to continue, or q to quit--- > dlg=0x7f42b011c258) at acc.c:1653 > extra_len = > buffer = {s = 0x7f42bfe53f3d "", len = 2} > #5 restore_dlg_extra (dlg=0x7f42b011c258, ctx_p=0x7ffe501b6f80) > at acc.c:1729 > ctx = 0x7f42b2a87ad0 > __FUNCTION__ = "restore_dlg_extra" > #6 0x00007f42aad458a3 in acc_loaded_callback (dlg=0x7f42b011c258, > type=1852132397, _params=0x1) at acc_logic.c:662 > flags_s = {s = 0x7ffe501b6f90 "\003", len = 8} > ctx_s = {s = 0x7ffe501b6fd8 "\b", len = 0} > table_s = {s = 0x7ffe501b6ff0 "|\362\345\277B\177", > len = 1343975760} > created_s = {s = 0x7ffe501b6f88 "b4\035\\", len = 8} > ctx = 0x0 > created = 1545417826 > flags = 9570149209145347 > __FUNCTION__ = "acc_loaded_callback" > #7 0x00007f42aaadadcd in run_load_callback_per_dlg ( > dlg=0x7f42afdc9000) at dlg_cb.c:212 > cb = 0x7f42b01fab30 > ---Type to continue, or q to quit--- > #8 0x00007f42aab16d27 in dlg_replicated_create ( > packet=0x7f42afdc9000, cell=0x7f42b011c258, ftag=0x0, > ttag=0x7ffe501b7110, safe=-265519414) at dlg_replication.c:271 > dir = 2863757965 > dst_leg = 32578 > callid = { > s = 0x7f42bfe5eef9 > "2EB67E0D-48711E9-8927CE48-FE93CD99 at 10.10.20.22 > \f", len = 46} > from_uri = { > s = 0x7f42bfe5ef46 "sip:6138245700 at 10.10.20.22\036 > ", > len = 26} > to_uri = { > s = 0x7f42bfe5ef62 > "sip:87441189781400 at 10.10.20.42f\235j\033r4\035\\\003 > ", > len = 30} > from_tag = {s = 0x7f42bfe5ef29 "B5B6DC0-2171\r", len = 12} > to_tag = {s = 0x7f42bfe5ef37 "KjvKKDN565gee\032", len = 13} > cseq1 = {s = 0x7f42bfe5efba "0\003", len = 1} > cseq2 = {s = 0x7f42bfe5efbd "101", len = 3} > contact1 = { > s = 0x7f42bfe5efeb "sip:6138245700 at 10.10.20.22:5060/", > len = 31} > ---Type to continue, or q to quit--- > contact2 = { > s = 0x7f42bfe5f00c > "sip:441189781400 at 10.10.20.28:5070;transport=udp", len = 47} > rroute1 = {s = 0x0, len = 0} > rroute2 = { > s = 0x7f42bfe5efc4 > "\037", len = 37} > mangled_fu = {s = 0x0, len = 0} > mangled_tu = { > s = 0x7f42bfe5f03f > "sip:441189781400 at 10.10.20.42\t\002accX_table#acc_caztel|accX_created#b4\035\\ > ", > len = 28} > sock = {s = 0x7f42bfe5efa4 "udp:10.10.20.42:5060\001", > len = 20} > vars = { > s = 0x7f42bfe5f05d > "accX_table#acc_caztel|accX_created#b4\035\\", len = 521} > profiles = { > s = 0x7f42bfe5f268 "ani#6138245700|dnis#441189781400|", > len = 33} > dlg = 0x10d30f2bf > callee_sock = 0x0 > ---Type to continue, or q to quit--- > d_entry = 0x7f42b00f0948 > __FUNCTION__ = "dlg_replicated_create" > #9 0x00007f42aab1a76e in receive_dlg_repl (packet=0x7ffe501b7350) > at dlg_replication.c:802 > rc = 0 > __FUNCTION__ = "receive_dlg_repl" > #10 0x00007f42aa8ac8b9 in bin_rcv_mod_packets (packet=0x7f42afdc9000, > packet_type=1852132397, ri=0x1, ptr=0x7f42bfe53f3f) > at clusterer.c:1972 > cl_cap = 0x1 > source_id = 2 > dest_id = 1 > cluster_id = 1 > ev_actions_required = 0 > __FUNCTION__ = "bin_rcv_mod_packets" > #11 0x0000000000429197 in call_callbacks ( > buffer=0x7ffe501b7350 "\330\356\345\277B\177", rcv=0x7f42b10e76a0) > at bin_interface.c:446 > p = 0x7f42bfe4ca68 > pkg_len = 3219442280 > packet = {buffer = {s = 0x7f42bfe5eed8 "P4CK\325\003", > ---Type to continue, or q to quit--- > len = 969}, front_pointer = 0x7f42bfe5f2a1 "\001", > size = 1031, type = 1, next = 0x0, src_id = 2} > __FUNCTION__ = "call_callbacks" > #12 0x00007f42a8e72a6c in bin_handle_req ( > _max_msg_chunks=, con=0x7f42b10e7680, > req=0x7f42a9076280 ) at proto_bin.c:672 > size = > #13 bin_read_req (con=0x7f42b10e7680, bytes_read=0x7ffe501b7450) > at proto_bin.c:827 > bytes = > total_bytes = 981 > req = 0x7f42a9076280 > __FUNCTION__ = "bin_read_req" > #14 0x000000000051e5e7 in handle_io (fm=0x7f42bfe8f318, idx=0, > event_type=-1324452224) at net/net_tcp_proc.c:241 > ret = 0 > n = -1324452224 > s = 7 > resp = 0 > response = {139924415084160, 1} > __FUNCTION__ = "handle_io" > ---Type to continue, or q to quit--- > #15 0x000000000052085f in io_wait_loop_epoll (h=, > t=, repeat=) > at net/../io_wait_loop.h:280 > ep_event = {events = 5640552, data = {ptr = 0x0, fd = 0, > u32 = 0, u64 = 0}} > r = 0 > i = 0 > #16 tcp_worker_proc_loop () at net/net_tcp_proc.c:386 > __FUNCTION__ = "tcp_worker_proc_loop" > #17 0x000000000052911d in tcp_start_processes ( > chd_rank=chd_rank at entry=0x7c9ea8 , > startup_done=startup_done at entry=0x0) at net/net_tcp.c:1892 > r = 0 > reader_fd = {230, 231} > pid = > __FUNCTION__ = "tcp_start_processes" > #18 0x000000000041c679 in main_loop () at main.c:788 > startup_done = 0x0 > chd_rank = 57 > rc = > #19 main (argc=, argv=) at main.c:1439 > ---Type to continue, or q to quit--- > cfg_stream = > c = > r = > tmp = 0x7ffe501b8ec3 "" > tmp_len = > port = > proto = > protos_no = > options = 0x55efc0 "f:cCm:M:b:l:n:N:rRvdDFEVhw:t:u:g:P:G:W:o:" > ret = -1 > seed = 3877236842 > __FUNCTION__ = "main" > (gdb) > --------------------------------------------------- > Jennifer Akemi Hashimoto > Caztel Communications > jennifer.hashimoto at caztel.com > 905-836-5445 > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An 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URL: From bogdan at opensips.org Thu Jan 3 06:58:24 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 13:58:24 +0200 Subject: [OpenSIPS-Users] handle_publish wrong destination In-Reply-To: References: Message-ID: Hi Schneur, For the incoming calls, the PUBLISH is sent to the internal IP address or the NOTIFYes ?? Regards and A Happy New Year, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/23/2018 08:37 AM, Schneur Rosenberg wrote: > HI, my presence was working properly and now I realized that its not > working anymore for incoming calls, I dont have a external presence > server, I use the pua_dialoginfo, I did a trace and I saw that for > outgoing calls the system sends a PUBLISH to itself and then properly > sends a NOTIFY to the subscribers, but on incoming calls the PUBLISH > is sent to the internal IP address of the subscribers and of course > does not get anywhere, what am I doing wrong? > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Thu Jan 3 07:08:59 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 14:08:59 +0200 Subject: [OpenSIPS-Users] Userblacklist Module Issues In-Reply-To: References: Message-ID: Hi Jonathan, 1) I guess you use the check_blacklist() script function with the wrong parameter. Indeed, the DB table is created as "globalblacklist", so you need to be careful and pass this table name via the script functions (and not the faulty "global_blacklist") 2) check your "version" table (in the opensips DB) - you must have there a record containing like this: https://github.com/OpenSIPS/opensips/blob/master/scripts/mysql/userblacklist-create.sql#L1 Regards and A Happy New Year, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/27/2018 03:40 PM, Jonathan Mabrito wrote: > I am working on implementing the userblacklist Module and having a few > issues with it. I am running OpenSIPS 2.4.4 on Ubuntu 16.04 (using APT > to install it). > > 1.) In regards to the Global Blacklist, the script creates a table > called "globalblacklist" and it looks like the code is looking for a > table called "global_blacklist" instead. I found the > userblacklist-create.sql script and created the global_blacklist table > and all is good. What is the proper DB table name? Was the script not > updated or is the code wrong? > > Here is a INFO debug statement when it queries the global_blacklist > table: INFO:userblacklist:reload_sources: got 1 entries from > 'global_blacklist' > > If I run the FIFO command to reload the list, it also errors out if I > do not have the global_blacklist table (which is how I figured out it > was looking for a different table name in the first place). > > 2.) Having some issues on the start of OpenSIPS over time after > turning on the userblacklist module. After I implemented the > userblacklist module and all was working, I was working on the > Event_RabbitMQ module and restarted the service to reload the script. > I started seeing the following: > > ERROR:core:db_check_table_version: querying version for table > userblacklist > ERROR:userblacklist:db_init: during table version check. > ERROR:core:init_mod_child: failed to initializing module > userblacklist, rank 10 > > ERROR:core:db_check_table_version: invalid version 0 for table > userblacklist found, expected 2 > ERROR:userblacklist:db_init: during table version check. > ERROR:core:init_mod_child: failed to initializing module > userblacklist, rank 10 > ERROR:core:tcp_start_processes: init_children failed > > I honestly have no idea how its getting a version 0 on the > userblacklist table. I see in the version table, its set to 2. I tried > setting that to 0 and no luck, same message. I went ahead and dropped > the userblacklist and globalblacklist tables, removed the entries in > the version table and re-ran the SQL script. The service came online. > > This morning, I updated the script to change a IP and restarted the > service. Same issue with the DB versions of the userblacklist table. > If I dropped the tables and re-ran the SQL script, the service came > online. > > Any ideas on this one? > -- > -Jonathan > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jan 3 07:22:33 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 14:22:33 +0200 Subject: [OpenSIPS-Users] postgresql In-Reply-To: <1034273120.76016.1545944069287.JavaMail.zimbra@skillsearch.ca> References: <1034273120.76016.1545944069287.JavaMail.zimbra@skillsearch.ca> Message-ID: <21b15458-4c96-9f0f-436f-cb24773e2cde@opensips.org> Hi Slava, I guess this report is related to you previous post "presence cluster​" - and this kind of errors in postgres are responsible for delaying the timer tasks. The latest master implements a timeout on the postgres queries : http://www.opensips.org/html/docs/modules/3.0.x/db_postgres.html#param_timeout but the default value is 5 seconds. Is your psql server taking more than 5 secs to run the 'delete' query ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/27/2018 10:54 PM, Slava Bendersky wrote: > Hello Everyone, > Presence and postgresql having some problem to query database on > latest master pull > Any help thank you. > > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: > ERROR:db_postgres:db_postgres_submit_query: 0x7f772a429088 PQsendQuery > Error: could not receive data from server: Connection timed out#012 > Query: delete from pua where expires<1545943081 > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: > ERROR:db_postgres:db_postgres_submit_query: 0x7f772a428f98 PQsendQuery > Error: could not receive data from server: Connection timed out#012 > Query: delete from active_watchers where expires<1545943091 > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: > ERROR:core:db_do_delete: error while submitting query > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: > ERROR:core:db_do_delete: error while submitting query > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: > ERROR:db_postgres:db_postgres_store_result: 0x7f772a428f98 - invalid > query, execution aborted > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: > ERROR:db_postgres:db_postgres_store_result: 0x7f772a428f98: > PGRES_FATAL_ERROR > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: > ERROR:db_postgres:db_postgres_store_result: 0x7f772a428f98: > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: > WARNING:db_postgres:db_postgres_delete: unexpected result returned > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: > ERROR:presence:update_db_subs: deleting expired information from database > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: > ERROR:db_postgres:db_postgres_store_result: 0x7f772a429088 - invalid > query, execution aborted > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: > ERROR:db_postgres:db_postgres_store_result: 0x7f772a429088: > PGRES_FATAL_ERROR > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: > ERROR:db_postgres:db_postgres_store_result: 0x7f772a429088: > Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: > WARNING:db_postgres:db_postgres_delete: unexpected result returned > > > and domain module > > Dec 27 14:38:34 aitossbc01 /usr/sbin/opensips[8867]: > ERROR:core:db_do_query: error while submitting query - [select > domain,attrs from domain where domain='dev.ait.local'] > Dec 27 14:38:34 aitossbc01 /usr/sbin/opensips[8867]: > ERROR:domain:is_domain_local_pvar: Error while querying database > > > volga629 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jan 3 07:25:32 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 14:25:32 +0200 Subject: [OpenSIPS-Users] Integrate OpenSIPS with VG202 Audio Gateway In-Reply-To: References: Message-ID: <3c8f7722-bdb4-f566-fdcf-241bca71c424@opensips.org> Hi Vikash, You said you created the user in the OPenSIPS database. Have you configured some end-devices (with those users) and tried to register them ? Regards and A Happy New Year, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 01/02/2019 02:41 PM, Vikash Tibrewal wrote: > > Hello All, > > I am new to OpenSIPS and I want to integrate the OpenSIPS with VG202 > audio gateway and connect that with hardphone. > > I have already installed OpenSIPS v 2.4 on Ubuntu 16.04 and its > running fine and I have installed the OpenSIPS control panel and > created the users in that. > > However, I am not getting any clue how to integrate OpenSIPS with > Hardphone and make the calls? > > Can you please suggest what are the options available for integration? > > Regards, > > Vikash Tibrewal > > ===================================================== > Please refer to http://www.aricent.com/email-disclaimer > for important disclosures regarding this electronic communication. > ===================================================== > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jan 3 10:43:19 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 17:43:19 +0200 Subject: [OpenSIPS-Users] handle_publish wrong destination In-Reply-To: References: Message-ID: <67f58a65-7008-c63d-373f-9118edcd5cbe@opensips.org> Schneur, There is a bit of a confusion in what you are saying. In your original post you mentioned " but on incoming calls the PUBLISH is sent to the internal IP address of the subscribers " . Maybe you meant NOTIFYes ? these are the only requests sent back by OpenSIPS to the subscriber. PUBLISH requests are generated by the pua_xxxx (like pua_dialoginfo) modules and consumed by the presence module (via the handle_publish). And when handling the publish, OpenSIPS is generating and sending out the NOTIFYes. So which ones are you complaining about ? Just for the fact, the PUBLISHes generated by pua_dialoginfo are sent to the fix destination give by the "presence_server" module parameter. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 01/03/2019 03:04 PM, Schneur Rosenberg wrote: > The publish goes to the internal address of the subscribers, instead > of going to itself, it goes nowhere and no NOTIFY is ever sent, here > is a sample PUBLISH > > PUBLISH sip:mysipuser at 192.168.1.2:5060 > SIP/2.0 > Via: SIP/2.0/UDP 81.18.5.78:5060;branch=z9hG4bKf248.d893da14.0 > To: sip:mysipuser at 192.168.1.2:5060 > From: >;tag=fcab47ce4481840ab954c8dc7f9d517f-9b26 > CSeq: 10 PUBLISH > Call-ID: 68b929aa6f905336-24591 at 81.18.5.78 > > Max-Forwards: 70 > Content-Length: 592 > User-Agent: OpenSIPS (2.4.1 (x86_64/linux)) > Event: dialog > Expires: 43201 > Content-Type: application/dialog-info+xml > > > state="partial" entity="sip:mysipuser at 192.168.1.2:5060 > "> id="532c2f370f16667850aa092440c55325 at 66.36.230.75 > > 060" call-id="532c2f370f16667850aa092440c55325 at 81.18.5.80:5060 > " > local-tag="425302105" remote-tag="as4b8896ec" > direction="recipient">early y>sip:2125551010 at 81.18.5.80 > uri="sip:2125551010 at 81.18.5.80 > "/>sip:mysipuser at 192.168.1.2:5060 > > > > On Thu, Jan 3, 2019, 1:58 PM Bogdan-Andrei Iancu wrote: > > Hi Schneur, > > For the incoming calls, the PUBLISH is sent to the internal IP > address > or the NOTIFYes ?? > > Regards and A Happy New Year, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 2019 > https://www.opensips.org/events/Summit-2019Amsterdam/ > > On 12/23/2018 08:37 AM, Schneur Rosenberg wrote: > > HI, my presence was working properly and now I realized that its not > > working anymore for incoming calls, I dont have a external presence > > server, I use the pua_dialoginfo, I did a trace and I saw that for > > outgoing calls the system sends a PUBLISH to itself and then > properly > > sends a NOTIFY to the subscribers, but on incoming calls the PUBLISH > > is sent to the internal IP address of the subscribers and of course > > does not get anywhere, what am I doing wrong? > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From E75A4669 at exemail.com.au Thu Jan 3 11:18:26 2019 From: E75A4669 at exemail.com.au (Alexander Jankowsky) Date: Fri, 4 Jan 2019 02:48:26 +1030 Subject: [OpenSIPS-Users] Integrate OpenSIPS with VG202 Audio Gateway In-Reply-To: References: Message-ID: <000501d4a37f$f52d0750$df8715f0$@exemail.com.au> Hello Vikash, You might need to try some cheap used second hand VOIP phones first. See if you really have OpenSIPS working alright from remote locations. Alex From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Vikash Tibrewal Sent: Wednesday, 2 January 2019 11:12 PM To: users at lists.opensips.org Subject: [OpenSIPS-Users] Integrate OpenSIPS with VG202 Audio Gateway Hello All, I am new to OpenSIPS and I want to integrate the OpenSIPS with VG202 audio gateway and connect that with hardphone. I have already installed OpenSIPS v 2.4 on Ubuntu 16.04 and its running fine and I have installed the OpenSIPS control panel and created the users in that. However, I am not getting any clue how to integrate OpenSIPS with Hardphone and make the calls? Can you please suggest what are the options available for integration? Regards, Vikash Tibrewal ===================================================== Please refer to http://www.aricent.com/email-disclaimer for important disclosures regarding this electronic communication. ===================================================== -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladp at opensips.org Thu Jan 3 12:22:55 2019 From: vladp at opensips.org (Vlad Patrascu) Date: Thu, 3 Jan 2019 19:22:55 +0200 Subject: [OpenSIPS-Users] I deleted an entry from clusterer table , but still opensips try to ping that node . In-Reply-To: References: Message-ID: Hi Sasmita, By default, there is no clusterer replication if "replicate_contacts_to" parameter is not set in usrloc. Also, even if another node is sending replication packets, they will no get processed on the receiving node unless "accept_replicated_contacts" is set. On a typical setup, both these parameter should be set on all nodes. Are you getting any other errors in the logs besides that "parameter not found" ? Btw, I strongly suggest updating to 2.4 as it has received major upgrades in terms of clustering. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 01/02/2019 12:10 PM, Sasmita Panda wrote: > Hi, > I have another doubt . Please do help me . > > When I am reading usrloc module document , its saying in a cluster if > we want to replicate the contacts across the cluster then we have to > set a parameter as below . > modparam("usrloc", "replicate_contacts_to", 1) > The default value is 0 , where no cluster id is mentioned . > I have not set this , I have a cluster having 2 node . While I am registering a user , the contact is getting replicated between 2 nodes . > If I am trying to mention this parameter , then opensips is not getting started . Its saying > *Parameter not found in module * > *So , my question is , if this parameter is not set , still how contact > replication is happening ? Is this the default behavior of cluster > module ? * > *May be my question is foolish ,it will be great if anybody will > explain this . * > ** > ** > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Wed, Jan 2, 2019 at 12:22 PM Sasmita Panda > wrote: > > Hi   Sammy, > > Yes , you are right . I need to reload the cluster data through MI > command . After reloading its seems fine . > > I was not aware about the fact that the cluster data also get > shared with all nodes when I am adding that in 1 node only . > > Thank you for your explanation . Its really helpful . > > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Mon, Dec 31, 2018 at 10:25 PM SamyGo > wrote: > > Hi, > Did you restart OpenSIPS process on both node1, and 2 > simultaneously ? The way I look at this is one of the two > nodes kept the 3rd one in the memory and restarting both nodes > one at a time resulted in both sharing their node structure > and hence node3 stayed visible. > I think possible way to remove a node gracefully would be to > disable the node via the MI command and then remove from DB. I > will try doing this on my test setup as well. > > Regards, > Sammy > > On Fri, Dec 28, 2018 at 6:40 AM Sasmita Panda > > wrote: > > Hi All, > > I have a cluster of 2 nodes . Both in working condition .  >   Then I added another node in the same cluster which is > down . > > I restarted the opensips process , so it starts pinging > the new node to check its status .  As the new node is > down  , other nodes in the cluster wont get any reply for > the ping  . Then I remove the 3rd node from the cluster > table and restart the opensips process . > > Now what I am getting in logs is , still the 2 working > node in the cluster try to ping the 3rd node which is not > in the DB . > > Is this an issue on the cluster module or I am doing > something wrong ?? Please help me . > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladp at opensips.org Thu Jan 3 12:32:41 2019 From: vladp at opensips.org (Vlad Patrascu) Date: Thu, 3 Jan 2019 19:32:41 +0200 Subject: [OpenSIPS-Users] Query regarding opensips cluster module . In-Reply-To: References: Message-ID: <18a514bb-a2ab-61c8-2e5f-58d247b2a1ca@opensips.org> Hi, You should issue the 'clusterer_reload' MI command on all running nodes after adding a new one in the DB and no restart is necessary. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 12/28/2018 09:56 AM, Sasmita Panda wrote: > version: opensips 2.2.4 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > svn revision: 3247:3606M > main.c compiled on 09:50:32 Nov 12 2018 with gcc 4.8.3 > > > This is the entire version . Can anyone let me know , whether I am in > the latest version or I need to update my code ?? > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Thu, Dec 27, 2018 at 6:33 PM Sasmita Panda > wrote: > > Hi All, > > I am using below version of opensips . > version: opensips 2.2.4 > > I have configured cluster module and I have 2 node in a single > cluster . > Lets say I am adding another node into the same cluster . Each > time I have to update the config file with the new node > information or add the new entry in the DB and restart the process . > > I don't want to disturb the running setup while adding new node .  > Is there a way through which I can do it in runtime . > > for example : when I am updating something in dynamic routing , I > am adding that through command line and reload the db to have the > updated information . I am not force to restart the process . > > Can cluster module also work in the same way ? There is no option > to add the DB entry through command line also . Is that's also a > limitation in cluster module ? > > Please do help me . I am not getting anything regarding this in > the module documentation . > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Jan 3 13:16:41 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 3 Jan 2019 20:16:41 +0200 Subject: [OpenSIPS-Users] handle_publish wrong destination In-Reply-To: References: <67f58a65-7008-c63d-373f-9118edcd5cbe@opensips.org> Message-ID: <50bc7c9b-f05f-5d78-03ba-7eaa9d746337@opensips.org> The RURI/To/From are correlated with the presentity URI (you want to publish). I guess you use dialoginfo_set() in your script ? Do it before lookup(location), to learn the AOR and not the contact of the callee. Or, use the caller_spec_param and callee_spec_param to construct your own presentities Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 01/03/2019 07:28 PM, Schneur Rosenberg wrote: > Ok, now its sending the PUBLISH to the proper place, but the RURI, To, > FROM all have the ip of the phone after the @ instead of having the > DOMAIN that it registered as, how do I change it? > > On Thu, Jan 3, 2019 at 7:20 PM Schneur Rosenberg > wrote: >> I just realized that I was missing modparam("pua_dialoginfo", >> "presence_server", "sip:xxx.xxx.xxx.xxx:5060") I will test with it now >> >> On Thu, Jan 3, 2019 at 7:10 PM Schneur Rosenberg >> wrote: >>> Sorry for not being clear, the PUBLISH goes to the address of the >>> SUBSCRIBER, no NOTIFY gets sent out because the presence server which >>> is itself never gets notified of the state change, the server address >>> is set in OpenSIPS as modparam("presence", "server_address", >>> "sip:xxx.xxx.xxx.xxx:5060") (address masked >>> >>> On Thu, Jan 3, 2019 at 5:43 PM Bogdan-Andrei Iancu wrote: >>>> Schneur, >>>> >>>> There is a bit of a confusion in what you are saying. In your original post you mentioned " but on incoming calls the PUBLISH is sent to the internal IP address of the subscribers " . Maybe you meant NOTIFYes ? these are the only requests sent back by OpenSIPS to the subscriber. >>>> PUBLISH requests are generated by the pua_xxxx (like pua_dialoginfo) modules and consumed by the presence module (via the handle_publish). And when handling the publish, OpenSIPS is generating and sending out the NOTIFYes. >>>> >>>> So which ones are you complaining about ? >>>> >>>> Just for the fact, the PUBLISHes generated by pua_dialoginfo are sent to the fix destination give by the "presence_server" module parameter. >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> OpenSIPS Summit 2019 >>>> https://www.opensips.org/events/Summit-2019Amsterdam/ >>>> >>>> On 01/03/2019 03:04 PM, Schneur Rosenberg wrote: >>>> >>>> The publish goes to the internal address of the subscribers, instead of going to itself, it goes nowhere and no NOTIFY is ever sent, here is a sample PUBLISH >>>> >>>> PUBLISH sip:mysipuser at 192.168.1.2:5060 SIP/2.0 >>>> Via: SIP/2.0/UDP 81.18.5.78:5060;branch=z9hG4bKf248.d893da14.0 >>>> To: sip:mysipuser at 192.168.1.2:5060 >>>> From: ;tag=fcab47ce4481840ab954c8dc7f9d517f-9b26 >>>> CSeq: 10 PUBLISH >>>> Call-ID: 68b929aa6f905336-24591 at 81.18.5.78 >>>> Max-Forwards: 70 >>>> Content-Length: 592 >>>> User-Agent: OpenSIPS (2.4.1 (x86_64/linux)) >>>> Event: dialog >>>> Expires: 43201 >>>> Content-Type: application/dialog-info+xml >>>> >>>> >>>> early>>> y>sip:2125551010 at 81.18.5.80sip:mysipuser at 192.168.1.2:5060 >>>> >>>> >>>> On Thu, Jan 3, 2019, 1:58 PM Bogdan-Andrei Iancu >>>> Hi Schneur, >>>>> >>>>> For the incoming calls, the PUBLISH is sent to the internal IP address >>>>> or the NOTIFYes ?? >>>>> >>>>> Regards and A Happy New Year, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> >>>>> OpenSIPS Founder and Developer >>>>> https://www.opensips-solutions.com >>>>> OpenSIPS Summit 2019 >>>>> https://www.opensips.org/events/Summit-2019Amsterdam/ >>>>> >>>>> On 12/23/2018 08:37 AM, Schneur Rosenberg wrote: >>>>>> HI, my presence was working properly and now I realized that its not >>>>>> working anymore for incoming calls, I dont have a external presence >>>>>> server, I use the pua_dialoginfo, I did a trace and I saw that for >>>>>> outgoing calls the system sends a PUBLISH to itself and then properly >>>>>> sends a NOTIFY to the subscribers, but on incoming calls the PUBLISH >>>>>> is sent to the internal IP address of the subscribers and of course >>>>>> does not get anywhere, what am I doing wrong? >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Fri Jan 4 06:21:49 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 4 Jan 2019 13:21:49 +0200 Subject: [OpenSIPS-Users] postgresql In-Reply-To: <21b15458-4c96-9f0f-436f-cb24773e2cde@opensips.org> References: <1034273120.76016.1545944069287.JavaMail.zimbra@skillsearch.ca> <21b15458-4c96-9f0f-436f-cb24773e2cde@opensips.org> Message-ID: For the people interested in an follow up, the discussion around the topic moved to the github tracker: https://github.com/OpenSIPS/opensips/issues/1579 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 01/03/2019 02:22 PM, Bogdan-Andrei Iancu wrote: > Hi Slava, > > I guess this report is related to you previous post "presence > cluster​" - and this kind of errors in postgres are responsible for > delaying the timer tasks. > > The latest master implements a timeout on the postgres queries : > http://www.opensips.org/html/docs/modules/3.0.x/db_postgres.html#param_timeout > but the default value is 5 seconds. Is your psql server taking more > than 5 secs to run the 'delete' query ?? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 2019 > https://www.opensips.org/events/Summit-2019Amsterdam/ > On 12/27/2018 10:54 PM, Slava Bendersky wrote: >> Hello Everyone, >> Presence and postgresql having some problem to query database on >> latest master pull >> Any help thank you. >> >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: >> ERROR:db_postgres:db_postgres_submit_query: 0x7f772a429088 >> PQsendQuery Error: could not receive data from server: Connection >> timed out#012 Query: delete from pua where expires<1545943081 >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: >> ERROR:db_postgres:db_postgres_submit_query: 0x7f772a428f98 >> PQsendQuery Error: could not receive data from server: Connection >> timed out#012 Query: delete from active_watchers where expires<1545943091 >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: >> ERROR:core:db_do_delete: error while submitting query >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: >> ERROR:core:db_do_delete: error while submitting query >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: >> ERROR:db_postgres:db_postgres_store_result: 0x7f772a428f98 - invalid >> query, execution aborted >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: >> ERROR:db_postgres:db_postgres_store_result: 0x7f772a428f98: >> PGRES_FATAL_ERROR >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: >> ERROR:db_postgres:db_postgres_store_result: 0x7f772a428f98: >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: >> WARNING:db_postgres:db_postgres_delete: unexpected result returned >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8868]: >> ERROR:presence:update_db_subs: deleting expired information from database >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: >> ERROR:db_postgres:db_postgres_store_result: 0x7f772a429088 - invalid >> query, execution aborted >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: >> ERROR:db_postgres:db_postgres_store_result: 0x7f772a429088: >> PGRES_FATAL_ERROR >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: >> ERROR:db_postgres:db_postgres_store_result: 0x7f772a429088: >> Dec 27 14:38:24 aitossbc01 /usr/sbin/opensips[8864]: >> WARNING:db_postgres:db_postgres_delete: unexpected result returned >> >> >> and domain module >> >> Dec 27 14:38:34 aitossbc01 /usr/sbin/opensips[8867]: >> ERROR:core:db_do_query: error while submitting query - [select >> domain,attrs from domain where domain='dev.ait.local'] >> Dec 27 14:38:34 aitossbc01 /usr/sbin/opensips[8867]: >> ERROR:domain:is_domain_local_pvar: Error while querying database >> >> >> volga629 >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From jim at devito.cc Fri Jan 4 09:40:38 2019 From: jim at devito.cc (Jim DeVito) Date: Fri, 4 Jan 2019 09:40:38 -0500 Subject: [OpenSIPS-Users] Dynamic Routing dr_carrier state column understanding Message-ID: Hi Guys, Maybe i'm misunderstanding this column. I understand how to enable and disable a carrier via MI commands during run time. However I want to be able to modify the database (dr_carriers) state column and have that respected at next reload. Currently I can change that to 0 or 1 but the carrier is always marked as enabled when I check at run time. I've tried with persistent_state set to 0 and 1 with no effect. Thanks! ------------- Jim DeVito -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Jan 4 11:08:55 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 4 Jan 2019 18:08:55 +0200 Subject: [OpenSIPS-Users] Dynamic Routing dr_carrier state column understanding In-Reply-To: References: Message-ID: Hi Jim, That column is read by OpenSIPS only at startup. Basically it give OpenSIPS the starting values for the GW states. If you change the value in DB, it will have no effect on OpenSIPS until the next restart. On the other side, OpenSIPS is able to update (write) that column if the in-memory state of the GW does change (via MI or ping detection) - but you need to be sure that 'persistent_state' is on (see http://www.opensips.org/html/docs/modules/2.4.x/drouting.html#param_persistent_state). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 01/04/2019 04:40 PM, Jim DeVito wrote: > Hi Guys, > > Maybe i'm misunderstanding this column. I understand how to enable and > disable a carrier via MI commands during run time. However I want to > be able to modify the database (dr_carriers) state column and have > that respected at next reload. Currently I can change that to 0 or 1 > but the carrier is always marked as enabled when I check at run time. > > I've tried with persistent_state set to 0 and 1 with no effect. > > Thanks! > > ------------- > Jim DeVito > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From jim at devito.cc Fri Jan 4 11:18:59 2019 From: jim at devito.cc (Jim DeVito) Date: Fri, 4 Jan 2019 11:18:59 -0500 Subject: [OpenSIPS-Users] Dynamic Routing dr_carrier state column understanding In-Reply-To: References: Message-ID: Hi Bogdan, I think I understand that my problem is that the carriers are always enabled when OpenSIPS starts even though the state column is set to 0. That is part I don't understand. Do you have any thoughts there? On Fri, Jan 4, 2019 at 11:09 AM Bogdan-Andrei Iancu wrote: > Hi Jim, > > That column is read by OpenSIPS only at startup. Basically it give > OpenSIPS the starting values for the GW states. If you change the value in > DB, it will have no effect on OpenSIPS until the next restart. > > On the other side, OpenSIPS is able to update (write) that column if the > in-memory state of the GW does change (via MI or ping detection) - but you > need to be sure that 'persistent_state' is on (see > http://www.opensips.org/html/docs/modules/2.4.x/drouting.html#param_persistent_state > ). > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 2019 > https://www.opensips.org/events/Summit-2019Amsterdam/ > > On 01/04/2019 04:40 PM, Jim DeVito wrote: > > Hi Guys, > > Maybe i'm misunderstanding this column. I understand how to enable and > disable a carrier via MI commands during run time. However I want to be > able to modify the database (dr_carriers) state column and have that > respected at next reload. Currently I can change that to 0 or 1 but the > carrier is always marked as enabled when I check at run time. > > I've tried with persistent_state set to 0 and 1 with no effect. > > Thanks! > > ------------- > Jim DeVito > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- ------------- Jim DeVito Mobile 216.507.9497 -------------- next part -------------- An HTML attachment was scrubbed... URL: From jim at devito.cc Fri Jan 4 11:33:28 2019 From: jim at devito.cc (Jim DeVito) Date: Fri, 4 Jan 2019 11:33:28 -0500 Subject: [OpenSIPS-Users] Dynamic Routing dr_carrier state column understanding In-Reply-To: References: Message-ID: OK. I think I found it. It is just confusing. In the DB a state of 0 means ENABLED and a state of 1 means DISABLED. When sending MI commands 0 will DISABLE and 1 will ENABLE. That seems a little backwards but I'm on 2.2.7 so maybe that was fixed in a newer version. Thanks for the reply! On Fri, Jan 4, 2019 at 11:09 AM Bogdan-Andrei Iancu wrote: > Hi Jim, > > That column is read by OpenSIPS only at startup. Basically it give > OpenSIPS the starting values for the GW states. If you change the value in > DB, it will have no effect on OpenSIPS until the next restart. > > On the other side, OpenSIPS is able to update (write) that column if the > in-memory state of the GW does change (via MI or ping detection) - but you > need to be sure that 'persistent_state' is on (see > http://www.opensips.org/html/docs/modules/2.4.x/drouting.html#param_persistent_state > ). > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 2019 > https://www.opensips.org/events/Summit-2019Amsterdam/ > > On 01/04/2019 04:40 PM, Jim DeVito wrote: > > Hi Guys, > > Maybe i'm misunderstanding this column. I understand how to enable and > disable a carrier via MI commands during run time. However I want to be > able to modify the database (dr_carriers) state column and have that > respected at next reload. Currently I can change that to 0 or 1 but the > carrier is always marked as enabled when I check at run time. > > I've tried with persistent_state set to 0 and 1 with no effect. > > Thanks! > > ------------- > Jim DeVito > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- ------------- Jim DeVito Mobile 216.507.9497 -------------- next part -------------- An HTML attachment was scrubbed... URL: From vikash.tibrewal at aricent.com Thu Jan 3 09:55:02 2019 From: vikash.tibrewal at aricent.com (Vikash Tibrewal) Date: Thu, 3 Jan 2019 14:55:02 +0000 Subject: [OpenSIPS-Users] Integrate OpenSIPS with VG202 Audio Gateway In-Reply-To: <3c8f7722-bdb4-f566-fdcf-241bca71c424@opensips.org> References: <3c8f7722-bdb4-f566-fdcf-241bca71c424@opensips.org> Message-ID: Hello Bogdan, Thanks for your quick response. I have just created the users in OpenSIPS CP but not configured with any devices as I am not getting any clue how to configure them with Hardphone. I have watched the video from your OpenSIPS page in which softphone are connected but I have to connect hardphone. We are trying to connect OpenSIPS with Cisco VG202 voice gateway, Do you have any idea how do we connect OpenSIPS with Voice gateway? Regards, Vikash Tibrewal Technical Leader vikash.tibrewal at aricent.com | Mobile +44 781 005 3819 [cid:c8057d60-269c-417f-a730-b9bb2a6eefec] From: Bogdan-Andrei Iancu Sent: 03 January 2019 12:26 To: OpenSIPS users mailling list ; Vikash Tibrewal Subject: Re: [OpenSIPS-Users] Integrate OpenSIPS with VG202 Audio Gateway ** This mail has been sent from an external source ** Hi Vikash, You said you created the user in the OPenSIPS database. Have you configured some end-devices (with those users) and tried to register them ? Regards and A Happy New Year, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 01/02/2019 02:41 PM, Vikash Tibrewal wrote: Hello All, I am new to OpenSIPS and I want to integrate the OpenSIPS with VG202 audio gateway and connect that with hardphone. I have already installed OpenSIPS v 2.4 on Ubuntu 16.04 and its running fine and I have installed the OpenSIPS control panel and created the users in that. However, I am not getting any clue how to integrate OpenSIPS with Hardphone and make the calls? Can you please suggest what are the options available for integration? Regards, Vikash Tibrewal ===================================================== Please refer to http://www.aricent.com/email-disclaimer for important disclosures regarding this electronic communication. ===================================================== _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ===================================================== Please refer to http://www.aricent.com/email-disclaimer for important disclosures regarding this electronic communication. ===================================================== -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 6531 bytes Desc: image001.png URL: From bogdan at opensips.org Fri Jan 4 11:46:34 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 4 Jan 2019 18:46:34 +0200 Subject: [OpenSIPS-Users] Dynamic Routing dr_carrier state column understanding In-Reply-To: References: Message-ID: Yes, I have to admit this is a bit confusion. It is not yet changed as it is a bit delicate from the backward compatibility perspective. Even more, from the potential mess it may create during an upgrade :(. But sooner or later, we will have to sort this out. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 01/04/2019 06:33 PM, Jim DeVito wrote: > OK. I think I found it. It is just confusing. In the DB a state of 0 > means ENABLED and a state of 1 means DISABLED. When sending MI > commands 0 will DISABLE and 1 will ENABLE. That seems a little > backwards but I'm on 2.2.7 so maybe that was fixed in a newer version. > > Thanks for the reply! > > On Fri, Jan 4, 2019 at 11:09 AM Bogdan-Andrei Iancu > > wrote: > > Hi Jim, > > That column is read by OpenSIPS only at startup. Basically it give > OpenSIPS the starting values for the GW states. If you change the > value in DB, it will have no effect on OpenSIPS until the next > restart. > > On the other side, OpenSIPS is able to update (write) that column > if the in-memory state of the GW does change (via MI or ping > detection) - but you need to be sure that 'persistent_state' is on > (see > http://www.opensips.org/html/docs/modules/2.4.x/drouting.html#param_persistent_state). > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 2019 > https://www.opensips.org/events/Summit-2019Amsterdam/ > > On 01/04/2019 04:40 PM, Jim DeVito wrote: >> Hi Guys, >> >> Maybe i'm misunderstanding this column. I understand how to >> enable and disable a carrier via MI commands during run time. >> However I want to be able to modify the database (dr_carriers) >> state column and have that respected at next reload. Currently I >> can change that to 0 or 1 but the carrier is always marked as >> enabled when I check at run time. >> >> I've tried with persistent_state set to 0 and 1 with no effect. >> >> Thanks! >> >> ------------- >> Jim DeVito >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > ------------- > Jim DeVito > Mobile 216.507.9497 -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Fri Jan 4 11:56:04 2019 From: govoiper at gmail.com (SamyGo) Date: Fri, 4 Jan 2019 11:56:04 -0500 Subject: [OpenSIPS-Users] Integrate OpenSIPS with VG202 Audio Gateway In-Reply-To: References: <3c8f7722-bdb4-f566-fdcf-241bca71c424@opensips.org> Message-ID: Hi Vikash, I assume your CIsco gateway support SIP ? can you register your cisco g/w with opensips ? or is it possible for the gateway to receive/send calls from a statis IP (of opensips) ? You might need to provide some more insights to your cisco gateway on it's capabilities and supported protocols for anybody to help you. Regards, Sammy. On Fri, Jan 4, 2019 at 11:39 AM Vikash Tibrewal wrote: > Hello Bogdan, > > > > Thanks for your quick response. > > > > I have just created the users in OpenSIPS CP but not configured with any > devices as I am not getting any clue how to configure them with Hardphone. > > > > I have watched the video from your OpenSIPS page in which softphone are > connected but I have to connect hardphone. > > > > We are trying to connect OpenSIPS with Cisco VG202 voice gateway, Do you > have any idea how do we connect OpenSIPS with Voice gateway? > > > > Regards, > > *Vikash Tibrewal* > > Technical Leader > > *vikash.tibrewal at aricent.com * | Mobile +44 > 781 005 3819 > > > > [image: cid:c8057d60-269c-417f-a730-b9bb2a6eefec] > > > > *From:* Bogdan-Andrei Iancu > *Sent:* 03 January 2019 12:26 > *To:* OpenSIPS users mailling list ; Vikash > Tibrewal > *Subject:* Re: [OpenSIPS-Users] Integrate OpenSIPS with VG202 Audio > Gateway > > > > ** This mail has been sent from an external source ** > > > > Hi Vikash, > > You said you created the user in the OPenSIPS database. Have you > configured some end-devices (with those users) and tried to register them ? > > Regards and A Happy New Year, > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > OpenSIPS Summit 2019 > > https://www.opensips.org/events/Summit-2019Amsterdam/ > > On 01/02/2019 02:41 PM, Vikash Tibrewal wrote: > > Hello All, > > > > I am new to OpenSIPS and I want to integrate the OpenSIPS with VG202 audio > gateway and connect that with hardphone. > > > > I have already installed OpenSIPS v 2.4 on Ubuntu 16.04 and its running > fine and I have installed the OpenSIPS control panel and created the users > in that. > > > > However, I am not getting any clue how to integrate OpenSIPS with > Hardphone and make the calls? > > > > Can you please suggest what are the options available for integration? > > > > Regards, > > Vikash Tibrewal > > > > ===================================================== > Please refer to http://www.aricent.com/email-disclaimer > for important disclosures regarding this electronic communication. > ===================================================== > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ===================================================== > Please refer to http://www.aricent.com/email-disclaimer > for important disclosures regarding this electronic communication. > ===================================================== > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 6531 bytes Desc: not available URL: From mickael at winlux.fr Sat Jan 5 14:02:27 2019 From: mickael at winlux.fr (Mickael Hubert) Date: Sat, 5 Jan 2019 20:02:27 +0100 Subject: [OpenSIPS-Users] Avoid plain text password in configuration files Message-ID: Hi all, I'm looking for a way to avoid all plain text password into configuration files. maybe store sensibles data into secret file and read variables into opensips configuration file ? Ex: *secret file:* MYSQL_USER: opensips MYSQL_PWD: 4845123121 ... *Configuration file:* from: modparam("drouting", "db_url", "mysql://opensips:4845123121 at 1.1.1.1/opensips") To: modparam("drouting", "db_url", "mysql://$MYSQL_USER:$MYSQL_PWD@$MYSQL_HOST/opensips") Or use Ansible with jinja template ? modparam("drouting", "db_url", "mysql://{{ MYSQL_USER }}:{{ MYSQL_PWD }}@{{ MYSQL_HOST }}/opensips") The goal is push all configuration on our gitlab (without the "secret file") Do you have another way ? thanks in advance PS: happy new year everyone ! -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexei.vasilyev at gmail.com Mon Jan 7 03:23:51 2019 From: alexei.vasilyev at gmail.com (vasilevalex) Date: Mon, 7 Jan 2019 01:23:51 -0700 (MST) Subject: [OpenSIPS-Users] Avoid plain text password in configuration files In-Reply-To: References: Message-ID: <1546849431415-0.post@n2.nabble.com> Hi Mickael, I think most people keep configuration in git. Just use m4. Like this opensips_defs.m4: divert(-1) ifdef(`PRODUCTION_ENV', `define(`MYSQL_DB',`opensips')', `define(`MYSQL_DB',`opensips_test')') define(`MYSQL1_URL', `mysql://MYSQL_USER:MYSQL_PASSWD at MYSQL1_HOST/MYSQL_DB') define(`MYSQL2_URL', `mysql://MYSQL_USER:MYSQL_PASSWD at MYSQL2_HOST/MYSQL_DB') define(`MYSQL3_URL', `mysql://MYSQL_USER:MYSQL_PASSWD at MYSQL3_HOST/MYSQL_DB') define(`MYSQL4_URL', `mysql://MYSQL_USER:MYSQL_PASSWD at MYSQL4_HOST/MYSQL_DB') divert(0)dnl include(`cfg/opensips_global.m4') include(`cfg/opensips_modules.m4') include(`cfg/opensips_routes.m4') and in secret file opensips.m4 (which is not in git): divert(-1) define(`MYSQL_USER', `opensips') define(`MYSQL_PASSWD', `password') divert(0)dnl include(`cfg/opensips_defs.m4') And final config: m4 opensips.m4 > opensips.cfg ----- --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From nick at altmann.pro Mon Jan 7 04:58:15 2019 From: nick at altmann.pro (Nick Altmann) Date: Mon, 7 Jan 2019 12:58:15 +0300 Subject: [OpenSIPS-Users] Avoid plain text password in configuration files In-Reply-To: <1546849431415-0.post@n2.nabble.com> References: <1546849431415-0.post@n2.nabble.com> Message-ID: > > And final config: > > m4 opensips.m4 > opensips.cfg > Startup script does this automatically when opensips.m4 found. -------------- next part -------------- An HTML attachment was scrubbed... URL: From gerwin.van.de.steeg at vadacom.com Mon Jan 7 06:42:15 2019 From: gerwin.van.de.steeg at vadacom.com (Gerwin van de Steeg) Date: Tue, 8 Jan 2019 00:42:15 +1300 Subject: [OpenSIPS-Users] Avoid plain text password in configuration files In-Reply-To: References: <1546849431415-0.post@n2.nabble.com> Message-ID: If you're ok with them being pulled from a secondary source like a second file not part of your VCS then you'll have to use some form of config generation script like a Makefile or m4 or ansible with some templating. Depends entirely on how comfortable you are with m4 and the other tools. I just filed a feature request earlier today regarding this if you don't want to store the credentials on the system in a config file at all. https://github.com/OpenSIPS/opensips/issues/1581 Cheers, Gerwin On Mon, 7 Jan 2019 at 22:59, Nick Altmann wrote: > And final config: >> >> m4 opensips.m4 > opensips.cfg >> > > Startup script does this automatically when opensips.m4 found. > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mabritoj at gmail.com Mon Jan 7 17:48:42 2019 From: mabritoj at gmail.com (Jonathan Mabrito) Date: Mon, 7 Jan 2019 17:48:42 -0500 Subject: [OpenSIPS-Users] Fraud Detection Module Inquiry Message-ID: Hi All, I am running OpenSIPS 2.3 and I have a inquiry on the Fraud Detection module. I had a critical alert get triggered on the call duration threshold today. No warning came in, just straight to the critical....and way over the critical duration threshold. Looking in the logs, it looks like the critical notification was triggered when the call disconnected/hung up. Guessing the module accounts the start and end time of the call and uses that duration? Does it not keep track of the call from a duration perspective as its happening? Luckily this was just one occurrence but it got me thinking, if fraud was being performed on the system, then the duration notification wont be triggered until the call is hung up...at least the way I have it configured. Is there a way to have it keep auditing ongoing calls? -- -Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: From mickael at winlux.fr Tue Jan 8 00:54:31 2019 From: mickael at winlux.fr (Mickael Hubert) Date: Tue, 8 Jan 2019 06:54:31 +0100 Subject: [OpenSIPS-Users] Avoid plain text password in configuration files In-Reply-To: References: <1546849431415-0.post@n2.nabble.com> Message-ID: Hi Thanks a lot for guys ! I'll try m4 to test. But If prefer ansible ;) maybe change opensips startup script to work with jinja template... ++ ++ Le lun. 7 janv. 2019 12:43, Gerwin van de Steeg < gerwin.van.de.steeg at vadacom.com> a écrit : > If you're ok with them being pulled from a secondary source like a second > file not part of your VCS then you'll have to use some form of config > generation script like a Makefile or m4 or ansible with some templating. > Depends entirely on how comfortable you are with m4 and the other tools. > > I just filed a feature request earlier today regarding this if you don't > want to store the credentials on the system in a config file at all. > https://github.com/OpenSIPS/opensips/issues/1581 > > Cheers, > > Gerwin > > > > > > On Mon, 7 Jan 2019 at 22:59, Nick Altmann wrote: > >> And final config: >>> >>> m4 opensips.m4 > opensips.cfg >>> >> >> Startup script does this automatically when opensips.m4 found. >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Jan 9 07:53:37 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 9 Jan 2019 14:53:37 +0200 Subject: [OpenSIPS-Users] [Reminder] OpenSIPS 3.0 feedback form In-Reply-To: <288ea123-91c9-d678-a8d0-5ede9f64e270@opensips.org> References: <288ea123-91c9-d678-a8d0-5ede9f64e270@opensips.org> Message-ID: <9ea88f74-b82e-a82e-554a-74f9ca25f08b@opensips.org> Hi all, We want to thank you all for providing us feedback in regards to the 3.0 Planing - this is valuable information for us, in order to understand what are the needs and priorities of the OpenSIPS user community. There were hundreds of opinions, here is the centralized score and sorting: http://www.opensips.org/Development/Opensips-3-0-Planning#poll-results We will align the development work to these results and also analyze and take into consideration the suggestions you passed to us (the list will be published later). There was very interesting idea shared with us, thank you for that. Thanks & regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/28/2018 11:23 AM, Bogdan-Andrei Iancu wrote: > Dear all, > > This is a quick reminder - getting involved in the 3.0 OpenSIPS > planning and development, and fill in this *Feature Survey > * > by *6th of January 2019*. > > Be part of OpenSIPS! > > A Happy New Year, > -- > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 2019 > https://www.opensips.org/events/Summit-2019Amsterdam/ > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From jens.erik.rive at ipco.no Wed Jan 9 09:27:18 2019 From: jens.erik.rive at ipco.no (Jens Erik Rive) Date: Wed, 9 Jan 2019 14:27:18 +0000 Subject: [OpenSIPS-Users] CGRateS redundancy Message-ID: Hi, I am running Opensips v.2.4.3 and two instances of cgrates bulidt 10. December 2018. I am trying to implement redundancy by defining the 2 CGRates servers as: modparam("cgrates", "cgrates_engine", "10.40.2.21:2012") # CGRateS01 modparam("cgrates", "cgrates_engine", "10.40.2.23:2012") # CGRateS02 When I kill (poweroff) the 10.40.2.21 instance and try to call cgrates with: cgrates_auth("$cgr(Account)", "$acc_extra(callee_id)") I receive -3 (No suitable CGRateS server found) in the response. I have tested that I can call 10.40.2.23:2012 if modparam("cgrates", "cgrates_engine", "10.40.2.21:2012") # CGRateS01 is omitted from the opensips.cfg script. Is there anything I am doing wrong? Thanks, Jens Erik From razvan at opensips.org Wed Jan 9 11:35:59 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 9 Jan 2019 18:35:59 +0200 Subject: [OpenSIPS-Users] CGRateS redundancy In-Reply-To: References: Message-ID: Hello, Jens Erik! This should work - can you run opensips in debug mode and send us the syslog output? Best regards, Răzvan On 1/9/19 4:27 PM, Jens Erik Rive wrote: > Hi, I am running Opensips v.2.4.3 and two instances of cgrates bulidt 10. December 2018. > I am trying to implement redundancy by defining the 2 CGRates servers as: > > modparam("cgrates", "cgrates_engine", "10.40.2.21:2012") # CGRateS01 > modparam("cgrates", "cgrates_engine", "10.40.2.23:2012") # CGRateS02 > > When I kill (poweroff) the 10.40.2.21 instance and try to call cgrates with: > cgrates_auth("$cgr(Account)", "$acc_extra(callee_id)") > I receive -3 (No suitable CGRateS server found) in the response. > > I have tested that I can call 10.40.2.23:2012 if modparam("cgrates", "cgrates_engine", "10.40.2.21:2012") # CGRateS01 is omitted from the opensips.cfg script. > > Is there anything I am doing wrong? > > Thanks, > Jens Erik > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From mabritoj at gmail.com Wed Jan 9 12:11:11 2019 From: mabritoj at gmail.com (Jonathan Mabrito) Date: Wed, 9 Jan 2019 12:11:11 -0500 Subject: [OpenSIPS-Users] Fraud Detection Module Inquiry In-Reply-To: References: Message-ID: One more inquiry on this module. Is it possible to have the module ignore prefixes being set in the fraud rules? For our scenarios, the FROM Address/user is the same for most of our calls. On our PBX's we do not really give out DID's to each of our users, so when a user makes an outbound call, the FROM address is always going to be same and set to the main number of our system. Within the fraud detection module, I have a prefix of 99 defined (how we denote an outbound call) and starting getting tons of false positive notifications, as the pair being matched would be <800number, 99numberDialed>. This quickly filled up the sequential calls threshold as every call being made had the same FROM Address and matched against the 99 prefix. The way we operate, I created 30 minute intervals Sun-Sat with the same threshold values (48 rows of rules). I am hoping I can omit the prefix matching and just match the dialed number to the appropriate 30 minute time interval? so the pair being matched would be <800 number, full number dialed> ? I tried setting a blank prefix and say "No rule matched" syslog message. On Mon, Jan 7, 2019 at 5:48 PM Jonathan Mabrito wrote: > Hi All, > > I am running OpenSIPS 2.3 and I have a inquiry on the Fraud Detection > module. I had a critical alert get triggered on the call duration threshold > today. No warning came in, just straight to the critical....and way over > the critical duration threshold. > > Looking in the logs, it looks like the critical notification was triggered > when the call disconnected/hung up. Guessing the module accounts the start > and end time of the call and uses that duration? Does it not keep track of > the call from a duration perspective as its happening? > > Luckily this was just one occurrence but it got me thinking, if fraud was > being performed on the system, then the duration notification wont be > triggered until the call is hung up...at least the way I have it > configured. Is there a way to have it keep auditing ongoing calls? > -- > -Jonathan > -- -Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: From jens.erik.rive at ipco.no Thu Jan 10 04:06:58 2019 From: jens.erik.rive at ipco.no (Jens Erik Rive) Date: Thu, 10 Jan 2019 09:06:58 +0000 Subject: [OpenSIPS-Users] CGRateS redundancy In-Reply-To: References: Message-ID: Hello, Răzvan! I am sorry for the inconvenience I have caused you. I have not been able to reproduce this. I don't know why I had the problem yesterday. It may have been the VMs that needed a hard restart after configuration. It is working in as expected now. Best Regards, Jens Erik -----Original Message----- From: Users On Behalf Of Razvan Crainea Sent: onsdag 9. januar 2019 17:36 To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] CGRateS redundancy Hello, Jens Erik! This should work - can you run opensips in debug mode and send us the syslog output? Best regards, Răzvan On 1/9/19 4:27 PM, Jens Erik Rive wrote: > Hi, I am running Opensips v.2.4.3 and two instances of cgrates bulidt 10. December 2018. > I am trying to implement redundancy by defining the 2 CGRates servers as: > > modparam("cgrates", "cgrates_engine", "10.40.2.21:2012") # CGRateS01 > modparam("cgrates", "cgrates_engine", "10.40.2.23:2012") # CGRateS02 > > When I kill (poweroff) the 10.40.2.21 instance and try to call cgrates with: > cgrates_auth("$cgr(Account)", "$acc_extra(callee_id)") I receive -3 > (No suitable CGRateS server found) in the response. > > I have tested that I can call 10.40.2.23:2012 if modparam("cgrates", "cgrates_engine", "10.40.2.21:2012") # CGRateS01 is omitted from the opensips.cfg script. > > Is there anything I am doing wrong? > > Thanks, > Jens Erik > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Thu Jan 10 04:44:29 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 10 Jan 2019 11:44:29 +0200 Subject: [OpenSIPS-Users] CGRateS redundancy In-Reply-To: References: Message-ID: No problem! My assumption is that for a short window both servers were unreachable, that's why opensips was not able to use them. I'm glad it all sorted out now. Best regards, Razvan On 1/10/19 11:06 AM, Jens Erik Rive wrote: > Hello, Răzvan! > > I am sorry for the inconvenience I have caused you. I have not been able to reproduce this. > I don't know why I had the problem yesterday. It may have been the VMs that needed a hard restart after configuration. > It is working in as expected now. > > Best Regards, > Jens Erik > -----Original Message----- > From: Users On Behalf Of Razvan Crainea > Sent: onsdag 9. januar 2019 17:36 > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] CGRateS redundancy > > Hello, Jens Erik! > > This should work - can you run opensips in debug mode and send us the syslog output? > > Best regards, > Răzvan > > On 1/9/19 4:27 PM, Jens Erik Rive wrote: >> Hi, I am running Opensips v.2.4.3 and two instances of cgrates bulidt 10. December 2018. >> I am trying to implement redundancy by defining the 2 CGRates servers as: >> >> modparam("cgrates", "cgrates_engine", "10.40.2.21:2012") # CGRateS01 >> modparam("cgrates", "cgrates_engine", "10.40.2.23:2012") # CGRateS02 >> >> When I kill (poweroff) the 10.40.2.21 instance and try to call cgrates with: >> cgrates_auth("$cgr(Account)", "$acc_extra(callee_id)") I receive -3 >> (No suitable CGRateS server found) in the response. >> >> I have tested that I can call 10.40.2.23:2012 if modparam("cgrates", "cgrates_engine", "10.40.2.21:2012") # CGRateS01 is omitted from the opensips.cfg script. >> >> Is there anything I am doing wrong? >> >> Thanks, >> Jens Erik >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > Meet the OpenSIPS team at the next OpenSIPS Summit: > https://www.opensips.org/events > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From Ben.Newlin at genesys.com Thu Jan 10 14:25:02 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 10 Jan 2019 19:25:02 +0000 Subject: [OpenSIPS-Users] DNS Cache module In-Reply-To: <9ff60eb4-8730-7fcb-c596-abe74773f526@opensips.org> References: <667279fc-8dab-7cb7-7301-e10416234aea@opensips.org> <1F2616F0-039C-4E50-B1B0-7AEED643C044@genesys.com> <9ff60eb4-8730-7fcb-c596-abe74773f526@opensips.org> Message-ID: Bogdan, We are now using 2.4.4. $ opensips -V version: opensips 2.4.4 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. git revision: a42226ccb main.c compiled on 18:03:24 Jan 10 2019 with gcc 7 Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, January 3, 2019 at 5:13 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] DNS Cache module Hi Ben, What exact version / revision of OpenSIPS do you use ? maybe I can help adding some extra debug logs. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/07/2018 08:14 PM, Ben Newlin wrote: Bogdan, That had occurred to me as well. I have verified testing locally that when successful the value is printed in the log. I have also verified manually on the system with the error that the key does exist in the cache, but the value in the cache is empty, just as in the log. That is what I believe is causing the failure, the lookup from the cache is successful (the key exists) but the value is empty, so OpenSIPS cannot route. I have attempted increasing the logging on the system, but it appears the dns_cache module does not log anything further of use. Ben Newlin From: Bogdan-Andrei Iancu Date: Friday, December 7, 2018 at 12:21 PM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] DNS Cache module Hi Ben, IMO, the log itself is broken as the data to be cached is not printable ....so the logs you see may be misleading. When you say "OpenSIPS appeared to not be able to resolve the domain", you mean OpenSIPS is not doing any attempt to solve the FQDN, or you mean OpenSIPS is loading from cash something wrong ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 12/06/2018 10:17 PM, Ben Newlin wrote: Hello, We use the DNS cache module to reduce the time spent querying DNS records. We recently had a customer call failing and we traced the failure to the customer using an FQDN in the Record-Route header. On the ACK, OpenSIPS appeared to not be able to resolve the domain even though it had been successfully resolved on the initial request. I found the log for the DNS Cache module and noticed that the value it was inserting was empty: INFO:dns_cache:put_dnscache_value: putting key [dnscache_customer.domain.com_a] with value [] ttl = 60 This prompted me to examine all of our logs and I found that the value for these DNS Cache logs is always empty, regardless of the domain. It appears the records are not being serialized properly into the cache. The DNS resolution must be succeeding or all of our requests using DNS would be failing, but I have also verified the domains all can be resolved manually on the same box: $ nslookup customer.domain.com Server: 10.27.0.2 Address: 10.27.0.2#53 Non-authoritative answer: Name: customer.domain.com Address: 10.27.172.132 Name: customer.domain.com Address: 10.27.192.211 Name: customer.domain.com Address: 10.27.255.53 Any thoughts? Is there more information I can obtain to determine the cause? Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Jan 11 03:03:33 2019 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 11 Jan 2019 13:33:33 +0530 Subject: [OpenSIPS-Users] I need some help in b2b module . Message-ID: Hi All, version: opensips 2.4.3 (x86_64/linux) I am using siprec module with orex (3rd party software ) . I wanted to send 1 customize header of main dialog to the dialog getting forwarded to recording server . As I know , opensips acts as a b2b UA , and generate its own Invite for recording server in this case . Its only sending the default headers and wont send any information of the initial dialog . I have a header X-Info . I wanted to pass this . I have tried the below thing . loadmodule "b2b_entities.so" loadmodule "b2b_logic.so" modparam("b2b_logic", "custom_headers", "X-Info") I have tried to send custom header through b2b_logic , but that wont work . It wont add the header . What should I do for this ? * Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Jan 11 03:07:10 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 11 Jan 2019 10:07:10 +0200 Subject: [OpenSIPS-Users] I need some help in b2b module . In-Reply-To: References: Message-ID: Hello! Is the X-Info header received in the initial INVITE, or is it added by your script? In case it is received, it should be added if you set it in the custom_headers parameters. If you add it yourself in the script, it won't be added. Nevertheless, you can modify the INVITE generated by b2b in local_route - you can add your header there. Hope this helps! Best regards, Razvan On 1/11/19 10:03 AM, Sasmita Panda wrote: > Hi All, > > version: opensips 2.4.3 (x86_64/linux) > > I am using siprec module with orex (3rd party software ) . > I wanted to send 1 customize header of main dialog to the dialog getting > forwarded to recording server . > > As I know , opensips acts as a b2b UA , and generate its own Invite for > recording server in this case . Its only sending the default headers and > wont send any information of the initial dialog . > > I have a header X-Info . I wanted to pass this . I have tried the below > thing . > > loadmodule "b2b_entities.so" > loadmodule "b2b_logic.so" > modparam("b2b_logic", "custom_headers", "X-Info") > > I have tried to send custom header through b2b_logic , but that wont > work . It wont add the header . What should I do for this ? > > > > > > */ Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From spanda at 3clogic.com Fri Jan 11 03:12:14 2019 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 11 Jan 2019 13:42:14 +0530 Subject: [OpenSIPS-Users] I need some help in b2b module . In-Reply-To: References: Message-ID: X-Info header is coming in the initial Invite , still its not adding . Even I am not getting anything in the logs too. *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Jan 11, 2019 at 1:38 PM Răzvan Crainea wrote: > Hello! > > Is the X-Info header received in the initial INVITE, or is it added by > your script? In case it is received, it should be added if you set it in > the custom_headers parameters. If you add it yourself in the script, it > won't be added. > Nevertheless, you can modify the INVITE generated by b2b in local_route > - you can add your header there. Hope this helps! > > Best regards, > Razvan > > On 1/11/19 10:03 AM, Sasmita Panda wrote: > > Hi All, > > > > version: opensips 2.4.3 (x86_64/linux) > > > > I am using siprec module with orex (3rd party software ) . > > I wanted to send 1 customize header of main dialog to the dialog getting > > forwarded to recording server . > > > > As I know , opensips acts as a b2b UA , and generate its own Invite for > > recording server in this case . Its only sending the default headers and > > wont send any information of the initial dialog . > > > > I have a header X-Info . I wanted to pass this . I have tried the below > > thing . > > > > loadmodule "b2b_entities.so" > > loadmodule "b2b_logic.so" > > modparam("b2b_logic", "custom_headers", "X-Info") > > > > I have tried to send custom header through b2b_logic , but that wont > > work . It wont add the header . What should I do for this ? > > > > > > > > > > > > */ Thanks & Regards/* > > /Sasmita Panda/ > > /Senior Network Testing and Software Engineer/ > > /3CLogic , ph:07827611765/ > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > Meet the OpenSIPS team at the next OpenSIPS Summit: > https://www.opensips.org/events > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Jan 11 04:46:16 2019 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 11 Jan 2019 15:16:16 +0530 Subject: [OpenSIPS-Users] I need some help in b2b module . In-Reply-To: References: Message-ID: My initial INVITE looks like below : Request-Line: INVITE sip:6001 at p2p.i3clogic.com:5507 SIP/2.0 Message Header Via: SIP/2.0/UDP 180.151.95.154:52075 ;rport;branch=z9hG4bKPjc71c77196989435d8a66406b2f3911e9 Max-Forwards: 70 From: sip:webuser at p2p.i3clogic.com:5507 ;tag=f4e76f6568f74697b55a865842214362 To: sip:6001 at p2p.i3clogic.com:5507 Contact: Call-ID: 7fa9909d3769498380c33e52758eca16 CSeq: 11538 INVITE X-Proxy: false Allow: SUBSCRIBE, NOTIFY, REFER, MESSAGE Supported: replaces, timer Session-Expires: 3600 Min-SE: 90 User-Agent: WebAstra X-Info: normal;;A=sip3;C=140;R=4 Content-Type: application/sdp Content-Length: 225 Message Body Below is the Invite opesips generate for recording server : *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Jan 11, 2019 at 1:42 PM Sasmita Panda wrote: > X-Info header is coming in the initial Invite , still its not adding . > > Even I am not getting anything in the logs too. > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Fri, Jan 11, 2019 at 1:38 PM Răzvan Crainea > wrote: > >> Hello! >> >> Is the X-Info header received in the initial INVITE, or is it added by >> your script? In case it is received, it should be added if you set it in >> the custom_headers parameters. If you add it yourself in the script, it >> won't be added. >> Nevertheless, you can modify the INVITE generated by b2b in local_route >> - you can add your header there. Hope this helps! >> >> Best regards, >> Razvan >> >> On 1/11/19 10:03 AM, Sasmita Panda wrote: >> > Hi All, >> > >> > version: opensips 2.4.3 (x86_64/linux) >> > >> > I am using siprec module with orex (3rd party software ) . >> > I wanted to send 1 customize header of main dialog to the dialog >> getting >> > forwarded to recording server . >> > >> > As I know , opensips acts as a b2b UA , and generate its own Invite for >> > recording server in this case . Its only sending the default headers >> and >> > wont send any information of the initial dialog . >> > >> > I have a header X-Info . I wanted to pass this . I have tried the below >> > thing . >> > >> > loadmodule "b2b_entities.so" >> > loadmodule "b2b_logic.so" >> > modparam("b2b_logic", "custom_headers", "X-Info") >> > >> > I have tried to send custom header through b2b_logic , but that wont >> > work . It wont add the header . What should I do for this ? >> > >> > >> > >> > >> > >> > */ Thanks & Regards/* >> > /Sasmita Panda/ >> > /Senior Network Testing and Software Engineer/ >> > /3CLogic , ph:07827611765/ >> > >> > _______________________________________________ >> > Users mailing list >> > Users at lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> >> -- >> Răzvan Crainea >> OpenSIPS Core Developer >> http://www.opensips-solutions.com >> Meet the OpenSIPS team at the next OpenSIPS Summit: >> https://www.opensips.org/events >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Jan 11 04:48:43 2019 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 11 Jan 2019 15:18:43 +0530 Subject: [OpenSIPS-Users] I need some help in b2b module . In-Reply-To: References: Message-ID: My initial INVITE looks like below : Request-Line: INVITE sip:6001 at p2p.i3clogic.com:5507 SIP/2.0 Message Header Via: SIP/2.0/UDP 180.151.95.154:52075 ;rport;branch=z9hG4bKPjc71c77196989435d8a66406b2f3911e9 Max-Forwards: 70 From: sip:webuser at p2p.i3clogic.com:5507 ;tag=f4e76f6568f74697b55a865842214362 To: sip:6001 at p2p.i3clogic.com:5507 Contact: Call-ID: 7fa9909d3769498380c33e52758eca16 CSeq: 11538 INVITE X-Proxy: false Allow: SUBSCRIBE, NOTIFY, REFER, MESSAGE Supported: replaces, timer Session-Expires: 3600 Min-SE: 90 User-Agent: WebAstra X-Info: normal;;A=sip3;C=140;R=4 Content-Type: application/sdp Content-Length: 225 Message Body Below is the Invite opesips generate for recording server : Request-Line: INVITE sip:x.x.x.x:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP x.x.x.y:5507;branch=z9hG4bK5ad5.d0081b53.0 To: sip:x.x.x.x:5060 From: ;tag=204a7ab5593cac3dd64ab293b65e4314-ce21 CSeq: 2 INVITE Call-ID: B2B.430.6291654.1547195241 Max-Forwards: 70 Content-Length: 1379 User-Agent: OpenSIPS (2.4.3 (x86_64/linux)) Require: siprec Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42 Contact: Message Body Even though I have mentioned the customer_headers parameter still its not adding . What must be the error ? Is there any issue with siprec module of opensips-2.4.3 ? Please do help me . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Jan 11, 2019 at 3:16 PM Sasmita Panda wrote: > My initial INVITE looks like below : > > Request-Line: INVITE sip:6001 at p2p.i3clogic.com:5507 SIP/2.0 > Message Header > Via: SIP/2.0/UDP 180.151.95.154:52075 > ;rport;branch=z9hG4bKPjc71c77196989435d8a66406b2f3911e9 > Max-Forwards: 70 > From: sip:webuser at p2p.i3clogic.com:5507 > ;tag=f4e76f6568f74697b55a865842214362 > To: sip:6001 at p2p.i3clogic.com:5507 > Contact: > Call-ID: 7fa9909d3769498380c33e52758eca16 > CSeq: 11538 INVITE > X-Proxy: false > Allow: SUBSCRIBE, NOTIFY, REFER, MESSAGE > Supported: replaces, timer > Session-Expires: 3600 > Min-SE: 90 > User-Agent: WebAstra > X-Info: normal;;A=sip3;C=140;R=4 > Content-Type: application/sdp > Content-Length: 225 > Message Body > > > Below is the Invite opesips generate for recording server : > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Fri, Jan 11, 2019 at 1:42 PM Sasmita Panda wrote: > >> X-Info header is coming in the initial Invite , still its not adding . >> >> Even I am not getting anything in the logs too. >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> On Fri, Jan 11, 2019 at 1:38 PM Răzvan Crainea >> wrote: >> >>> Hello! >>> >>> Is the X-Info header received in the initial INVITE, or is it added by >>> your script? In case it is received, it should be added if you set it in >>> the custom_headers parameters. If you add it yourself in the script, it >>> won't be added. >>> Nevertheless, you can modify the INVITE generated by b2b in local_route >>> - you can add your header there. Hope this helps! >>> >>> Best regards, >>> Razvan >>> >>> On 1/11/19 10:03 AM, Sasmita Panda wrote: >>> > Hi All, >>> > >>> > version: opensips 2.4.3 (x86_64/linux) >>> > >>> > I am using siprec module with orex (3rd party software ) . >>> > I wanted to send 1 customize header of main dialog to the dialog >>> getting >>> > forwarded to recording server . >>> > >>> > As I know , opensips acts as a b2b UA , and generate its own Invite >>> for >>> > recording server in this case . Its only sending the default headers >>> and >>> > wont send any information of the initial dialog . >>> > >>> > I have a header X-Info . I wanted to pass this . I have tried the >>> below >>> > thing . >>> > >>> > loadmodule "b2b_entities.so" >>> > loadmodule "b2b_logic.so" >>> > modparam("b2b_logic", "custom_headers", "X-Info") >>> > >>> > I have tried to send custom header through b2b_logic , but that wont >>> > work . It wont add the header . What should I do for this ? >>> > >>> > >>> > >>> > >>> > >>> > */ Thanks & Regards/* >>> > /Sasmita Panda/ >>> > /Senior Network Testing and Software Engineer/ >>> > /3CLogic , ph:07827611765/ >>> > >>> > _______________________________________________ >>> > Users mailing list >>> > Users at lists.opensips.org >>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> > >>> >>> -- >>> Răzvan Crainea >>> OpenSIPS Core Developer >>> http://www.opensips-solutions.com >>> Meet the OpenSIPS team at the next OpenSIPS Summit: >>> https://www.opensips.org/events >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Jan 11 04:52:46 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 11 Jan 2019 11:52:46 +0200 Subject: [OpenSIPS-Users] I need some help in b2b module . In-Reply-To: References: Message-ID: <40529502-6f6f-c3e5-c519-500c45a720c8@opensips.org> Hi, Sasmita! You are right, I've checked now and unfortunately in this case the siprec module does not send the request headers to B2B. Therefore the only way you can do this is to "manually" add them in the INVITE caught by local_route. Best regards, Răzvan On 1/11/19 11:48 AM, Sasmita Panda wrote: >    My initial INVITE looks like below : > >     Request-Line: INVITE sip:6001 at p2p.i3clogic.com:5507 >  SIP/2.0 >     Message Header >         Via: SIP/2.0/UDP > 180.151.95.154:52075;rport;branch=z9hG4bKPjc71c77196989435d8a66406b2f3911e9 >         Max-Forwards: 70 >         From: > sip:webuser at p2p.i3clogic.com:5507;tag=f4e76f6568f74697b55a865842214362 >         To: sip:6001 at p2p.i3clogic.com:5507 > >         Contact: > >         Call-ID: 7fa9909d3769498380c33e52758eca16 >         CSeq: 11538 INVITE >         X-Proxy: false >         Allow: SUBSCRIBE, NOTIFY, REFER, MESSAGE >         Supported: replaces, timer >         Session-Expires: 3600 >         Min-SE: 90 >         User-Agent: WebAstra > X-Info: normal;;A=sip3;C=140;R=4 >         Content-Type: application/sdp >         Content-Length:   225 >     Message Body > > > Below is the Invite opesips generate for recording server : > >  Request-Line: INVITE sip:x.x.x.x:5060 SIP/2.0 >     Message Header >         Via: SIP/2.0/UDP x.x.x.y:5507;branch=z9hG4bK5ad5.d0081b53.0 >         To: sip:x.x.x.x:5060 >         From: ;tag=204a7ab5593cac3dd64ab293b65e4314-ce21 >         CSeq: 2 INVITE >         Call-ID: B2B.430.6291654.1547195241 >         Max-Forwards: 70 >         Content-Length: 1379 >         User-Agent: OpenSIPS (2.4.3 (x86_64/linux)) >         Require: siprec >         Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42 >         Contact: >     Message Body > > Even though I have mentioned the customer_headers parameter still its > not adding . What must be the error ? Is there any issue with siprec > module of opensips-2.4.3 ? > > Please do help me . > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Fri, Jan 11, 2019 at 3:16 PM Sasmita Panda > wrote: > > My initial INVITE looks like below : > >     Request-Line: INVITE sip:6001 at p2p.i3clogic.com:5507 > SIP/2.0 >     Message Header >         Via: SIP/2.0/UDP > 180.151.95.154:52075;rport;branch=z9hG4bKPjc71c77196989435d8a66406b2f3911e9 >         Max-Forwards: 70 >         From: > sip:webuser at p2p.i3clogic.com:5507;tag=f4e76f6568f74697b55a865842214362 >         To: sip:6001 at p2p.i3clogic.com:5507 > >         Contact: > >         Call-ID: 7fa9909d3769498380c33e52758eca16 >         CSeq: 11538 INVITE >         X-Proxy: false >         Allow: SUBSCRIBE, NOTIFY, REFER, MESSAGE >         Supported: replaces, timer >         Session-Expires: 3600 >         Min-SE: 90 >         User-Agent: WebAstra >         X-Info: normal;;A=sip3;C=140;R=4 >         Content-Type: application/sdp >         Content-Length:   225 >     Message Body > > > Below is the Invite opesips generate for recording server : > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Fri, Jan 11, 2019 at 1:42 PM Sasmita Panda > wrote: > > X-Info header is coming in the initial Invite , still its not > adding . > > Even I am not getting anything in the logs too. > > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Fri, Jan 11, 2019 at 1:38 PM Răzvan Crainea > > wrote: > > Hello! > > Is the X-Info header received in the initial INVITE, or is > it added by > your script? In case it is received, it should be added if > you set it in > the custom_headers parameters. If you add it  yourself in > the script, it > won't be added. > Nevertheless, you can modify the INVITE generated by b2b in > local_route > - you can add your header there. Hope this helps! > > Best regards, > Razvan > > On 1/11/19 10:03 AM, Sasmita Panda wrote: > > Hi All, > > > > version: opensips 2.4.3 (x86_64/linux) > > > > I am using siprec module with orex (3rd party software ) . > > I wanted to send 1 customize header of main dialog to the > dialog getting > > forwarded to recording server . > > > > As I know , opensips acts as a b2b UA , and generate its > own Invite for > > recording server in this case . Its only sending the > default headers and > > wont send any information of the initial dialog . > > > > I have a header X-Info . I wanted to pass this . I have > tried the below > > thing . > > > > loadmodule "b2b_entities.so" > > loadmodule "b2b_logic.so" > > modparam("b2b_logic", "custom_headers", "X-Info") > > > > I have tried to send custom header through b2b_logic , > but that wont > > work . It wont add the header . What should I do for this ? > > > > > > > > > > > > */ Thanks & Regards/* > > /Sasmita Panda/ > > /Senior Network Testing and Software Engineer/ > > /3CLogic , ph:07827611765/ > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > Meet the OpenSIPS team at the next OpenSIPS Summit: > https://www.opensips.org/events > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From spanda at 3clogic.com Fri Jan 11 06:44:50 2019 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 11 Jan 2019 17:14:50 +0530 Subject: [OpenSIPS-Users] I need some help in b2b module . In-Reply-To: <40529502-6f6f-c3e5-c519-500c45a720c8@opensips.org> References: <40529502-6f6f-c3e5-c519-500c45a720c8@opensips.org> Message-ID: Thank You. *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Jan 11, 2019 at 3:23 PM Răzvan Crainea wrote: > Hi, Sasmita! > > You are right, I've checked now and unfortunately in this case the > siprec module does not send the request headers to B2B. Therefore the > only way you can do this is to "manually" add them in the INVITE caught > by local_route. > > Best regards, > Răzvan > > On 1/11/19 11:48 AM, Sasmita Panda wrote: > > My initial INVITE looks like below : > > > > Request-Line: INVITE sip:6001 at p2p.i3clogic.com:5507 > > SIP/2.0 > > Message Header > > Via: SIP/2.0/UDP > > 180.151.95.154:52075 > ;rport;branch=z9hG4bKPjc71c77196989435d8a66406b2f3911e9 > > Max-Forwards: 70 > > From: > > sip:webuser at p2p.i3clogic.com:5507;tag=f4e76f6568f74697b55a865842214362 > > To: sip:6001 at p2p.i3clogic.com:5507 > > > > Contact: > > > > Call-ID: 7fa9909d3769498380c33e52758eca16 > > CSeq: 11538 INVITE > > X-Proxy: false > > Allow: SUBSCRIBE, NOTIFY, REFER, MESSAGE > > Supported: replaces, timer > > Session-Expires: 3600 > > Min-SE: 90 > > User-Agent: WebAstra > > X-Info: normal;;A=sip3;C=140;R=4 > > Content-Type: application/sdp > > Content-Length: 225 > > Message Body > > > > > > Below is the Invite opesips generate for recording server : > > > > Request-Line: INVITE sip:x.x.x.x:5060 SIP/2.0 > > Message Header > > Via: SIP/2.0/UDP x.x.x.y:5507;branch=z9hG4bK5ad5.d0081b53.0 > > To: sip:x.x.x.x:5060 > > From: > ;tag=204a7ab5593cac3dd64ab293b65e4314-ce21 > > CSeq: 2 INVITE > > Call-ID: B2B.430.6291654.1547195241 > > Max-Forwards: 70 > > Content-Length: 1379 > > User-Agent: OpenSIPS (2.4.3 (x86_64/linux)) > > Require: siprec > > Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42 > > Contact: > > Message Body > > > > Even though I have mentioned the customer_headers parameter still its > > not adding . What must be the error ? Is there any issue with siprec > > module of opensips-2.4.3 ? > > > > Please do help me . > > > > */Thanks & Regards/* > > /Sasmita Panda/ > > /Senior Network Testing and Software Engineer/ > > /3CLogic , ph:07827611765/ > > > > > > On Fri, Jan 11, 2019 at 3:16 PM Sasmita Panda > > wrote: > > > > My initial INVITE looks like below : > > > > Request-Line: INVITE sip:6001 at p2p.i3clogic.com:5507 > > SIP/2.0 > > Message Header > > Via: SIP/2.0/UDP > > 180.151.95.154:52075 > ;rport;branch=z9hG4bKPjc71c77196989435d8a66406b2f3911e9 > > Max-Forwards: 70 > > From: > > sip:webuser at p2p.i3clogic.com > :5507;tag=f4e76f6568f74697b55a865842214362 > > To: sip:6001 at p2p.i3clogic.com:5507 > > > > Contact: > > > > Call-ID: 7fa9909d3769498380c33e52758eca16 > > CSeq: 11538 INVITE > > X-Proxy: false > > Allow: SUBSCRIBE, NOTIFY, REFER, MESSAGE > > Supported: replaces, timer > > Session-Expires: 3600 > > Min-SE: 90 > > User-Agent: WebAstra > > X-Info: normal;;A=sip3;C=140;R=4 > > Content-Type: application/sdp > > Content-Length: 225 > > Message Body > > > > > > Below is the Invite opesips generate for recording server : > > > > */Thanks & Regards/* > > /Sasmita Panda/ > > /Senior Network Testing and Software Engineer/ > > /3CLogic , ph:07827611765/ > > > > > > On Fri, Jan 11, 2019 at 1:42 PM Sasmita Panda > > wrote: > > > > X-Info header is coming in the initial Invite , still its not > > adding . > > > > Even I am not getting anything in the logs too. > > > > > > */Thanks & Regards/* > > /Sasmita Panda/ > > /Senior Network Testing and Software Engineer/ > > /3CLogic , ph:07827611765/ > > > > > > On Fri, Jan 11, 2019 at 1:38 PM Răzvan Crainea > > > wrote: > > > > Hello! > > > > Is the X-Info header received in the initial INVITE, or is > > it added by > > your script? In case it is received, it should be added if > > you set it in > > the custom_headers parameters. If you add it yourself in > > the script, it > > won't be added. > > Nevertheless, you can modify the INVITE generated by b2b in > > local_route > > - you can add your header there. Hope this helps! > > > > Best regards, > > Razvan > > > > On 1/11/19 10:03 AM, Sasmita Panda wrote: > > > Hi All, > > > > > > version: opensips 2.4.3 (x86_64/linux) > > > > > > I am using siprec module with orex (3rd party software ) . > > > I wanted to send 1 customize header of main dialog to the > > dialog getting > > > forwarded to recording server . > > > > > > As I know , opensips acts as a b2b UA , and generate its > > own Invite for > > > recording server in this case . Its only sending the > > default headers and > > > wont send any information of the initial dialog . > > > > > > I have a header X-Info . I wanted to pass this . I have > > tried the below > > > thing . > > > > > > loadmodule "b2b_entities.so" > > > loadmodule "b2b_logic.so" > > > modparam("b2b_logic", "custom_headers", "X-Info") > > > > > > I have tried to send custom header through b2b_logic , > > but that wont > > > work . It wont add the header . What should I do for this > ? > > > > > > > > > > > > > > > > > > */ Thanks & Regards/* > > > /Sasmita Panda/ > > > /Senior Network Testing and Software Engineer/ > > > /3CLogic , ph:07827611765/ > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.com > > Meet the OpenSIPS team at the next OpenSIPS Summit: > > https://www.opensips.org/events > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > Meet the OpenSIPS team at the next OpenSIPS Summit: > https://www.opensips.org/events > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gr.sabery at gmail.com Fri Jan 11 10:15:53 2019 From: gr.sabery at gmail.com (Gholamreza Sabery) Date: Fri, 11 Jan 2019 18:45:53 +0330 Subject: [OpenSIPS-Users] Using sip_trace increases the number of failed_dialogs Message-ID: Hi, Recently, I tried to integrate OpenSips with Homer5 and it was successful (using HEPv2). However, when I use sip_trace; the number of "failed_dialogs" (which is a statistics exported by dialog module) increases dramatically. Notice that this increase, does not have a negative effect on the real number of successful calls (CDR statistics). I wonder why this happens. Any ideas? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From cc3283 at att.com Mon Jan 14 13:07:51 2019 From: cc3283 at att.com (CARTWRIGHT, CORY C) Date: Mon, 14 Jan 2019 18:07:51 +0000 Subject: [OpenSIPS-Users] Escaping URL parameter values Message-ID: Hello, I have a need to escape URL parameters in the Contact header received from an application. Here is an example: Contact: I believe according to the RFC I need to escape the " and any spaces? I would like to do that within opensips based on certain conditions. I welcome any thoughts or suggestions. Thanks, C -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Tue Jan 15 07:14:21 2019 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 15 Jan 2019 17:44:21 +0530 Subject: [OpenSIPS-Users] It seems there is memory issue in opensips-2.4.3 Message-ID: Hi , I have 2 instances , 1 having openisps 2.2 another 2.4.3 . I am simply registering a sip client and taking call on that . In 2.2 everything works fine without any error . But in 2.4 its not able to process any request . Its sending 500 internal server error . memory allocation is same in both cases . NOTICE:core:main: version: opensips 2.2.4 (x86_64/linux) INFO:core:main: using 32 Mb shared memory INFO:core:main: using 2 Mb private memory per process NOTICE:core:main: version: opensips 2.4.3 (x86_64/linux) INFO:core:main: using 32 Mb of shared memory INFO:core:main: using 2 Mb of private process memory Error in 2.4 : INFO:core:fm_malloc: attempting defragmentation... INFO:core:fm_malloc: unable to alloc a big enough fragment! ERROR:tm:relay_reply: no more share memory ERROR:core:fm_malloc: not enough free shm memory (0 bytes left, need 7160), please increase the "-m" command line parameter! This is the default memory defined in config.h . I have not changed anything . I have 4GB RAM with 2 CPU . I can increase the memory . But i am curious to know if its working on 2.2 then why its not working in 2.4 . Can anybody help me on this please ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Tue Jan 15 22:06:37 2019 From: pasandev at ymail.com (Pasan Meemaduma) Date: Wed, 16 Jan 2019 03:06:37 +0000 (UTC) Subject: [OpenSIPS-Users] sip capturing using hep References: <1188396276.205073.1547607997540.ref@mail.yahoo.com> Message-ID: <1188396276.205073.1547607997540@mail.yahoo.com> Hi Guys, I have following setup for homer integration with opensips. I'm having an issue where opensips complains with following error time to time. I tried tuning up buffer sizes, but it doesn't seems to help. Any suggestion on how to get rid of this ? I'm using opensips 2.3.6 /usr/sbin/opensips[27139]: ERROR:proto_hep:hep_udp_send: sendto(sock,0x7fc2facaff58,5792,0,0x7fc2fa84d858,16): Resource temporarily unavailable(11) sip traffic  ->   internet facing interface (public ip) (opensips)                        internal admin interface (private ip ) udp:9060  ---- >   udp:9060 internal admin interface (private ip)  (opensips sip capturer) When the above error pops out it appears all legit connections get drop too for a brief period. I tried to increate send/recev buffer sizes, but that didn't help. net.core.rmem_maxnet.core.rmem_default net.core.wmem_maxnet.core.wmem_default -------------- next part -------------- An HTML attachment was scrubbed... URL: From wilhelm.lundgren at gmail.com Wed Jan 16 11:03:19 2019 From: wilhelm.lundgren at gmail.com (Wilhelm Lundgren) Date: Wed, 16 Jan 2019 17:03:19 +0100 Subject: [OpenSIPS-Users] sips: automatically downgraded to sip: Message-ID: Hi, Im trying out some tls / sips things in my client. But im seeing something strange in opensips. I am using opensips 2.2.2, x64. Im registering my client with TLS and SIPS. Another client is registering over UDP. I expected that when i send a SIP MESSAGE with sips: in request and to / from tags etc, the UDP client should not receive the message. However, opensips downgrades this message to normal sip and sends it. Have i missunderstood "sips:" feature or have i configed something wrong or is it not supported? Best Regards Wilhelm -------------- next part -------------- An HTML attachment was scrubbed... URL: From mohitsachan762 at gmail.com Thu Jan 17 00:39:31 2019 From: mohitsachan762 at gmail.com (Mohit Sachan) Date: Thu, 17 Jan 2019 11:09:31 +0530 Subject: [OpenSIPS-Users] Query regarding opensips cluster module . In-Reply-To: <18a514bb-a2ab-61c8-2e5f-58d247b2a1ca@opensips.org> References: <18a514bb-a2ab-61c8-2e5f-58d247b2a1ca@opensips.org> Message-ID: hi Could you help me to setup the opensips cluster replication in opensips-2.4. thanks. On Thu, Jan 3, 2019 at 11:05 PM Vlad Patrascu wrote: > Hi, > > You should issue the 'clusterer_reload' MI command on all running nodes > after adding a new one in the DB and no restart is necessary. > > Regards, > > Vlad Patrascu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 12/28/2018 09:56 AM, Sasmita Panda wrote: > > version: opensips 2.2.4 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > svn revision: 3247:3606M > main.c compiled on 09:50:32 Nov 12 2018 with gcc 4.8.3 > > > This is the entire version . Can anyone let me know , whether I am in the > latest version or I need to update my code ?? > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Thu, Dec 27, 2018 at 6:33 PM Sasmita Panda wrote: > >> Hi All, >> >> I am using below version of opensips . >> version: opensips 2.2.4 >> >> I have configured cluster module and I have 2 node in a single cluster . >> Lets say I am adding another node into the same cluster . Each time I >> have to update the config file with the new node information or add the new >> entry in the DB and restart the process . >> >> I don't want to disturb the running setup while adding new node . Is >> there a way through which I can do it in runtime . >> >> for example : when I am updating something in dynamic routing , I am >> adding that through command line and reload the db to have the updated >> information . I am not force to restart the process . >> >> Can cluster module also work in the same way ? There is no option to add >> the DB entry through command line also . Is that's also a limitation in >> cluster module ? >> >> Please do help me . I am not getting anything regarding this in the >> module documentation . >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mohitsachan762 at gmail.com Thu Jan 17 00:47:33 2019 From: mohitsachan762 at gmail.com (Mohit Sachan) Date: Thu, 17 Jan 2019 11:17:33 +0530 Subject: [OpenSIPS-Users] I deleted an entry from clusterer table , but still opensips try to ping that node . In-Reply-To: References: Message-ID: hi Can you suggest me how to install oversip for webrtc for opensips-2.4 in centos. On Thu, Jan 3, 2019 at 10:56 PM Vlad Patrascu wrote: > Hi Sasmita, > > By default, there is no clusterer replication if "replicate_contacts_to" > parameter is not set in usrloc. Also, even if another node is sending > replication packets, they will no get processed on the receiving node > unless "accept_replicated_contacts" is set. On a typical setup, both these > parameter should be set on all nodes. > > Are you getting any other errors in the logs besides that "parameter not > found" ? > > Btw, I strongly suggest updating to 2.4 as it has received major upgrades > in terms of clustering. > > Regards, > > Vlad Patrascu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 01/02/2019 12:10 PM, Sasmita Panda wrote: > > Hi, > I have another doubt . Please do help me . > > When I am reading usrloc module document , its saying in a cluster if we > want to replicate the contacts across the cluster then we have to set a > parameter as below . > > modparam("usrloc", "replicate_contacts_to", 1) > > The default value is 0 , where no cluster id is mentioned . > > I have not set this , I have a cluster having 2 node . While I am registering a user , the contact is getting replicated between 2 nodes . > > If I am trying to mention this parameter , then opensips is not getting started . Its saying > > *Parameter not found in module * > > *So , my question is , if this parameter is not set , still how contact replication is happening ? Is this the default behavior of cluster module ? * > > *May be my question is foolish ,it will be great if anybody will explain this . * > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Wed, Jan 2, 2019 at 12:22 PM Sasmita Panda wrote: > >> Hi Sammy, >> >> Yes , you are right . I need to reload the cluster data through MI >> command . After reloading its seems fine . >> >> I was not aware about the fact that the cluster data also get shared with >> all nodes when I am adding that in 1 node only . >> >> Thank you for your explanation . Its really helpful . >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> On Mon, Dec 31, 2018 at 10:25 PM SamyGo wrote: >> >>> Hi, >>> Did you restart OpenSIPS process on both node1, and 2 simultaneously ? >>> The way I look at this is one of the two nodes kept the 3rd one in the >>> memory and restarting both nodes one at a time resulted in both sharing >>> their node structure and hence node3 stayed visible. >>> I think possible way to remove a node gracefully would be to disable the >>> node via the MI command and then remove from DB. I will try doing this on >>> my test setup as well. >>> >>> Regards, >>> Sammy >>> >>> >>> On Fri, Dec 28, 2018 at 6:40 AM Sasmita Panda >>> wrote: >>> >>>> Hi All, >>>> >>>> I have a cluster of 2 nodes . Both in working condition . Then I >>>> added another node in the same cluster which is down . >>>> >>>> I restarted the opensips process , so it starts pinging the new node to >>>> check its status . As the new node is down , other nodes in the cluster >>>> wont get any reply for the ping . Then I remove the 3rd node from the >>>> cluster table and restart the opensips process . >>>> >>>> Now what I am getting in logs is , still the 2 working node in the >>>> cluster try to ping the 3rd node which is not in the DB . >>>> >>>> Is this an issue on the cluster module or I am doing something wrong ?? >>>> Please help me . >>>> *Thanks & Regards* >>>> *Sasmita Panda* >>>> *Senior Network Testing and Software Engineer* >>>> *3CLogic , ph:07827611765* >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalmpai at gmail.com Thu Jan 17 02:15:27 2019 From: vishalmpai at gmail.com (Vishal Pai) Date: Thu, 17 Jan 2019 12:45:27 +0530 Subject: [OpenSIPS-Users] Opensips caching Message-ID: Hello everyone what is the best way to implement the caching in opensips. I am doing following we have a rate table with NPANXX with rate per minute. Every time when we do outbound calls we get NPANXX of dialed number and search it in database. Since in this way it will create a load on cpu for each select query. Can anyone help me in this also let me know if there is any caching functionality is available in opensips ? Vishal Pai -------------- next part -------------- An HTML attachment was scrubbed... URL: From khamlichi.khalil at gmail.com Thu Jan 17 02:31:45 2019 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Thu, 17 Jan 2019 08:31:45 +0100 Subject: [OpenSIPS-Users] Opensips caching In-Reply-To: References: Message-ID: Are you using dr_rules table ? On Thu, Jan 17, 2019, 8:18 AM Vishal Pai Hello everyone > > what is the best way to implement the caching in opensips. I am doing > following > > we have a rate table with NPANXX with rate per minute. Every time when we > do outbound calls we get NPANXX of dialed number and search it in database. > Since in this way it will create a load on cpu for each select query. > > Can anyone help me in this also let me know if there is any caching > functionality is available in opensips ? > > > Vishal Pai > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosenberg11219 at gmail.com Thu Jan 17 02:40:25 2019 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Thu, 17 Jan 2019 09:40:25 +0200 Subject: [OpenSIPS-Users] Opensips caching In-Reply-To: References: Message-ID: In the script you have multiple caching options, cache_store and cache_fetch from cachedb_mamcached work without external servers like cachedb_mongo that requires mongodb or other nosql modules. On Thu, Jan 17, 2019, 9:18 AM Vishal Pai Hello everyone > > what is the best way to implement the caching in opensips. I am doing > following > > we have a rate table with NPANXX with rate per minute. Every time when we > do outbound calls we get NPANXX of dialed number and search it in database. > Since in this way it will create a load on cpu for each select query. > > Can anyone help me in this also let me know if there is any caching > functionality is available in opensips ? > > > Vishal Pai > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosenberg11219 at gmail.com Thu Jan 17 02:43:38 2019 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Thu, 17 Jan 2019 09:43:38 +0200 Subject: [OpenSIPS-Users] Opensips caching In-Reply-To: References: Message-ID: I believe that dynamic routing runs a SQL query every time, I don't think that can be cached, but if you do your own logic in opensips, you should be able to use cashedb_memcached On Thu, Jan 17, 2019, 9:40 AM Schneur Rosenberg In the script you have multiple caching options, cache_store and > cache_fetch from cachedb_mamcached work without external servers like > cachedb_mongo that requires mongodb or other nosql modules. > > On Thu, Jan 17, 2019, 9:18 AM Vishal Pai >> Hello everyone >> >> what is the best way to implement the caching in opensips. I am doing >> following >> >> we have a rate table with NPANXX with rate per minute. Every time when we >> do outbound calls we get NPANXX of dialed number and search it in database. >> Since in this way it will create a load on cpu for each select query. >> >> Can anyone help me in this also let me know if there is any caching >> functionality is available in opensips ? >> >> >> Vishal Pai >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosenberg11219 at gmail.com Thu Jan 17 02:53:20 2019 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Thu, 17 Jan 2019 09:53:20 +0200 Subject: [OpenSIPS-Users] Query regarding opensips cluster module . In-Reply-To: References: <18a514bb-a2ab-61c8-2e5f-58d247b2a1ca@opensips.org> Message-ID: It's not a cookie cutter pattern, there are different types of clustering and how they sync and how they catch up after a restart, you might also need to take care of who sends notifies for presence etc, there is also the option of anycast which is a great option if your hardware and ISP supports it but more flags need to be set to avoid lots of network noise, the online docs are your best bet. On Thu, Jan 17, 2019, 7:42 AM Mohit Sachan hi > Could you help me to setup the opensips cluster replication in > opensips-2.4. > thanks. > > > > > On Thu, Jan 3, 2019 at 11:05 PM Vlad Patrascu wrote: > >> Hi, >> >> You should issue the 'clusterer_reload' MI command on all running nodes >> after adding a new one in the DB and no restart is necessary. >> >> Regards, >> >> Vlad Patrascu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> On 12/28/2018 09:56 AM, Sasmita Panda wrote: >> >> version: opensips 2.2.4 (x86_64/linux) >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535 >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> svn revision: 3247:3606M >> main.c compiled on 09:50:32 Nov 12 2018 with gcc 4.8.3 >> >> >> This is the entire version . Can anyone let me know , whether I am in the >> latest version or I need to update my code ?? >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> On Thu, Dec 27, 2018 at 6:33 PM Sasmita Panda wrote: >> >>> Hi All, >>> >>> I am using below version of opensips . >>> version: opensips 2.2.4 >>> >>> I have configured cluster module and I have 2 node in a single cluster . >>> Lets say I am adding another node into the same cluster . Each time I >>> have to update the config file with the new node information or add the new >>> entry in the DB and restart the process . >>> >>> I don't want to disturb the running setup while adding new node . Is >>> there a way through which I can do it in runtime . >>> >>> for example : when I am updating something in dynamic routing , I am >>> adding that through command line and reload the db to have the updated >>> information . I am not force to restart the process . >>> >>> Can cluster module also work in the same way ? There is no option to add >>> the DB entry through command line also . Is that's also a limitation in >>> cluster module ? >>> >>> Please do help me . I am not getting anything regarding this in the >>> module documentation . >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Senior Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From hamid2kviii at hotmail.com Thu Jan 17 05:48:52 2019 From: hamid2kviii at hotmail.com (Hamid Hashmi) Date: Thu, 17 Jan 2019 10:48:52 +0000 Subject: [OpenSIPS-Users] Cause of tcp_receive_timeout Message-ID: I am trying to load test TLS listener through sipp. REGISTER Request is working on TLS but INVITE is not working. I changed the transport to TCP and receive following logs version: opensips 2.4.3 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. git revision: c49ae1d53 main.c compiled on 14:23:04 Dec 5 2018 with gcc 4.8.5 Jan 17 13:10:23 siplb_19 192.168.3.19[29333]: DBG:core:tcp_read_req: Accepted connection from 192.168.3.60:36157 on interface 192.168.3.60:2381! Jan 17 13:10:23 siplb_19 192.168.3.19[29333]: DBG:core:tcp_read_req: Using the global ( per process ) buff Jan 17 13:10:23 siplb_19 192.168.3.19[29333]: DBG:core:tcp_handle_req: We didn't manage to read a full request Jan 17 13:10:23 siplb_19 192.168.3.19[29333]: DBG:core:tcp_read_req: tcp_read_req end Jan 17 13:10:27 siplb_19 192.168.3.19[29333]: DBG:core:tcp_receive_timeout: 0x7fe984baf740 expired - (18, 18) lt=129 Jan 17 13:10:27 siplb_19 192.168.3.19[29333]: DBG:core:io_watch_del: [TCP_worker] io_watch_del op on index -1 7 (0x8c2da0, 7, -1, 0x10,0x1) fd_no=5 called Jan 17 13:10:27 siplb_19 192.168.3.19[29333]: DBG:core:tcpconn_release: releasing con 0x7fe984baf740, state -2, fd=-1, id=1475809186 Jan 17 13:10:27 siplb_19 192.168.3.19[29333]: DBG:core:tcpconn_release: extra_data (nil) Jan 17 13:10:27 siplb_19 192.168.3.19[29342]: DBG:core:handle_tcp_worker: response= 7fe984baf740, -2 from tcp worker 29333 (0) Jan 17 13:10:27 siplb_19 192.168.3.19[29342]: DBG:core:tcpconn_destroy: destroying connection 0x7fe98 PCAP file is also attached. SIPP Caller 192.168.3.60 ------- > SIP Server 192.168.3.19 TLS connection working fine with the softphone. This only occurs when I send TLS/TCP request through SIPp script. Regards Hamid R. Hashmi -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: tcp_receive_timeout.pcap Type: application/octet-stream Size: 1615 bytes Desc: tcp_receive_timeout.pcap URL: From kurgan-rus at inbox.ru Thu Jan 17 06:18:50 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Thu, 17 Jan 2019 14:18:50 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?set=5Fadvertised=5Faddress_usage?= Message-ID: <1547723930.282367028@f478.i.mail.ru> Hi list I'm trying to use set_advertised_address(); function in onreply_route but I see that nothing changes. [VoIP ISP] <--- [NAT box x.x.116.2] <--- [OpenSIPS 10.45.144.77] <--- [other VoIP server 10.1.30.12] This is a sip debug, 180 Ringing, leaving OpenSIPS towards Internet: http://rgho.st/private/7lxQHLFgz/c0af9edb35c175a8dba80228f31ba7a2 I think some addresses should be re-written here, but they are not, as we see 10.1.30.12 in some headers. By the way, the next 200 OK with SDP has correct IP address in SDP (as I set it using rtpengine_answer(... media-address=x.x.116.2). So, I'm sure that I use set_advertised_address() in the right place. But why it does not change the message? onreply_route {     set_advertised_address(x.x.116.2);     if (has_body("application/sdp")) {         # rewrite SDP for replies within calls from PBX         if ($fd=="pbx. ... .ru") {             rtpengine_answer("RTP/AVP replace-origin replace-session-connection ICE=remove to-tag");         } else {             # fix external address and rewrite SDP for replies within calls from VoIP ISP         rtpengine_answer("RTP/AVP media-address=x.x.116.2 replace-origin replace-session-connection ICE=remove");         }     } }   ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Thu Jan 17 06:20:36 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Thu, 17 Jan 2019 14:20:36 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?=5BRe=5D=3A_set=5Fadvertised=5Faddress?= =?utf-8?q?_usage?= Message-ID: <1547724036.274916745@f478.i.mail.ru> this link opens a bigger image http://rgho.st/private/7lxQHLFgz/c0af9edb35c175a8dba80228f31ba7a2/image.png   ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From ffshoh at gmail.com Thu Jan 17 06:32:46 2019 From: ffshoh at gmail.com (Jon Abrams) Date: Thu, 17 Jan 2019 05:32:46 -0600 Subject: [OpenSIPS-Users] Opensips caching In-Reply-To: References: Message-ID: There is the sqlcacher module in 2.2+ which will cache whole or parts of sql tables into OpenSIPs memory. There is also the excellent drouting module which caches routing data on startup (or command) into local memory. You could store the rate in the attributes field of the dr_rules table with each NPANXX. Redis and memcached modules are available too. That said, if you don't need hundreds of CPS throughput, you can probably get by with mysql if you make sure you have the table cached in mysql correctly and don't have a lot of other things going on in the database. - Jon On Thu, Jan 17, 2019 at 1:46 AM Schneur Rosenberg wrote: > I believe that dynamic routing runs a SQL query every time, I don't think > that can be cached, but if you do your own logic in opensips, you should be > able to use cashedb_memcached > > On Thu, Jan 17, 2019, 9:40 AM Schneur Rosenberg wrote: > >> In the script you have multiple caching options, cache_store and >> cache_fetch from cachedb_mamcached work without external servers like >> cachedb_mongo that requires mongodb or other nosql modules. >> >> On Thu, Jan 17, 2019, 9:18 AM Vishal Pai > >>> Hello everyone >>> >>> what is the best way to implement the caching in opensips. I am doing >>> following >>> >>> we have a rate table with NPANXX with rate per minute. Every time when >>> we do outbound calls we get NPANXX of dialed number and search it in >>> database. Since in this way it will create a load on cpu for each select >>> query. >>> >>> Can anyone help me in this also let me know if there is any caching >>> functionality is available in opensips ? >>> >>> >>> Vishal Pai >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Thu Jan 17 07:38:20 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Thu, 17 Jan 2019 15:38:20 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?set=5Fadvertised=5Faddress_usage?= Message-ID: <1547728700.267139173@f478.i.mail.ru> Just added some lines to route[relay]. Thanks to Kirill Galinurov. Not ideal in some cases but mostly what I need. route[relay] {   # for INVITEs enable some additional helper routes   if (is_method("INVITE")) {     t_on_branch("per_branch_ops");     t_on_reply("handle_nat");     t_on_failure("missed_call");   }   # fix address in Record-Route if this is a 100/200 reply to Multifon // this is only until OpenSIPS is behind NAT!   if($td=="multifon.ru") {     set_advertised_address("x.X.116.2");   }   if (!t_relay()) {     send_reply("500","Internal Error");   };   exit; } ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Jan 17 08:45:10 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 17 Jan 2019 13:45:10 +0000 Subject: [OpenSIPS-Users] set_advertised_address usage In-Reply-To: <1547723930.282367028@f478.i.mail.ru> References: <1547723930.282367028@f478.i.mail.ru> Message-ID: <0111AE38-D365-46F3-83C0-33275C7FD151@genesys.com> I don’t believe you can do that in a reply route because all the headers are already in the message. No new headers or addresses are being inserted. You need to use set_advertised_address() on the original request so that the addresses are inserted in the request properly. Ben Newlin From: Users on behalf of Alexey Kazantsev via Users Reply-To: Alexey Kazantsev , OpenSIPS users mailling list Date: Thursday, January 17, 2019 at 6:20 AM To: "users at lists.opensips.org" Subject: [OpenSIPS-Users] set_advertised_address usage Hi list I'm trying to use set_advertised_address(); function in onreply_route but I see that nothing changes. [VoIP ISP] <--- [NAT box x.x.116.2] <--- [OpenSIPS 10.45.144.77] <--- [other VoIP server 10.1.30.12] This is a sip debug, 180 Ringing, leaving OpenSIPS towards Internet: http://rgho.st/private/7lxQHLFgz/c0af9edb35c175a8dba80228f31ba7a2 I think some addresses should be re-written here, but they are not, as we see 10.1.30.12 in some headers. By the way, the next 200 OK with SDP has correct IP address in SDP (as I set it using rtpengine_answer(... media-address=x.x.116.2). So, I'm sure that I use set_advertised_address() in the right place. But why it does not change the message? onreply_route { set_advertised_address(x.x.116.2); if (has_body("application/sdp")) { # rewrite SDP for replies within calls from PBX if ($fd=="pbx. ... .ru") { rtpengine_answer("RTP/AVP replace-origin replace-session-connection ICE=remove to-tag"); } else { # fix external address and rewrite SDP for replies within calls from VoIP ISP rtpengine_answer("RTP/AVP media-address=x.x.116.2 replace-origin replace-session-connection ICE=remove"); } } } ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Jan 17 08:49:48 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 17 Jan 2019 13:49:48 +0000 Subject: [OpenSIPS-Users] Opensips caching In-Reply-To: References: Message-ID: No, the dynamic routing module will cache the entire configuration and only read from memory. It must be reloaded when the configuration changes. It’s very quick. Ben Newlin From: Users on behalf of Schneur Rosenberg Reply-To: OpenSIPS users mailling list Date: Thursday, January 17, 2019 at 2:44 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips caching I believe that dynamic routing runs a SQL query every time, I don't think that can be cached, but if you do your own logic in opensips, you should be able to use cashedb_memcached On Thu, Jan 17, 2019, 9:40 AM Schneur Rosenberg wrote: In the script you have multiple caching options, cache_store and cache_fetch from cachedb_mamcached work without external servers like cachedb_mongo that requires mongodb or other nosql modules. On Thu, Jan 17, 2019, 9:18 AM Vishal Pai wrote: Hello everyone what is the best way to implement the caching in opensips. I am doing following we have a rate table with NPANXX with rate per minute. Every time when we do outbound calls we get NPANXX of dialed number and search it in database. Since in this way it will create a load on cpu for each select query. Can anyone help me in this also let me know if there is any caching functionality is available in opensips ? Vishal Pai _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Thu Jan 17 10:19:12 2019 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Thu, 17 Jan 2019 16:19:12 +0100 Subject: [OpenSIPS-Users] Escaping URL parameter values Message-ID: <2C38E227-9DE8-4F02-A1F8-9DA5589F2F89@free.fr> Hi, You can use s.escape.user, https://opensips.org/pipermail/users/2015-September/032545.html Reagrds De : Users au nom de "CARTWRIGHT, CORY C" Répondre à : OpenSIPS users mailling list Date : lundi 14 janvier 2019 à 19:09 À : "users at lists.opensips.org" Objet : [OpenSIPS-Users] Escaping URL parameter values Hello, I have a need to escape URL parameters in the Contact header received from an application. Here is an example: Contact: I believe according to the RFC I need to escape the “ and any spaces? I would like to do that within opensips based on certain conditions. I welcome any thoughts or suggestions. Thanks, C _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From cc3283 at att.com Thu Jan 17 13:32:28 2019 From: cc3283 at att.com (CARTWRIGHT, CORY C) Date: Thu, 17 Jan 2019 18:32:28 +0000 Subject: [OpenSIPS-Users] Escaping URL parameter values In-Reply-To: <2C38E227-9DE8-4F02-A1F8-9DA5589F2F89@free.fr> References: <2C38E227-9DE8-4F02-A1F8-9DA5589F2F89@free.fr> Message-ID: Thank you! From: Users On Behalf Of Alain Bieuzent Sent: Thursday, January 17, 2019 10:19 To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Escaping URL parameter values Hi, You can use s.escape.user, https://opensips.org/pipermail/users/2015-September/032545.html Reagrds De : Users > au nom de "CARTWRIGHT, CORY C" > Répondre à : OpenSIPS users mailling list > Date : lundi 14 janvier 2019 à 19:09 À : "users at lists.opensips.org" > Objet : [OpenSIPS-Users] Escaping URL parameter values Hello, I have a need to escape URL parameters in the Contact header received from an application. Here is an example: Contact: > I believe according to the RFC I need to escape the “ and any spaces? I would like to do that within opensips based on certain conditions. I welcome any thoughts or suggestions. Thanks, C _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From jskorczynski at milosolutions.com Fri Jan 18 08:41:56 2019 From: jskorczynski at milosolutions.com (=?UTF-8?B?SmFuIFNrw7NyY3p5xYRza2k=?=) Date: Fri, 18 Jan 2019 14:41:56 +0100 Subject: [OpenSIPS-Users] [TLS] TLS read error 5 on hanging up call and TLS decrypt error on new connection. Message-ID: Hello there, I'm running opensips 2.4 server with tls support (but without cert verification). For SIP clients I use pjsiplib 2.8. When user ends call with pjsua_call_hangup() server throws this error: opensips[14258]: ERROR:proto_tls:_tls_read: SYSCALL error -> (104) opensips[14258]: ERROR:proto_tls:_tls_read: TLS connection to 185.63.109.74:44828 read failed opensips[14258]: ERROR:proto_tls:_tls_read: TLS read error: 5 opensips[14258]: ERROR:proto_tls:tls_read_req: failed to read Second issue occurs when someone wants to connect to the sip server. It throws: opensips[12482]: ERROR:proto_tls:tls_accept: New TLS connection from 213.205.230.197:42038 failed to accept opensips[12482]: ERROR:proto_tls:tls_print_errstack: TLS errstack: error:1409441B:SSL routines:ssl3_read_bytes:tlsv1 alert decrypt error opensips[12482]: ERROR:proto_tls:tls_read_req: failed to do pre-tls reading Above happens from time to time, but there was a situation when clients cannot establish any connection with opensips till I restart server. Server logs was filled with those errors. Could you help me with those issues? Maybe my tls configuration is wrong? Any ideas? I'm pasting my configuration. I replaced my server ip and port with My.Server.Ip and MY.PORT respectively. ####### Global Parameters ######### log_level=1 log_stderror=no log_facility=LOG_LOCAL0 children=4 /* uncomment the following lines to enable debugging */ #debug_mode=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* comment the next line to enable the auto discovery of local aliases based on reverse DNS on IPs */ auto_aliases=no listen=udp:My.Server.IP:MY.PORT listen=tls:My.Server.IP:MY.PORT ####### Modules Section ######## #set module path mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/" #### STUN server loadmodule "stun.so" modparam("stun", "primary_ip", "My.Server.IP") modparam("stun", "primary_port", "MY.PORT") modparam("stun", "alternate_ip", "My.Server.IP") modparam("stun", "alternate_port", "MY.PORT") #### SIGNALING module loadmodule "signaling.so" #### StateLess module loadmodule "sl.so" #### Transaction Module loadmodule "tm.so" modparam("tm", "fr_timeout", 30) modparam("tm", "fr_inv_timeout", 600) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "onreply_avp_mode", 1) #### Dialog module loadmodule "dialog.so" modparam("dialog", "enable_stats", 0) #### Record Route Module loadmodule "rr.so" /* do not append from tag to the RR (no need for this script) */ modparam("rr", "append_fromtag", 0) #### MAX ForWarD module loadmodule "maxfwd.so" #### SIP MSG OPerationS module loadmodule "sipmsgops.so" #### FIFO Management Interface loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") modparam("mi_fifo", "fifo_mode", 0666) #### URI module loadmodule "uri.so" modparam("uri", "use_uri_table", 0) #### USeR LOCation module loadmodule "usrloc.so" modparam("usrloc", "nat_bflag", "NAT") modparam("usrloc", "working_mode_preset", "single-instance-no-db") #### NAT HELPER #### loadmodule "nathelper.so" modparam("nathelper", "sipping_bflag", "SIPPING_ENABLE") modparam("nathelper", "remove_on_timeout_bflag", "SIPPING_RTO") modparam("nathelper", "natping_tcp", 1) #### REGISTRAR module loadmodule "registrar.so" modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT") /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) #### ACCounting module loadmodule "acc.so" /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_cancels", 0) /* by default we do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) ### MediaProxy loadmodule "mediaproxy.so" modparam("mediaproxy", "disable", 0) modparam("mediaproxy", "ice_candidate", "low-priority") modparam("mediaproxy", "ice_candidate_avp", "$avp(ice_candidate)") loadmodule "proto_udp.so" ### TLS MODULE #loadmodule "proto_hep.so" loadmodule "proto_tls.so" loadmodule "tls_mgm.so" #set global tls parameters modparam("tls_mgm", "verify_cert", "0") modparam("tls_mgm", "require_cert", "0") # modparam("tls_mgm", "tls_method", "TLSv1") modparam("tls_mgm", "certificate", "/root/tls_cnf/tls/rootCA/cacert.pem") modparam("tls_mgm", "private_key", "/root/tls_cnf/tls/rootCA/private/cakey.pem") modparam("tls_mgm", "ca_list", "/root/tls_cnf/tls/rootCA/cacert.pem") modparam("tls_mgm", "ca_dir", "/root/tls_cnf/tls/rootCA/") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { send_reply("483","Too Many Hops"); exit; } if (has_totag()) { # handle hop-by-hop ACK (no routing required) if ( is_method("ACK") && t_check_trans() ) { t_relay(); exit; } # sequential request within a dialog should # take the path determined by record-routing if ( !loose_route() ) { # we do record-routing for all our traffic, so we should not # receive any sequential requests without Route hdr. send_reply("404","Not here"); exit; } if (is_method("BYE")) { # do accounting even if the transaction fails #do_accounting("log","failed"); } # route it out to whatever destination was set by loose_route() # in $du (destination URI). route(relay); exit; } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # absorb retransmissions, but do not create transaction t_check_trans(); if ( !(is_method("REGISTER") ) ) { if (is_myself("$fd")) { } else { # if caller is not local, then called number must be local if (!is_myself("$rd")) { send_reply("403","Relay Forbidden"); exit; } } } # preloaded route checking if (loose_route()) { xlog("L_ERR", "Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]"); if (!is_method("ACK")) send_reply("403","Preload Route denied"); exit; } # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { create_dialog(); engage_media_proxy(); #do_accounting("log"); } if (!is_myself("$rd")) { append_hf("P-hint: outbound\r\n"); route(relay); } # requests for my domain if (is_method("PUBLISH|SUBSCRIBE")) { send_reply("503", "Service Unavailable"); exit; } if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI send_reply("484","Address Incomplete"); exit; } # do lookup with method filtering if (!lookup("location","m")) { t_reply("404", "Not Found"); exit; } # when routing via usrloc, log the missed calls also do_accounting("log","missed"); route(relay); } route[relay] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("per_branch_ops"); t_on_reply("handle_nat"); t_on_failure("missed_call"); } if (!t_relay()) { send_reply("500","Internal Error"); } exit; } branch_route[per_branch_ops] { xlog("new branch at $ru\n"); } onreply_route[handle_nat] { xlog("incoming reply\n"); } failure_route[missed_call] { if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} } -------------- next part -------------- An HTML attachment was scrubbed... URL: From mark at sowegatel.com Fri Jan 18 17:01:02 2019 From: mark at sowegatel.com (Mark Thomas) Date: Fri, 18 Jan 2019 17:01:02 -0500 Subject: [OpenSIPS-Users] Cross compile OpenWRT Message-ID: <5c424c9e.1c69fb81.e1451.8b40@mx.google.com> I’m attempting a cross compile to OpenWRT and I’m getting mipsel-openwrt-linux-ld: cfg.tab.c:(.text+0x12a18): undefined reference to `syslog' I really want to get it to run on OpenWRT as the mid-registrar module is doing a phenomenal job at achieving what I’m wanting to achieve, but I really want it to run on a router. -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Jan 21 05:20:03 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 21 Jan 2019 12:20:03 +0200 Subject: [OpenSIPS-Users] Escaping URL parameter values In-Reply-To: References: Message-ID: <57c08110-b0f0-ed9e-3ca1-1035038f440e@opensips.org> You can use the {s.escape.param} transformation to escape a value[1]. [1] http://www.opensips.org/Documentation/Script-Tran-2-4#toc13 Best regards, Răzvan On 1/14/19 8:07 PM, CARTWRIGHT, CORY C wrote: > Hello, > I have a need to escape URL parameters in the Contact header received > from an application. > Here is an example: > Contact: > <_sip:__55555555555__ at __proxy1__;rn=+1__5555555550__;ocn=__1234__;__carrier="__local > cl__ec__";cat="WIRELESS";__lata=123;npdi_ > clec";cat="WIRELESS";lata=123;npdi>> > I believe according to the RFC I need to escape the “ and any spaces?  I > would like to do that within opensips based on certain conditions.  I > welcome any thoughts or suggestions. > Thanks, > C > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From razvan at opensips.org Mon Jan 21 05:20:50 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 21 Jan 2019 12:20:50 +0200 Subject: [OpenSIPS-Users] It seems there is memory issue in opensips-2.4.3 In-Reply-To: References: Message-ID: <21d5623a-def7-cfb1-006b-1e83cf60f6ff@opensips.org> Please troubleshoot the memory increased issue according to this tutorial: http://www.opensips.org/Documentation/TroubleShooting-OutOfMem Best regards, Răzvan On 1/15/19 2:14 PM, Sasmita Panda wrote: > Hi , > > I have 2 instances , 1 having openisps 2.2 another 2.4.3 . I am simply > registering a sip client and taking call on that . In 2.2 everything > works fine without any error . But in 2.4 its not able to process any > request . Its sending 500 internal server error . memory allocation is > same in both cases . > >  NOTICE:core:main: version: opensips 2.2.4 (x86_64/linux) >  INFO:core:main: using 32 Mb shared memory >  INFO:core:main: using 2 Mb private memory per process > >   NOTICE:core:main: version: opensips 2.4.3 (x86_64/linux) >  INFO:core:main: using 32 Mb of shared memory >  INFO:core:main: using 2 Mb of private process memory > > Error in 2.4 : >  INFO:core:fm_malloc: attempting defragmentation... >  INFO:core:fm_malloc: unable to alloc a big enough fragment! >  ERROR:tm:relay_reply: no more share memory >  ERROR:core:fm_malloc: not enough free shm memory (0 bytes left, need > 7160), please increase the "-m" command line parameter! > > This is the default memory defined in  config.h . I have not changed > anything . > > I have 4GB RAM with 2 CPU . I can increase the memory . But i am curious > to know if its working on 2.2 then why its not working in 2.4 . > > Can anybody help me on this  please ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From Ben.Newlin at genesys.com Tue Jan 22 13:56:40 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 22 Jan 2019 18:56:40 +0000 Subject: [OpenSIPS-Users] Opensips caching In-Reply-To: References: Message-ID: <8FBDD2A2-508E-4048-8B76-BCD75B64F56A@genesys.com> Hi, Since upgrading to 2.4.4 we are seeing the following logs scrolling nearly continuously on our servers: ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:439 It seems to be related to our use of the json module. We often pass json variable types as parameters to other routes and I believe the errors are caused by that. But it’s hard to say as there are a few different script lines referenced in the errors, but some of them point to return statements and other code sections that don’t really make sense. For example, line 583 referenced in the error above is: return(-1); Any ideas? Ben Newlin From: Users on behalf of Ben Newlin Reply-To: OpenSIPS users mailling list Date: Thursday, January 17, 2019 at 8:51 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips caching No, the dynamic routing module will cache the entire configuration and only read from memory. It must be reloaded when the configuration changes. It’s very quick. Ben Newlin From: Users on behalf of Schneur Rosenberg Reply-To: OpenSIPS users mailling list Date: Thursday, January 17, 2019 at 2:44 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips caching I believe that dynamic routing runs a SQL query every time, I don't think that can be cached, but if you do your own logic in opensips, you should be able to use cashedb_memcached On Thu, Jan 17, 2019, 9:40 AM Schneur Rosenberg wrote: In the script you have multiple caching options, cache_store and cache_fetch from cachedb_mamcached work without external servers like cachedb_mongo that requires mongodb or other nosql modules. On Thu, Jan 17, 2019, 9:18 AM Vishal Pai wrote: Hello everyone what is the best way to implement the caching in opensips. I am doing following we have a rate table with NPANXX with rate per minute. Every time when we do outbound calls we get NPANXX of dialed number and search it in database. Since in this way it will create a load on cpu for each select query. Can anyone help me in this also let me know if there is any caching functionality is available in opensips ? Vishal Pai _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Jan 22 13:58:14 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 22 Jan 2019 18:58:14 +0000 Subject: [OpenSIPS-Users] Opensips caching In-Reply-To: <8FBDD2A2-508E-4048-8B76-BCD75B64F56A@genesys.com> References: <8FBDD2A2-508E-4048-8B76-BCD75B64F56A@genesys.com> Message-ID: <883F6552-9BA0-417E-8C21-EF4A9C540AE1@genesys.com> Apologies, that shouldn’t have been added to this thread. Will post again separately. Ben Newlin From: Users on behalf of Ben Newlin Reply-To: OpenSIPS users mailling list Date: Tuesday, January 22, 2019 at 1:57 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips caching Hi, Since upgrading to 2.4.4 we are seeing the following logs scrolling nearly continuously on our servers: ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:439 It seems to be related to our use of the json module. We often pass json variable types as parameters to other routes and I believe the errors are caused by that. But it’s hard to say as there are a few different script lines referenced in the errors, but some of them point to return statements and other code sections that don’t really make sense. For example, line 583 referenced in the error above is: return(-1); Any ideas? Ben Newlin From: Users on behalf of Ben Newlin Reply-To: OpenSIPS users mailling list Date: Thursday, January 17, 2019 at 8:51 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips caching No, the dynamic routing module will cache the entire configuration and only read from memory. It must be reloaded when the configuration changes. It’s very quick. Ben Newlin From: Users on behalf of Schneur Rosenberg Reply-To: OpenSIPS users mailling list Date: Thursday, January 17, 2019 at 2:44 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips caching I believe that dynamic routing runs a SQL query every time, I don't think that can be cached, but if you do your own logic in opensips, you should be able to use cashedb_memcached On Thu, Jan 17, 2019, 9:40 AM Schneur Rosenberg wrote: In the script you have multiple caching options, cache_store and cache_fetch from cachedb_mamcached work without external servers like cachedb_mongo that requires mongodb or other nosql modules. On Thu, Jan 17, 2019, 9:18 AM Vishal Pai wrote: Hello everyone what is the best way to implement the caching in opensips. I am doing following we have a rate table with NPANXX with rate per minute. Every time when we do outbound calls we get NPANXX of dialed number and search it in database. Since in this way it will create a load on cpu for each select query. Can anyone help me in this also let me know if there is any caching functionality is available in opensips ? Vishal Pai _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Jan 22 13:58:49 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 22 Jan 2019 18:58:49 +0000 Subject: [OpenSIPS-Users] Invalid parameter errors Message-ID: Hi, Since upgrading to 2.4.4 we are seeing the following logs scrolling nearly continuously on our servers: ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:439 It seems to be related to our use of the json module. We often pass json variable types as parameters to other routes and I believe the errors are caused by that. But it’s hard to say as there are a few different script lines referenced in the errors, but some of them point to return statements and other code sections that don’t really make sense. For example, line 583 referenced in the error above is: return(-1); Any ideas? Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Jan 22 18:07:01 2019 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 23 Jan 2019 01:07:01 +0200 Subject: [OpenSIPS-Users] Invalid parameter errors In-Reply-To: References: Message-ID: <3290d7c2-2577-d130-3411-702c153bf41a@opensips.org> Hi, Ben! The strange "...type 1836017711" errors seem to be caused by a poorly handed error condition (a secondary bug), which is now fixed [1].  If this theory holds, you must have a "cannot get spec value" error (or slew of errors) in the earlier section of your OpenSIPS log (possibly right after restart or shortly after starting to process traffic). Could you please confirm/infirm the above?  If true, are there any other relevant errors thrown around that initial "cannot get spec value" error message?  Those error logs could be key to making progress in understanding the main bug. Best regards, [1]: https://github.com/OpenSIPS/opensips/commit/52ff74af8702a Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 22.01.2019 20:58, Ben Newlin wrote: > > Hi, > > Since upgrading to 2.4.4 we are seeing the following logs scrolling > nearly continuously on our servers: > > > ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 > > ERROR:core:comp_scriptvar: cannot get left var value > > WARNING:core:do_action: errorin expression at opensips.cfg:583 > ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 > > ERROR:core:comp_scriptvar: cannot get left var value > > WARNING:core:do_action: errorin expression at opensips.cfg:583 > ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 > > ERROR:core:comp_scriptvar: cannot get left var value > > ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 > > ERROR:core:comp_scriptvar: cannot get left var value > > WARNING:core:do_action: errorin expression at opensips.cfg:439 > > It seems to be related to our use of the json module. We often pass > json variable types as parameters to other routes and I believe the > errors are caused by that. But it’s hard to say as there are a few > different script lines referenced in the errors, but some of them > point to return statements and other code sections that don’t really > make sense. For example, line 583 referenced in the error above is: > >   return(-1); > > Any ideas? > > Ben Newlin > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Wed Jan 23 04:05:02 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Wed, 23 Jan 2019 12:05:02 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?string_from_avp_transformation?= Message-ID: <1548234302.212396588@f329.i.mail.ru> Hi list! I have AVPs like "Shop-0 73522222222 3522" or "Office-0 73522222222 3522" I'd like to extract the middle part from them (73522222222 in these examples). Something that can see a whitespace as a delimiter will be great, because the length of words in AVP may differ. I checked the list of core functions [1], but not sure if there is something appropriate there. [1] https://www.opensips.org/Documentation/Script-Tran-2-4 ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ From aqsyounas at gmail.com Wed Jan 23 04:14:45 2019 From: aqsyounas at gmail.com (Aqs Younas) Date: Wed, 23 Jan 2019 14:14:45 +0500 Subject: [OpenSIPS-Users] string from avp transformation In-Reply-To: <1548234302.212396588@f329.i.mail.ru> References: <1548234302.212396588@f329.i.mail.ru> Message-ID: I think below function will do the trick. Use space as delimiter. https://www.opensips.org/Documentation/Script-Tran-2-4#toc6 Best Regards, Aqs On Wed, 23 Jan 2019 at 14:07, Alexey Kazantsev via Users < users at lists.opensips.org> wrote: > Hi list! > > I have AVPs like > "Shop-0 73522222222 3522" > or > "Office-0 73522222222 3522" > > I'd like to extract the middle part from them (73522222222 in these > examples). > Something that can see a whitespace as a delimiter > will be great, because the length of words in AVP may differ. > > I checked the list of core functions [1], but not sure if there is > something appropriate there. > > > [1] https://www.opensips.org/Documentation/Script-Tran-2-4 > > ----------------------------------------------- > BR, Alexey > http://alexeyka.zantsev.com/ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Wed Jan 23 04:15:22 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Wed, 23 Jan 2019 12:15:22 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?string_from_avp_transformation?= In-Reply-To: <1548234302.212396588@f329.i.mail.ru> References: <1548234302.212396588@f329.i.mail.ru> Message-ID: <1548234922.906830269@f329.i.mail.ru> This seems to be what I need, but I can not define a separator when it is a whitespace. https://www.opensips.org/Documentation/Script-Tran-2-4#toc6 I tried: $avp(clid){s.substr,1, } $avp(clid){s.substr,1,\ } $avp(clid){s.substr,1,\\ } ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ From kurgan-rus at inbox.ru Wed Jan 23 05:03:58 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Wed, 23 Jan 2019 13:03:58 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?string_from_avp_transformation?= Message-ID: <1548237838.937549872@f329.i.mail.ru> $avp(clid){s.substr,1,' '} and $avp(clid){s.substr,1,\s} don't work either ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Jan 23 06:19:57 2019 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 23 Jan 2019 13:19:57 +0200 Subject: [OpenSIPS-Users] string from avp transformation In-Reply-To: <1548234922.906830269@f329.i.mail.ru> References: <1548234302.212396588@f329.i.mail.ru> <1548234922.906830269@f329.i.mail.ru> Message-ID: <33d177ea-4632-ca2e-e718-966dc81a398b@opensips.org> Hi Alexey, The first syntax is good, however, notice the following code:     $avp(in) = "Shop-0 73522222222 3522";     $avp(in2) = "Office-0  73522222222 3522";     xlog("XXXX: '$(avp(in){s.select,1, })'\n");     xlog("XXXX: '$(avp(in2){s.select,1, })'\n"); ... and its effect: Jan 23 13:15:32 [20385] XXXX: '73522222222' Jan 23 13:15:32 [20385] XXXX: '' So, in short, there isn't a transformation which is equipped to quickly deal with this input.  It looks like you have to implement a "while result not empty, select the next index" logic in order to obtain parameter number X.  (maybe there's a better solution, but I can't come up with it now) Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 23.01.2019 11:15, Alexey Kazantsev via Users wrote: > This seems to be what I need, but I can not define a separator > when it is a whitespace. > > https://www.opensips.org/Documentation/Script-Tran-2-4#toc6 > > I tried: > $avp(clid){s.substr,1, } > $avp(clid){s.substr,1,\ } > $avp(clid){s.substr,1,\\ } > > ----------------------------------------------- > BR, Alexey > http://alexeyka.zantsev.com/ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From kurgan-rus at inbox.ru Wed Jan 23 06:30:18 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Wed, 23 Jan 2019 14:30:18 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?string_from_avp_transformation?= In-Reply-To: <33d177ea-4632-ca2e-e718-966dc81a398b@opensips.org> References: <1548234302.212396588@f329.i.mail.ru> <1548234922.906830269@f329.i.mail.ru> <33d177ea-4632-ca2e-e718-966dc81a398b@opensips.org> Message-ID: <1548243018.954646745@f389.i.mail.ru> Hi Liviu, thank you, this helped!    append_hf("X-DSTPHONE: $(avp(clid){s.select,1, }) \r\n"); By the way, I noticed only now, that I used wrong way to achieve the result - {s.substr...} instead of {s.select...} So, now it's OK, thanks to everybody. ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From osas at voipembedded.com Wed Jan 23 18:23:45 2019 From: osas at voipembedded.com (Ovidiu Sas) Date: Wed, 23 Jan 2019 18:23:45 -0500 Subject: [OpenSIPS-Users] Cross compile OpenWRT In-Reply-To: <5c424c9e.1c69fb81.e1451.8b40@mx.google.com> References: <5c424c9e.1c69fb81.e1451.8b40@mx.google.com> Message-ID: Opensips was cross compiling fine for optware up to version 1.10. You need to check that your build environment is sane. There's no reference to syslog in cfg.tab.c. Inspect the content of the generated cfg.tab.c file and see why you have a reference to syslog. Regards, Ovidiu Sas On Fri, Jan 18, 2019 at 5:01 PM Mark Thomas wrote: > > I’m attempting a cross compile to OpenWRT and I’m getting > > mipsel-openwrt-linux-ld: cfg.tab.c:(.text+0x12a18): undefined reference to `syslog' > > > > I really want to get it to run on OpenWRT as the mid-registrar module is doing a phenomenal job at achieving what I’m wanting to achieve, but I really want it to run on a router. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- VoIP Embedded, Inc. http://www.voipembedded.com From Ben.Newlin at genesys.com Wed Jan 23 18:37:40 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 23 Jan 2019 23:37:40 +0000 Subject: [OpenSIPS-Users] Invalid parameter errors In-Reply-To: <3290d7c2-2577-d130-3411-702c153bf41a@opensips.org> References: <3290d7c2-2577-d130-3411-702c153bf41a@opensips.org> Message-ID: Liviu, Thank you for the quick response. I do see 2 such errors shortly after startup: ERROR:core:pv_get_param: cannot get spec value ERROR:core:pv_get_param: cannot get spec value However, after that it just continues on with more of the same errors that keep scrolling. There is a variation of the scrolling errors that was I didn’t included before, in case it helps: ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ERROR:json:expand_tag_list: Non string value in key ERROR:json:pv_set_json: Cannot expand variables in path ERROR:core:do_assign: setting PV failed ERROR:core:do_assign: error at opensips.cfg:346 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 There aren’t any other non-repeating errors. I have picked up your commit and will try to gather more information from it, but this issue is primarily happening in our production environment so it may take a bit. Also, I haven’t completely verified this yet, but it seems that enabling TLS has made the errors stop somehow. Continuing to investigate that. Ben Newlin From: Users on behalf of Liviu Chircu Reply-To: OpenSIPS users mailling list Date: Tuesday, January 22, 2019 at 6:08 PM To: "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] Invalid parameter errors Hi, Ben! The strange "...type 1836017711" errors seem to be caused by a poorly handed error condition (a secondary bug), which is now fixed [1]. If this theory holds, you must have a "cannot get spec value" error (or slew of errors) in the earlier section of your OpenSIPS log (possibly right after restart or shortly after starting to process traffic). Could you please confirm/infirm the above? If true, are there any other relevant errors thrown around that initial "cannot get spec value" error message? Those error logs could be key to making progress in understanding the main bug. Best regards, [1]: https://github.com/OpenSIPS/opensips/commit/52ff74af8702a Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 22.01.2019 20:58, Ben Newlin wrote: Hi, Since upgrading to 2.4.4 we are seeing the following logs scrolling nearly continuously on our servers: ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:439 It seems to be related to our use of the json module. We often pass json variable types as parameters to other routes and I believe the errors are caused by that. But it’s hard to say as there are a few different script lines referenced in the errors, but some of them point to return statements and other code sections that don’t really make sense. For example, line 583 referenced in the error above is: return(-1); Any ideas? Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Thu Jan 24 05:10:10 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Thu, 24 Jan 2019 13:10:10 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?shmem_statistics?= Message-ID: <1548324610.261055421@f338.i.mail.ru> Hi list! Where can I read more about what all this means? I'd like to monitor some statistics with Zabbix and create graphs, so I'd like to understand what parameters from these it's necessary to monitor. shmem:total_size:: 67108864 shmem:used_size:: 2858880 shmem:real_used_size:: 2904352 shmem:max_used_size:: 2919144 shmem:free_size:: 64204512 shmem:fragments:: 478 The same question about pkmem: group (no example due to too long listing). ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ From Ben.Newlin at genesys.com Thu Jan 24 08:54:59 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 24 Jan 2019 13:54:59 +0000 Subject: [OpenSIPS-Users] shmem statistics In-Reply-To: <1548324610.261055421@f338.i.mail.ru> References: <1548324610.261055421@f338.i.mail.ru> Message-ID: <68DA2169-7595-4B9D-8DFE-7618E13A0CD3@genesys.com> Alexey, The statistics exported by the OpenSIPS core are described in the documentation here: http://www.opensips.org/Documentation/Interface-CoreStatistics-2-4 Ben Newlin On 1/24/19, 5:10 AM, "Users on behalf of Alexey Kazantsev via Users" wrote: Hi list! Where can I read more about what all this means? I'd like to monitor some statistics with Zabbix and create graphs, so I'd like to understand what parameters from these it's necessary to monitor. shmem:total_size:: 67108864 shmem:used_size:: 2858880 shmem:real_used_size:: 2904352 shmem:max_used_size:: 2919144 shmem:free_size:: 64204512 shmem:fragments:: 478 The same question about pkmem: group (no example due to too long listing). ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Thu Jan 24 11:37:27 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 24 Jan 2019 18:37:27 +0200 Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages Message-ID: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> Hi, Everyone! Check out the latest OpenSIPS module, proto_smpp[1], that you can use to create a two-way bridge between SIP and SMPP text messages. Read more about this on our blog[2]. [1] https://opensips.org/html/docs/modules/3.0.x/proto_smpp.html [2] https://blog.opensips.org/2019/01/24/gateway-between-sip-and-smpp-messages/ Cheers, -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com From volga629 at networklab.ca Thu Jan 24 11:56:48 2019 From: volga629 at networklab.ca (Slava Bendersky) Date: Thu, 24 Jan 2019 11:56:48 -0500 (EST) Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages In-Reply-To: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> References: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> Message-ID: <2033822742.71373.1548349008818.JavaMail.zimbra@skillsearch.ca> Great, Thank you. From: "Răzvan Crainea" To: "OpenSIPS users mailling list" , "OpenSIPS devel mailling list" Sent: Thursday, January 24, 2019 12:37:27 PM Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages Hi, Everyone! Check out the latest OpenSIPS module, proto_smpp[1], that you can use to create a two-way bridge between SIP and SMPP text messages. Read more about this on our blog[2]. [1] https://opensips.org/html/docs/modules/3.0.x/proto_smpp.html [2] https://blog.opensips.org/2019/01/24/gateway-between-sip-and-smpp-messages/ Cheers, -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Thu Jan 24 12:21:29 2019 From: johan at democon.be (johan de clercq) Date: Thu, 24 Jan 2019 18:21:29 +0100 Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages In-Reply-To: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> References: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> Message-ID: <012f01d4b409$3ec9b9a0$bc5d2ce0$@democon.be> Do you backport this to 2.4 ? BR, -----Original Message----- From: Users On Behalf Of Razvan Crainea Sent: Thursday, January 24, 2019 5:37 PM To: OpenSIPS users mailling list ; OpenSIPS devel mailling list Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages Hi, Everyone! Check out the latest OpenSIPS module, proto_smpp[1], that you can use to create a two-way bridge between SIP and SMPP text messages. Read more about this on our blog[2]. [1] https://opensips.org/html/docs/modules/3.0.x/proto_smpp.html [2] https://blog.opensips.org/2019/01/24/gateway-between-sip-and-smpp-messages/ Cheers, -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From james at fivecats.org Thu Jan 24 12:38:45 2019 From: james at fivecats.org (James Sharp) Date: Thu, 24 Jan 2019 09:38:45 -0800 Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages In-Reply-To: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> References: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> Message-ID: <19C44470-2737-40B7-817A-87A9DAD1C061@fivecats.org> > Hi, Everyone! > > Check out the latest OpenSIPS module, proto_smpp[1], that you can use to create a two-way bridge between SIP and SMPP text messages. Read more about this on our blog[2]. This is awesome. I can get rid of an ugly message routing hack now. From liviu at opensips.org Thu Jan 24 15:10:31 2019 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 24 Jan 2019 22:10:31 +0200 Subject: [OpenSIPS-Users] Invalid parameter errors In-Reply-To: References: <3290d7c2-2577-d130-3411-702c153bf41a@opensips.org> Message-ID: Hi Ben, We are actually dealing with two bugs here, which may or may not be related to one another. Bug #1: bad? variable during a route() call ------------------------------------------------------- For this one, can you enumerate all "route()" calls in your script which pass at least one variable, along with their full parameter call syntax?  Example call: route(sequential_requests, $rm, $avp(myinfo)); Bug #2: bad "key variable" during a $json expansion ---------------------------------------------------------------------- For this one, can you enumerate all $json() variable appearances which include at least one parameterized key access?  I realize there may be lots of these, but you may group them into "categories" and print out a few ones that might be relevant (i.e. their index may contain an INT-only variable, which is >wrong<).  Example appearances: $json(http_body/$var(tag)) $json(http_body/users[0]/$avp(username)) Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 24.01.2019 01:37, Ben Newlin wrote: > > Liviu, > > Thank you for the quick response. I do see 2 such errors shortly after > startup: > > ERROR:core:pv_get_param: cannot get spec value > > ERROR:core:pv_get_param: cannot get spec value > > However, after that it just continues on with more of the same errors > that keep scrolling. There is a variation of the scrolling errors that > was I didn’t included before, in case it helps: > > ERROR:core:comp_scriptvar: cannot get left var value > > WARNING:core:do_action: errorin expression at opensips.cfg:583 > > ERROR:json:expand_tag_list: Non string value in key > > ERROR:json:pv_set_json: Cannot expand variables in path > > ERROR:core:do_assign: setting PV failed > > ERROR:core:do_assign: errorat opensips.cfg:346 > > ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 > > There aren’t any other non-repeating errors. I have picked up your > commit and will try to gather more information from it, but this issue > is primarily happening in our production environment so it may take a bit. > > Also, I haven’t completely verified this yet, but it seems that > enabling TLS has made the errors stop somehow. Continuing to > investigate that. > > Ben Newlin > > *From: *Users on behalf of Liviu > Chircu > *Reply-To: *OpenSIPS users mailling list > *Date: *Tuesday, January 22, 2019 at 6:08 PM > *To: *"users at lists.opensips.org" > *Subject: *Re: [OpenSIPS-Users] Invalid parameter errors > > Hi, Ben! > > The strange "...type 1836017711" errors seem to be caused by a poorly > handed error condition (a secondary bug), which is now fixed [1]. If > this theory holds, you must have a "cannot get spec value" error (or > slew of errors) in the earlier section of your OpenSIPS log (possibly > right after restart or shortly after starting to process traffic). > > Could you please confirm/infirm the above?  If true, are there any > other relevant errors thrown around that initial "cannot get spec > value" error message?  Those error logs could be key to making > progress in understanding the main bug. > > Best regards, > > [1]: https://github.com/OpenSIPS/opensips/commit/52ff74af8702a > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 22.01.2019 20:58, Ben Newlin wrote: > > Hi, > > Since upgrading to 2.4.4 we are seeing the following logs > scrolling nearly continuously on our servers: > > > > ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 > > ERROR:core:comp_scriptvar: cannot get left var value > > WARNING:core:do_action: errorin expression at opensips.cfg:583 > ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 > > ERROR:core:comp_scriptvar: cannot get left var value > > WARNING:core:do_action: errorin expression at opensips.cfg:583 > ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 > > ERROR:core:comp_scriptvar: cannot get left var value > > ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 > > ERROR:core:comp_scriptvar: cannot get left var value > > WARNING:core:do_action: errorin expression at opensips.cfg:439 > > It seems to be related to our use of the json module. We often > pass json variable types as parameters to other routes and I > believe the errors are caused by that. But it’s hard to say as > there are a few different script lines referenced in the errors, > but some of them point to return statements and other code > sections that don’t really make sense. For example, line 583 > referenced in the error above is: > >   return(-1); > > Any ideas? > > Ben Newlin > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Jan 24 17:14:27 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 24 Jan 2019 22:14:27 +0000 Subject: [OpenSIPS-Users] error_route not triggering Message-ID: Hi, I recently noticed some parsing errors in our logs and after digging further I’ve realized that our error route is not triggering when this occurs. Is there some sort of subscribe or attach operation needed to get calls to the error route? The documentation states it will be called automatically. I’ve been able to reproduce the issue in our testbed. We are running OpenSIPS 2.4.4. My error route is defined like this: error_route { xlog("L_ALERT", "Error route called!\n"); } This is what I get from OpenSIPS logs: Jan 24 21:59:30 [329] ERROR:core:receive_msg: Unable to parse msg received from [203.0.113.4:48096] Jan 24 21:59:30 [336] ERROR:core:parse_first_line: bad request first line Jan 24 21:59:30 [336] ERROR:core:parse_first_line: at line 0 char 17: Jan 24 21:59:30 [336] ERROR:core:parse_first_line: parsed so far: INVITE sip:bad to Jan 24 21:59:30 [336] INFO:core:parse_first_line: bad message Jan 24 21:59:30 [336] ERROR:core:parse_msg: message= From vishalmpai at gmail.com Thu Jan 24 20:17:08 2019 From: vishalmpai at gmail.com (Vishal Pai) Date: Fri, 25 Jan 2019 06:47:08 +0530 Subject: [OpenSIPS-Users] SHM Memory issue Message-ID: Hello everyone I am using opensips version version: opensips 2.4.3 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. git revision: c49ae1d53 I am getting when there is increase the number of calls Jan 24 19:51:20 ns101756 /usr/local/sbin/opensips[1431]: ERROR:core:fm_malloc: not enough free shm memory (59920 bytes left, need 432), please increase the "-m" command line parameter! Jan 24 19:51:20 ns101756 /usr/local/sbin/opensips[1431]: INFO:core:fm_malloc: attempting defragmentation... Jan 24 19:51:20 ns101756 /usr/local/sbin/opensips[1423]: WARNING:core:utimer_ticker: utimer task already scheduled for 30849800 ms (now 31196060 ms), it may overlap.. Jan 24 19:51:20 ns101756 /usr/local/sbin/opensips[1431]: INFO:core:fm_malloc: unable to alloc a big enough fragment! Jan 24 19:51:20 ns101756 /usr/local/sbin/opensips[1431]: ERROR:tm:build_local: no more share memory Jan 24 19:51:20 ns101756 /usr/local/sbin/opensips[1431]: ERROR:tm:cancel_branch: attempt to build a CANCEL failed Think it's memory issue. What is the best way to fix it up. -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Fri Jan 25 00:00:06 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Fri, 25 Jan 2019 08:00:06 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?SHM_Memory_issue?= In-Reply-To: References: Message-ID: <1548392406.387444048@f548.i.mail.ru> Hi Vishal, what's the output of ps aux | grep [o]pensips command? It should show the value of '-m' parameter. ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/ From vishalmpai at gmail.com Fri Jan 25 01:36:50 2019 From: vishalmpai at gmail.com (Vishal Pai) Date: Fri, 25 Jan 2019 12:06:50 +0530 Subject: [OpenSIPS-Users] SHM Memory issue In-Reply-To: <1548392406.387444048@f548.i.mail.ru> References: <1548392406.387444048@f548.i.mail.ru> Message-ID: I am getting this [image: grep.PNG] On Fri, Jan 25, 2019 at 10:33 AM Alexey Kazantsev via Users < users at lists.opensips.org> wrote: > Hi Vishal, > > what's the output of > > ps aux | grep [o]pensips > > command? It should show the value of '-m' parameter. > > > ----------------------------------------------- > BR, Alexey > http://alexeyka.zantsev.com/ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: grep.PNG Type: image/png Size: 5589 bytes Desc: not available URL: From razvan at opensips.org Fri Jan 25 03:34:38 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 25 Jan 2019 10:34:38 +0200 Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages In-Reply-To: <012f01d4b409$3ec9b9a0$bc5d2ce0$@democon.be> References: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> <012f01d4b409$3ec9b9a0$bc5d2ce0$@democon.be> Message-ID: <652258ae-5160-921c-f6bb-555bb196ab8b@opensips.org> Unfortunately this is a feature, not a bug, so it won't be backported to 2.4. Cheers, Răzvan On 1/24/19 7:21 PM, johan de clercq wrote: > Do you backport this to 2.4 ? > > BR, > > -----Original Message----- > From: Users On Behalf Of Razvan Crainea > Sent: Thursday, January 24, 2019 5:37 PM > To: OpenSIPS users mailling list ; OpenSIPS devel mailling list > Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages > > Hi, Everyone! > > Check out the latest OpenSIPS module, proto_smpp[1], that you can use to create a two-way bridge between SIP and SMPP text messages. Read more about this on our blog[2]. > > [1] https://opensips.org/html/docs/modules/3.0.x/proto_smpp.html > [2] > https://blog.opensips.org/2019/01/24/gateway-between-sip-and-smpp-messages/ > > Cheers, > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From vitalik.voip at gmail.com Fri Jan 25 09:43:16 2019 From: vitalik.voip at gmail.com (Vitalii Aleksandrov) Date: Fri, 25 Jan 2019 16:43:16 +0200 Subject: [OpenSIPS-Users] empty reply for db_http Message-ID: Hi, Using db_http and permission modules and can't figure out what exactly should be returned from an HTTP API when 'address' table is empty and SELECT returns no results. Here is what I tried with no luck: 1.     Status:         200 OK     Body:         string,int,int,int,string,string,string,int 2.     Status:         200 OK     Body:         string,int,int,int,string,string,string,int         ,,,,,,, 3.     Status:         200 OK     Body:         empty body 4.     Status:         404 Not Found     Body:         empty db_http documentation says "_If the query is ok (even if the answer is empty) the server must reply with a 200 OK HTTP reply with a body containing the types and values of the columns._". I have no idea what "values" should I return from http API in this corner case. Adding some fake records to a table doesn't look like a good solution. Any input would be greatly appreciated! -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Jan 25 09:51:22 2019 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 25 Jan 2019 14:51:22 +0000 Subject: [OpenSIPS-Users] Dynamic Routing never routes call Message-ID: Hello all Very new OpenSIPS user trying to build my first real server. Using debug logging I can see that my alias/dialplan operations seem to be working but when it reaches the do_routing the call never actually gets routed. I've been trying to get this working for nearly 2 weeks now and I'm at a loss now. Please can someone help me? >From my script: $avp(gw_whitelist) = "testpbx1"; if ( !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) { send_reply("404","DID not found"); xlog("do_routing: No rules matching the URI\n"); exit; Oddly, I never get the no rules log entry but it drops out of here & into the else if below which forces proxy auth. My DB: dr gateways +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ | id | gwid | type | address | strip | pri_prefix | attrs | probe_mode | state | socket | description | +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ | 2 | BT_SDIN_BCTE | 1 | sip:xxx.xxx.xxx.xxx | 0 | | BT_SDIN_BCTE | 2 | 0 | | Inbound | | 3 | BT_SDIN_GDH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | BT_SDIN_GDH | 2 | 0 | | Inbound | | 1 | BT_SDIN_LFH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | BT_SDIN_LFH | 2 | 0 | | Inbound | | 4 | BT_SDIN_SEH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | BT_SDIN_SEH | 2 | 0 | | Inbound | | 5 | testpbx1 | 1 | sip:10.98.0.11 | 2 | 0 | testpbx1 | 2 | 0 | | Inbound | +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ dr groups +----+--------------+----------------------------+---------+-------------+ | id | username | domain | groupid | description | +----+--------------+----------------------------+---------+-------------+ | 7 | 441423369031 | 10.98.0.11 | 1 | | | 6 | 441423369031 | my.domain | 2 | | +----+--------------+----------------------------+---------+-------------+ dr carriers +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ | id | carrierid | gwlist | flags | state | attrs | description | +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ | 1 | BT | BT_SDIN_BCTE,BT_SDIN_GDH,BT_SDIN_LFH,BT_SDIN_SEH | 0 | 0 | | BT SDIN | +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ dr rules +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ | ruleid | groupid | prefix | timerec | priority | routeid | gwlist | attrs | description | +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ | 13 | 1 | 441423369031 | | 0 | | testpbx1 | rule_441423369031 | Send to testpbx1 | +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From jim at devito.cc Fri Jan 25 10:04:56 2019 From: jim at devito.cc (Jim DeVito) Date: Fri, 25 Jan 2019 10:04:56 -0500 Subject: [OpenSIPS-Users] Dynamic Routing never routes call In-Reply-To: References: Message-ID: Are you missing the actual relay part? I'm pretty sure do_routing just loads routing info from the database but you sill need to call the actual relay. Put a t_relay() after the IF statement and see what happens. Also look at the docs regarding capturing and displaying the return code from do_routing. On Fri, Jan 25, 2019 at 9:52 AM Mark Farmer wrote: > Hello all > > Very new OpenSIPS user trying to build my first real server. > > Using debug logging I can see that my alias/dialplan operations seem to be > working but when it reaches the do_routing the call never actually gets > routed. > > I've been trying to get this working for nearly 2 weeks now and I'm at a > loss now. Please can someone help me? > > From my script: > > $avp(gw_whitelist) = "testpbx1"; > if ( > !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) > { > send_reply("404","DID not found"); > xlog("do_routing: No rules matching the URI\n"); > exit; > > Oddly, I never get the no rules log entry but it drops out of here & into > the else if below which forces proxy auth. > > My DB: > > dr gateways > > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > | id | gwid | type | address | strip | pri_prefix | > attrs | probe_mode | state | socket | description | > > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > | 2 | BT_SDIN_BCTE | 1 | sip:xxx.xxx.xxx.xxx | 0 | | > BT_SDIN_BCTE | 2 | 0 | | Inbound | > | 3 | BT_SDIN_GDH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | > BT_SDIN_GDH | 2 | 0 | | Inbound | > | 1 | BT_SDIN_LFH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | > BT_SDIN_LFH | 2 | 0 | | Inbound | > | 4 | BT_SDIN_SEH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | > BT_SDIN_SEH | 2 | 0 | | Inbound | > | 5 | testpbx1 | 1 | sip:10.98.0.11 | 2 | 0 | > testpbx1 | 2 | 0 | | Inbound | > > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > dr groups > +----+--------------+----------------------------+---------+-------------+ > | id | username | domain | groupid | description | > +----+--------------+----------------------------+---------+-------------+ > | 7 | 441423369031 | 10.98.0.11 | 1 | | > | 6 | 441423369031 | my.domain | 2 | | > +----+--------------+----------------------------+---------+-------------+ > dr carriers > > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > | id | carrierid | gwlist | > flags | state | attrs | description | > > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > | 1 | BT | BT_SDIN_BCTE,BT_SDIN_GDH,BT_SDIN_LFH,BT_SDIN_SEH | > 0 | 0 | | BT SDIN | > > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > dr rules > > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > | ruleid | groupid | prefix | timerec | priority | routeid | gwlist > | attrs | description | > > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > | 13 | 1 | 441423369031 | | 0 | | > testpbx1 | rule_441423369031 | Send to testpbx1 | > > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- ------------- Jim DeVito Mobile 216.507.9497 -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Jan 25 10:06:17 2019 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 25 Jan 2019 15:06:17 +0000 Subject: [OpenSIPS-Users] Dynamic Routing never routes call In-Reply-To: References: Message-ID: A little more info from the logs: DBG:db_mysql:db_mysql_str2val: converting STRING [+441423369031] DBG:db_mysql:db_mysql_str2val: converting STRING [10.98.0.11] DBG:alias_db:alias_db_query: new URI [0] is [sip:+441423369031 at 10.98.0.11] DBG:core:db_free_columns: freeing result columns at 0x7f5cf8f9b720 DBG:core:db_free_rows: freeing 1 rows DBG:core:db_free_row: freeing row values at 0x7f5cf8f9b780 DBG:core:db_free_rows: freeing rows at 0x7f5cf8f9b770 DBG:core:db_free_result: freeing result set at 0x7f5cf8f9b6d8 DBG:dialplan:dp_translate_f: dpid is 1 partition is default DBG:dialplan:dp_get_svalue: searching 15 DBG:dialplan:dp_translate_f: input is +441423369031 DBG:dialplan:dp_translate_f: Checking with dpid 1 DBG:dialplan:translate: Regex operator testing. Got result: -1 DBG:dialplan:test_match: test_match:[0] +441423369031 DBG:dialplan:translate: Regex operator testing. Got result: 0 DBG:dialplan:translate: Found a matching rule 0x7f5cf7233908: pr 1, match_exp \+[1-9][0-9]+$ DBG:dialplan:test_match: test_match:[0] +441423369031 DBG:dialplan:test_match: test_match:[1] 441423369031 DBG:dialplan:dp_translate_f: input +441423369031 with dpid 1 => output 441423369031 DBG:drouting:do_routing_1: matching prefix with strict len DBG:drouting:do_routing: using dr group 1, rule_idx 0, username 441423369031 DBG:drouting:internal_check_rt: found rgid 1 (rule list 0x7f5cf72360a8) DBG:drouting:push_gw_for_usage: adding gw [testpbx1] as " sip:01423369031 at 10.98.0.11" in order 0 DBG:drouting:push_gw_for_usage: setting GW id [testpbx1] as avp DBG:drouting:push_gw_for_usage: setting GW attr [testpbx1] as avp DBG:drouting:do_routing: setting RULE attr [rule_441423369031] DBG:core:parse_headers: flags=10000 DBG:auth:pre_auth: credentials with given realm not found Jan 25 15:00:39 tsip3 /usr/local/sbin/opensips[12173]: ----- gw attr is Jan 25 15:00:39 tsip3 /usr/local/sbin/opensips[12173]: ----- ruri is sip:01423369031 at 10.98.0.11 Jan 25 15:00:39 tsip3 /usr/local/sbin/opensips[12173]: proxy: MF - Fell into proxy auth On Fri, 25 Jan 2019 at 14:51, Mark Farmer wrote: > Hello all > > Very new OpenSIPS user trying to build my first real server. > > Using debug logging I can see that my alias/dialplan operations seem to be > working but when it reaches the do_routing the call never actually gets > routed. > > I've been trying to get this working for nearly 2 weeks now and I'm at a > loss now. Please can someone help me? > > From my script: > > $avp(gw_whitelist) = "testpbx1"; > if ( > !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) > { > send_reply("404","DID not found"); > xlog("do_routing: No rules matching the URI\n"); > exit; > > Oddly, I never get the no rules log entry but it drops out of here & into > the else if below which forces proxy auth. > > My DB: > > dr gateways > > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > | id | gwid | type | address | strip | pri_prefix | > attrs | probe_mode | state | socket | description | > > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > | 2 | BT_SDIN_BCTE | 1 | sip:xxx.xxx.xxx.xxx | 0 | | > BT_SDIN_BCTE | 2 | 0 | | Inbound | > | 3 | BT_SDIN_GDH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | > BT_SDIN_GDH | 2 | 0 | | Inbound | > | 1 | BT_SDIN_LFH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | > BT_SDIN_LFH | 2 | 0 | | Inbound | > | 4 | BT_SDIN_SEH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | > BT_SDIN_SEH | 2 | 0 | | Inbound | > | 5 | testpbx1 | 1 | sip:10.98.0.11 | 2 | 0 | > testpbx1 | 2 | 0 | | Inbound | > > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > dr groups > +----+--------------+----------------------------+---------+-------------+ > | id | username | domain | groupid | description | > +----+--------------+----------------------------+---------+-------------+ > | 7 | 441423369031 | 10.98.0.11 | 1 | | > | 6 | 441423369031 | my.domain | 2 | | > +----+--------------+----------------------------+---------+-------------+ > dr carriers > > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > | id | carrierid | gwlist | > flags | state | attrs | description | > > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > | 1 | BT | BT_SDIN_BCTE,BT_SDIN_GDH,BT_SDIN_LFH,BT_SDIN_SEH | > 0 | 0 | | BT SDIN | > > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > dr rules > > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > | ruleid | groupid | prefix | timerec | priority | routeid | gwlist > | attrs | description | > > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > | 13 | 1 | 441423369031 | | 0 | | > testpbx1 | rule_441423369031 | Send to testpbx1 | > > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > > > > -- > Mark Farmer > farmorg at gmail.com > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Jan 25 10:24:48 2019 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 25 Jan 2019 15:24:48 +0000 Subject: [OpenSIPS-Users] Dynamic Routing never routes call In-Reply-To: References: Message-ID: Thanks Jim. I now have: $avp(gw_whitelist) = "testpbx1"; if ( !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) { t_relay(); send_reply("404","DID not found"); xlog("do_routing: No rules matching the URI\n"); exit; Sadly no change. I'll head over to the docs again now. On Fri, 25 Jan 2019 at 15:08, Jim DeVito wrote: > Are you missing the actual relay part? I'm pretty sure do_routing just > loads routing info from the database but you sill need to call the actual > relay. Put a t_relay() after the IF statement and see what happens. Also > look at the docs regarding capturing and displaying the return code from > do_routing. > > > > On Fri, Jan 25, 2019 at 9:52 AM Mark Farmer wrote: > >> Hello all >> >> Very new OpenSIPS user trying to build my first real server. >> >> Using debug logging I can see that my alias/dialplan operations seem to >> be working but when it reaches the do_routing the call never actually gets >> routed. >> >> I've been trying to get this working for nearly 2 weeks now and I'm at a >> loss now. Please can someone help me? >> >> From my script: >> >> $avp(gw_whitelist) = "testpbx1"; >> if ( >> !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) >> { >> send_reply("404","DID not found"); >> xlog("do_routing: No rules matching the URI\n"); >> exit; >> >> Oddly, I never get the no rules log entry but it drops out of here & into >> the else if below which forces proxy auth. >> >> My DB: >> >> dr gateways >> >> +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ >> | id | gwid | type | address | strip | pri_prefix | >> attrs | probe_mode | state | socket | description | >> >> +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ >> | 2 | BT_SDIN_BCTE | 1 | sip:xxx.xxx.xxx.xxx | 0 | | >> BT_SDIN_BCTE | 2 | 0 | | Inbound | >> | 3 | BT_SDIN_GDH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | >> BT_SDIN_GDH | 2 | 0 | | Inbound | >> | 1 | BT_SDIN_LFH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | >> BT_SDIN_LFH | 2 | 0 | | Inbound | >> | 4 | BT_SDIN_SEH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | >> BT_SDIN_SEH | 2 | 0 | | Inbound | >> | 5 | testpbx1 | 1 | sip:10.98.0.11 | 2 | 0 | >> testpbx1 | 2 | 0 | | Inbound | >> >> +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ >> dr groups >> +----+--------------+----------------------------+---------+-------------+ >> | id | username | domain | groupid | description | >> +----+--------------+----------------------------+---------+-------------+ >> | 7 | 441423369031 | 10.98.0.11 | 1 | | >> | 6 | 441423369031 | my.domain | 2 | | >> +----+--------------+----------------------------+---------+-------------+ >> dr carriers >> >> +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ >> | id | carrierid | gwlist | >> flags | state | attrs | description | >> >> +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ >> | 1 | BT | BT_SDIN_BCTE,BT_SDIN_GDH,BT_SDIN_LFH,BT_SDIN_SEH | >> 0 | 0 | | BT SDIN | >> >> +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ >> dr rules >> >> +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ >> | ruleid | groupid | prefix | timerec | priority | routeid | gwlist >> | attrs | description | >> >> +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ >> | 13 | 1 | 441423369031 | | 0 | | >> testpbx1 | rule_441423369031 | Send to testpbx1 | >> >> +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > ------------- > Jim DeVito > Mobile 216.507.9497 > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From jennifer.hashimoto at caztel.com Fri Jan 25 10:29:28 2019 From: jennifer.hashimoto at caztel.com (Jennifer Hashimoto) Date: Fri, 25 Jan 2019 10:29:28 -0500 Subject: [OpenSIPS-Users] Dynamic Routing never routes call In-Reply-To: References: Message-ID: <000A0D78-A504-4B1E-9EAA-4A3B39D903CC@caztel.com> You are only going into the block that you put the relay in if your do_routing fails. pay attention to the ! which means NOT --------------------------------------------------- Jennifer Akemi Hashimoto Caztel Communications jennifer.hashimoto at caztel.com 905-836-5445 > On Jan 25, 2019, at 10:24 AM, Mark Farmer wrote: > > Thanks Jim. > > I now have: > > $avp(gw_whitelist) = "testpbx1"; > if ( !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) { > t_relay(); > send_reply("404","DID not found"); > xlog("do_routing: No rules matching the URI\n"); > exit; > > Sadly no change. > > I'll head over to the docs again now. > > > > On Fri, 25 Jan 2019 at 15:08, Jim DeVito wrote: > Are you missing the actual relay part? I'm pretty sure do_routing just loads routing info from the database but you sill need to call the actual relay. Put a t_relay() after the IF statement and see what happens. Also look at the docs regarding capturing and displaying the return code from do_routing. > > > > On Fri, Jan 25, 2019 at 9:52 AM Mark Farmer > wrote: > Hello all > > Very new OpenSIPS user trying to build my first real server. > > Using debug logging I can see that my alias/dialplan operations seem to be working but when it reaches the do_routing the call never actually gets routed. > > I've been trying to get this working for nearly 2 weeks now and I'm at a loss now. Please can someone help me? > > From my script: > > $avp(gw_whitelist) = "testpbx1"; > if ( !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) { > send_reply("404","DID not found"); > xlog("do_routing: No rules matching the URI\n"); > exit; > > Oddly, I never get the no rules log entry but it drops out of here & into the else if below which forces proxy auth. > > My DB: > > dr gateways > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > | id | gwid | type | address | strip | pri_prefix | attrs | probe_mode | state | socket | description | > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > | 2 | BT_SDIN_BCTE | 1 | sip:xxx.xxx.xxx.xxx | 0 | | BT_SDIN_BCTE | 2 | 0 | | Inbound | > | 3 | BT_SDIN_GDH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | BT_SDIN_GDH | 2 | 0 | | Inbound | > | 1 | BT_SDIN_LFH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | BT_SDIN_LFH | 2 | 0 | | Inbound | > | 4 | BT_SDIN_SEH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | BT_SDIN_SEH | 2 | 0 | | Inbound | > | 5 | testpbx1 | 1 | sip:10.98.0.11 | 2 | 0 | testpbx1 | 2 | 0 | | Inbound | > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > dr groups > +----+--------------+----------------------------+---------+-------------+ > | id | username | domain | groupid | description | > +----+--------------+----------------------------+---------+-------------+ > | 7 | 441423369031 | 10.98.0.11 | 1 | | > | 6 | 441423369031 | my.domain | 2 | | > +----+--------------+----------------------------+---------+-------------+ > dr carriers > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > | id | carrierid | gwlist | flags | state | attrs | description | > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > | 1 | BT | BT_SDIN_BCTE,BT_SDIN_GDH,BT_SDIN_LFH,BT_SDIN_SEH | 0 | 0 | | BT SDIN | > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > dr rules > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > | ruleid | groupid | prefix | timerec | priority | routeid | gwlist | attrs | description | > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > | 13 | 1 | 441423369031 | | 0 | | testpbx1 | rule_441423369031 | Send to testpbx1 | > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > > > > -- > Mark Farmer > farmorg at gmail.com _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- > ------------- > Jim DeVito > Mobile 216.507.9497 > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- > Mark Farmer > farmorg at gmail.com _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From ffshoh at gmail.com Fri Jan 25 10:31:40 2019 From: ffshoh at gmail.com (Jon Abrams) Date: Fri, 25 Jan 2019 09:31:40 -0600 Subject: [OpenSIPS-Users] Dynamic Routing never routes call In-Reply-To: References: Message-ID: You have the t_relay() in the do_routing() failure code section. Move it afterwards: $avp(gw_whitelist) = "testpbx1"; if ( !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) { send_reply("404","DID not found"); xlog("do_routing: No rules matching the URI\n"); exit; } t_relay(); - Jon On Fri, Jan 25, 2019 at 9:27 AM Mark Farmer wrote: > Thanks Jim. > > I now have: > > $avp(gw_whitelist) = "testpbx1"; > if ( > !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) > { > t_relay(); > send_reply("404","DID not found"); > xlog("do_routing: No rules matching the URI\n"); > exit; > > Sadly no change. > > I'll head over to the docs again now. > > > > On Fri, 25 Jan 2019 at 15:08, Jim DeVito wrote: > >> Are you missing the actual relay part? I'm pretty sure do_routing just >> loads routing info from the database but you sill need to call the actual >> relay. Put a t_relay() after the IF statement and see what happens. Also >> look at the docs regarding capturing and displaying the return code from >> do_routing. >> >> >> >> On Fri, Jan 25, 2019 at 9:52 AM Mark Farmer wrote: >> >>> Hello all >>> >>> Very new OpenSIPS user trying to build my first real server. >>> >>> Using debug logging I can see that my alias/dialplan operations seem to >>> be working but when it reaches the do_routing the call never actually gets >>> routed. >>> >>> I've been trying to get this working for nearly 2 weeks now and I'm at a >>> loss now. Please can someone help me? >>> >>> From my script: >>> >>> $avp(gw_whitelist) = "testpbx1"; >>> if ( >>> !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) >>> { >>> send_reply("404","DID not found"); >>> xlog("do_routing: No rules matching the URI\n"); >>> exit; >>> >>> Oddly, I never get the no rules log entry but it drops out of here & >>> into the else if below which forces proxy auth. >>> >>> My DB: >>> >>> dr gateways >>> >>> +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ >>> | id | gwid | type | address | strip | pri_prefix | >>> attrs | probe_mode | state | socket | description | >>> >>> +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ >>> | 2 | BT_SDIN_BCTE | 1 | sip:xxx.xxx.xxx.xxx | 0 | >>> | BT_SDIN_BCTE | 2 | 0 | | Inbound | >>> | 3 | BT_SDIN_GDH | 1 | sip:xxx.xxx.xxx.xxx | 0 | >>> | BT_SDIN_GDH | 2 | 0 | | Inbound | >>> | 1 | BT_SDIN_LFH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | >>> BT_SDIN_LFH | 2 | 0 | | Inbound | >>> | 4 | BT_SDIN_SEH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | >>> BT_SDIN_SEH | 2 | 0 | | Inbound | >>> | 5 | testpbx1 | 1 | sip:10.98.0.11 | 2 | 0 | >>> testpbx1 | 2 | 0 | | Inbound | >>> >>> +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ >>> dr groups >>> >>> +----+--------------+----------------------------+---------+-------------+ >>> | id | username | domain | groupid | description >>> | >>> >>> +----+--------------+----------------------------+---------+-------------+ >>> | 7 | 441423369031 | 10.98.0.11 | 1 | >>> | >>> | 6 | 441423369031 | my.domain | 2 | | >>> >>> +----+--------------+----------------------------+---------+-------------+ >>> dr carriers >>> >>> +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ >>> | id | carrierid | gwlist | >>> flags | state | attrs | description | >>> >>> +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ >>> | 1 | BT | BT_SDIN_BCTE,BT_SDIN_GDH,BT_SDIN_LFH,BT_SDIN_SEH | >>> 0 | 0 | | BT SDIN | >>> >>> +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ >>> dr rules >>> >>> +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ >>> | ruleid | groupid | prefix | timerec | priority | routeid | >>> gwlist | attrs | description | >>> >>> +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ >>> | 13 | 1 | 441423369031 | | 0 | | >>> testpbx1 | rule_441423369031 | Send to testpbx1 | >>> >>> +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ >>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> ------------- >> Jim DeVito >> Mobile 216.507.9497 >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Fri Jan 25 10:46:52 2019 From: volga629 at networklab.ca (Slava Bendersky) Date: Fri, 25 Jan 2019 10:46:52 -0500 (EST) Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages In-Reply-To: <652258ae-5160-921c-f6bb-555bb196ab8b@opensips.org> References: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> <012f01d4b409$3ec9b9a0$bc5d2ce0$@democon.be> <652258ae-5160-921c-f6bb-555bb196ab8b@opensips.org> Message-ID: <135235270.72204.1548431212249.JavaMail.zimbra@skillsearch.ca> Hello Răzvan, proto_smpp missing README in main tree or just wasn't generated ? volga629 From: "Răzvan Crainea" To: "OpenSIPS users mailling list" Sent: Friday, January 25, 2019 4:34:38 AM Subject: Re: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages Unfortunately this is a feature, not a bug, so it won't be backported to 2.4. Cheers, Răzvan On 1/24/19 7:21 PM, johan de clercq wrote: > Do you backport this to 2.4 ? > > BR, > > -----Original Message----- > From: Users On Behalf Of Razvan Crainea > Sent: Thursday, January 24, 2019 5:37 PM > To: OpenSIPS users mailling list ; OpenSIPS devel mailling list > Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages > > Hi, Everyone! > > Check out the latest OpenSIPS module, proto_smpp[1], that you can use to create a two-way bridge between SIP and SMPP text messages. Read more about this on our blog[2]. > > [1] https://opensips.org/html/docs/modules/3.0.x/proto_smpp.html > [2] > https://blog.opensips.org/2019/01/24/gateway-between-sip-and-smpp-messages/ > > Cheers, > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Fri Jan 25 10:48:06 2019 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 25 Jan 2019 17:48:06 +0200 Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages In-Reply-To: <135235270.72204.1548431212249.JavaMail.zimbra@skillsearch.ca> References: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> <012f01d4b409$3ec9b9a0$bc5d2ce0$@democon.be> <652258ae-5160-921c-f6bb-555bb196ab8b@opensips.org> <135235270.72204.1548431212249.JavaMail.zimbra@skillsearch.ca> Message-ID: <6c99641c-279f-6cf1-78cc-f41dcd592263@opensips.org> OpenSIPS GitHub [1] will build it on Sunday :) [1]: https://github.com/opensips-github Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 25.01.2019 17:46, Slava Bendersky wrote: > Hello Răzvan, > > proto_smpp missing README in main tree  or just wasn't generated ? > > volga629 > > ------------------------------------------------------------------------ > *From: *"Răzvan Crainea" > *To: *"OpenSIPS users mailling list" > *Sent: *Friday, January 25, 2019 4:34:38 AM > *Subject: *Re: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP > messages > > Unfortunately this is a feature, not a bug, so it won't be backported to > 2.4. > > Cheers, > Răzvan > > On 1/24/19 7:21 PM, johan de clercq wrote: > > Do you backport this to 2.4 ? > > > > BR, > > > > -----Original Message----- > > From: Users On Behalf Of Razvan > Crainea > > Sent: Thursday, January 24, 2019 5:37 PM > > To: OpenSIPS users mailling list ; > OpenSIPS devel mailling list > > Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages > > > > Hi, Everyone! > > > > Check out the latest OpenSIPS module, proto_smpp[1], that you can > use to create a two-way bridge between SIP and SMPP text messages. > Read more about this on our blog[2]. > > > > [1] https://opensips.org/html/docs/modules/3.0.x/proto_smpp.html > > [2] > > > https://blog.opensips.org/2019/01/24/gateway-between-sip-and-smpp-messages/ > > > > Cheers, > > -- > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.com > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer >    http://www.opensips-solutions.com > Meet the OpenSIPS team at the next OpenSIPS Summit: >    https://www.opensips.org/events > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Jan 25 10:50:00 2019 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 25 Jan 2019 15:50:00 +0000 Subject: [OpenSIPS-Users] Dynamic Routing never routes call In-Reply-To: References: Message-ID: OK, that's changed something! I'm now getting 'do_routing: No rules matching the URI' in my log - so I guess I need to review the rules. Many thanks for the help so far!! On Fri, 25 Jan 2019 at 15:35, Jon Abrams wrote: > You have the t_relay() in the do_routing() failure code section. Move it > afterwards: > > > $avp(gw_whitelist) = "testpbx1"; > if ( > !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) > { > > send_reply("404","DID not found"); > xlog("do_routing: No rules matching the URI\n"); > exit; > } > > t_relay(); > > - Jon > > On Fri, Jan 25, 2019 at 9:27 AM Mark Farmer wrote: > >> Thanks Jim. >> >> I now have: >> >> $avp(gw_whitelist) = "testpbx1"; >> if ( >> !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) >> { >> t_relay(); >> send_reply("404","DID not found"); >> xlog("do_routing: No rules matching the URI\n"); >> exit; >> >> Sadly no change. >> >> I'll head over to the docs again now. >> >> >> >> On Fri, 25 Jan 2019 at 15:08, Jim DeVito wrote: >> >>> Are you missing the actual relay part? I'm pretty sure do_routing just >>> loads routing info from the database but you sill need to call the actual >>> relay. Put a t_relay() after the IF statement and see what happens. Also >>> look at the docs regarding capturing and displaying the return code from >>> do_routing. >>> >>> >>> >>> On Fri, Jan 25, 2019 at 9:52 AM Mark Farmer wrote: >>> >>>> Hello all >>>> >>>> Very new OpenSIPS user trying to build my first real server. >>>> >>>> Using debug logging I can see that my alias/dialplan operations seem to >>>> be working but when it reaches the do_routing the call never actually gets >>>> routed. >>>> >>>> I've been trying to get this working for nearly 2 weeks now and I'm at >>>> a loss now. Please can someone help me? >>>> >>>> From my script: >>>> >>>> $avp(gw_whitelist) = "testpbx1"; >>>> if ( >>>> !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) >>>> { >>>> send_reply("404","DID not found"); >>>> xlog("do_routing: No rules matching the URI\n"); >>>> exit; >>>> >>>> Oddly, I never get the no rules log entry but it drops out of here & >>>> into the else if below which forces proxy auth. >>>> >>>> My DB: >>>> >>>> dr gateways >>>> >>>> +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ >>>> | id | gwid | type | address | strip | pri_prefix | >>>> attrs | probe_mode | state | socket | description | >>>> >>>> +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ >>>> | 2 | BT_SDIN_BCTE | 1 | sip:xxx.xxx.xxx.xxx | 0 | >>>> | BT_SDIN_BCTE | 2 | 0 | | Inbound | >>>> | 3 | BT_SDIN_GDH | 1 | sip:xxx.xxx.xxx.xxx | 0 | >>>> | BT_SDIN_GDH | 2 | 0 | | Inbound | >>>> | 1 | BT_SDIN_LFH | 1 | sip:xxx.xxx.xxx.xxx | 0 | | >>>> BT_SDIN_LFH | 2 | 0 | | Inbound | >>>> | 4 | BT_SDIN_SEH | 1 | sip:xxx.xxx.xxx.xxx | 0 | >>>> | BT_SDIN_SEH | 2 | 0 | | Inbound | >>>> | 5 | testpbx1 | 1 | sip:10.98.0.11 | 2 | 0 | >>>> testpbx1 | 2 | 0 | | Inbound | >>>> >>>> +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ >>>> dr groups >>>> >>>> +----+--------------+----------------------------+---------+-------------+ >>>> | id | username | domain | groupid | >>>> description | >>>> >>>> +----+--------------+----------------------------+---------+-------------+ >>>> | 7 | 441423369031 | 10.98.0.11 | 1 | >>>> | >>>> | 6 | 441423369031 | my.domain | 2 | | >>>> >>>> +----+--------------+----------------------------+---------+-------------+ >>>> dr carriers >>>> >>>> +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ >>>> | id | carrierid | gwlist | >>>> flags | state | attrs | description | >>>> >>>> +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ >>>> | 1 | BT | BT_SDIN_BCTE,BT_SDIN_GDH,BT_SDIN_LFH,BT_SDIN_SEH | >>>> 0 | 0 | | BT SDIN | >>>> >>>> +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ >>>> dr rules >>>> >>>> +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ >>>> | ruleid | groupid | prefix | timerec | priority | routeid | >>>> gwlist | attrs | description | >>>> >>>> +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ >>>> | 13 | 1 | 441423369031 | | 0 | | >>>> testpbx1 | rule_441423369031 | Send to testpbx1 | >>>> >>>> +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ >>>> >>>> >>>> >>>> -- >>>> Mark Farmer >>>> farmorg at gmail.com >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> -- >>> ------------- >>> Jim DeVito >>> Mobile 216.507.9497 >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mark at sowegatel.com Fri Jan 25 12:56:54 2019 From: mark at sowegatel.com (Mark Thomas) Date: Fri, 25 Jan 2019 12:56:54 -0500 Subject: [OpenSIPS-Users] RTPPROXY ENGAGEMENT Message-ID: <5c4b4de6.1c69fb81.db8dd.701b@mx.google.com> I have an issue and I don’t know how to resolve it. I’ve got a 486 route that has to engage rtpproxy to bridge to another network. My problem is I have to engage from UAC to UAS on the public side before it sends to voicemail. Whenever I attempt to engage rtpproxy on the leg going to the voicemail servers which are on load balancer it re-writes the sdp twice. I’ve battled with the thing for a while now and could really use some assistance in rectifying this issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: From osas at voipembedded.com Fri Jan 25 13:12:58 2019 From: osas at voipembedded.com (Ovidiu Sas) Date: Fri, 25 Jan 2019 13:12:58 -0500 Subject: [OpenSIPS-Users] RTPPROXY ENGAGEMENT In-Reply-To: <5c4b4de6.1c69fb81.db8dd.701b@mx.google.com> References: <5c4b4de6.1c69fb81.db8dd.701b@mx.google.com> Message-ID: If you are experiencing double sdp re-wites it means that you are engaging rtpproxy more then once. Add some xlogs in your script where you engage rtpproxy and figure out why it is engaged twice. Regards, Ovidiu Sas On Fri, Jan 25, 2019 at 12:57 PM Mark Thomas wrote: > > I have an issue and I don’t know how to resolve it. I’ve got a 486 route that has to engage rtpproxy to bridge to another network. My problem is I have to engage from UAC to UAS on the public side before it sends to voicemail. Whenever I attempt to engage rtpproxy on the leg going to the voicemail servers which are on load balancer it re-writes the sdp twice. I’ve battled with the thing for a while now and could really use some assistance in rectifying this issue. > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- VoIP Embedded, Inc. http://www.voipembedded.com From aqsyounas at gmail.com Fri Jan 25 15:36:58 2019 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 26 Jan 2019 01:36:58 +0500 Subject: [OpenSIPS-Users] Cannot execute MI command on opensips-cp 8.2.4 Message-ID: Greetings list, I am trying to execute mi commands from opensips-cp but looks like opensips-cp is unable to find the json URL and sending me the below error. *16:06:52* | *uptime* | ------------------------------ Unknwon/Unsupported type[] for MI URL <> By looking at the below snippet of mi.php file, I see if ($_GET['action']=="change_box" && !empty($_POST['box_val'])) { $current_box=$_POST['box_val']; $_SESSION['mi_current_box']=$current_box ; } else if (!empty($_SESSION['mi_current_box'])) { $current_box=$_SESSION['mi_current_box']; } else { *$current_box="";* } Somehow variable current_box is being populated as empty. My boxes.global.inc.php is configured like below. $box_id=0; // MI connector (via JSON backend): json:host:port/json $boxes[$box_id]['mi']['conn']="json:127.0.0.1:8888/json"; And i am able to use mi commands from the terminal. root at opensips:~# curl http://127.0.0.1:8888/json/uptime {"Now": "Fri Jan 25 16:32:45 2019", "Up since": "Fri Jan 25 16:21:57 2019", "Up time": "648 [sec]"}root at opensips:~# I am at lost where else i need to define my server json URL. Any help is much appreciated. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Fri Jan 25 15:49:37 2019 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 25 Jan 2019 22:49:37 +0200 Subject: [OpenSIPS-Users] Dynamic Routing never routes call In-Reply-To: References: Message-ID: Hi, Mark! Notice this log: Jan 25 15:00:39 tsip3 /usr/local/sbin/opensips[12173]: ----- ruri is sip:01423369031 at 10.98.0.11 ... and your drouting rule: +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ | ruleid | groupid | prefix       | timerec | priority | routeid | gwlist   | attrs             | description      | +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ |     13 | 1       | 441423369031 |         |        0 |       | testpbx1 | rule_441423369031 | Send to testpbx1 | Although OpenSIPS does a lot of things behind the curtain, detecting and auto-stripping UK prefixes is a job that it does not. Here's an additional tip: to quickly test your prefix matching using a modern OpenSIPS, you can do: opensipsctl fifo dr_number_routing 1 441423369031 In your case, it would have outputted nothing, meaning: "no match found" Cheers, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 25.01.2019 17:06, Mark Farmer wrote: > A little more info from the logs: > > DBG:db_mysql:db_mysql_str2val: converting STRING [+441423369031] > DBG:db_mysql:db_mysql_str2val: converting STRING [10.98.0.11] > DBG:alias_db:alias_db_query: new URI [0] is > [sip:+441423369031 at 10.98.0.11 ] > DBG:core:db_free_columns: freeing result columns at 0x7f5cf8f9b720 > DBG:core:db_free_rows: freeing 1 rows > DBG:core:db_free_row: freeing row values at 0x7f5cf8f9b780 > DBG:core:db_free_rows: freeing rows at 0x7f5cf8f9b770 > DBG:core:db_free_result: freeing result set at 0x7f5cf8f9b6d8 > DBG:dialplan:dp_translate_f: dpid is 1 partition is default > DBG:dialplan:dp_get_svalue: searching 15 > DBG:dialplan:dp_translate_f: input is +441423369031 > DBG:dialplan:dp_translate_f: Checking with dpid 1 > DBG:dialplan:translate: Regex operator testing. Got result: -1 > DBG:dialplan:test_match: test_match:[0] +441423369031 > DBG:dialplan:translate: Regex operator testing. Got result: 0 > DBG:dialplan:translate: Found a matching rule 0x7f5cf7233908: pr 1, > match_exp \+[1-9][0-9]+$ > DBG:dialplan:test_match: test_match:[0] +441423369031 > DBG:dialplan:test_match: test_match:[1] 441423369031 > DBG:dialplan:dp_translate_f: input +441423369031 with dpid 1 => output > 441423369031 > DBG:drouting:do_routing_1: matching prefix with strict len > DBG:drouting:do_routing: using dr group 1, rule_idx 0, username > 441423369031 > DBG:drouting:internal_check_rt: found rgid 1 (rule list 0x7f5cf72360a8) > DBG:drouting:push_gw_for_usage: adding gw [testpbx1] as > "sip:01423369031 at 10.98.0.11 " in > order 0 > DBG:drouting:push_gw_for_usage: setting GW id [testpbx1] as avp > DBG:drouting:push_gw_for_usage: setting GW attr [testpbx1] as avp > DBG:drouting:do_routing: setting RULE attr [rule_441423369031] > DBG:core:parse_headers: flags=10000 > DBG:auth:pre_auth: credentials with given realm not found > Jan 25 15:00:39 tsip3 /usr/local/sbin/opensips[12173]: ----- gw attr > is > Jan 25 15:00:39 tsip3 /usr/local/sbin/opensips[12173]: ----- ruri is > sip:01423369031 at 10.98.0.11 > Jan 25 15:00:39 tsip3 /usr/local/sbin/opensips[12173]: proxy: MF - > Fell into proxy auth > > > > On Fri, 25 Jan 2019 at 14:51, Mark Farmer > wrote: > > Hello all > > Very new OpenSIPS user trying to build my first real server. > > Using debug logging I can see that my alias/dialplan operations > seem to be working but when it reaches the do_routing the call > never actually gets routed. > > I've been trying to get this working for nearly 2 weeks now and > I'm at a loss now. Please can someone help me? > > From my script: > >         $avp(gw_whitelist) = "testpbx1"; >         if ( > !do_routing("1","L","$avp(gw_whitelist)","$avp(rules_attributes)","$avp(gw_attributes)")) > { >           send_reply("404","DID not found"); >           xlog("do_routing: No rules matching the URI\n"); >           exit; > > Oddly, I never get the no rules log entry but it drops out of here > & into the else if below which forces proxy auth. > > My DB: > > dr gateways > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > | id | gwid         | type | address             | strip | > pri_prefix | attrs        | probe_mode | state | socket | > description | > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > |  2 | BT_SDIN_BCTE |    1 | sip:xxx.xxx.xxx.xxx   |     0 |       >  | BT_SDIN_BCTE |          2 |     0 |        | Inbound     | > |  3 | BT_SDIN_GDH  |    1 | sip:xxx.xxx.xxx.xxx   |     0 |       >  | BT_SDIN_GDH  |          2 |     0 |        | Inbound     | > |  1 | BT_SDIN_LFH  |    1 | sip:xxx.xxx.xxx.xxx |     0 |      | > BT_SDIN_LFH  |          2 |   0 |        | Inbound     | > |  4 | BT_SDIN_SEH  |    1 | sip:xxx.xxx.xxx.xxx  |     0 |      | > BT_SDIN_SEH  |          2 |   0 |        | Inbound     | > |  5 | testpbx1     |    1 | sip:10.98.0.11      |     2 | 0     >  | testpbx1     |          2 |   0 |        | Inbound     | > +----+--------------+------+---------------------+-------+------------+--------------+------------+-------+--------+-------------+ > dr groups > +----+--------------+----------------------------+---------+-------------+ > | id | username     | domain                 | groupid | description | > +----+--------------+----------------------------+---------+-------------+ > |  7 | 441423369031 | 10.98.0.11                 |       1 |     | > |  6 | 441423369031 | my.domain |       2 |             | > +----+--------------+----------------------------+---------+-------------+ > dr carriers > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > | id | carrierid | gwlist                                   | > flags | state | attrs | description | > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > |  1 | BT        | > BT_SDIN_BCTE,BT_SDIN_GDH,BT_SDIN_LFH,BT_SDIN_SEH |     0 |     0 | >       | BT SDIN   | > +----+-----------+--------------------------------------------------+-------+-------+-------+-------------+ > dr rules > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > | ruleid | groupid | prefix | timerec | priority | routeid | > gwlist   | attrs             | description      | > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > |     13 | 1       | 441423369031 |         |        0 |         | > testpbx1 | rule_441423369031 | Send to testpbx1 | > +--------+---------+--------------+---------+----------+---------+----------+-------------------+------------------+ > > > > -- > Mark Farmer > farmorg at gmail.com > > > > -- > Mark Farmer > farmorg at gmail.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Sat Jan 26 05:05:30 2019 From: john.quick at smartvox.co.uk (John Quick) Date: Sat, 26 Jan 2019 10:05:30 -0000 Subject: [OpenSIPS-Users] Dynamic Routing never routes call Message-ID: <000601d4b55e$ab7398e0$025acaa0$@smartvox.co.uk> Hi Mark, Liviu made a mistake in his last response to you. He picked out the debug at the end of the drouting process, not the beginning. Perhaps he had the screen upside down. >From the debug it is clear that do_routing behaved correctly and has worked. So the problem is either that you have misunderstood what the do_routing() function is meant to do OR you have a mistake in the code in your script just after the snippet that you posted to the forum. do_routing() will only alter the R-URI. It does not then send the request to the destination and exit. You must do this explicitly, for example: t_relay() exit As a general point for anyone using OpenSIPS, it is worth noting that just about every function will return. The only exception I can think of right now is t_check_trans() and even that will sometimes return - it depends on specific conditions. If you want the main route in your script to stop executing after a function has been called, you need to insert an exit statement. Also note that each type of route has a default behaviour on exit as described in the documentation under the heading "Types of routes" John Quick Smartvox Limited Web: www.smartvox.co.uk From mickael at winlux.fr Sun Jan 27 01:51:44 2019 From: mickael at winlux.fr (Mickael Hubert) Date: Sun, 27 Jan 2019 07:51:44 +0100 Subject: [OpenSIPS-Users] phonenumber format into mysql database In-Reply-To: <669db766-70e1-598e-2820-847fc7818ec8@opensips.org> References: <669db766-70e1-598e-2820-847fc7818ec8@opensips.org> Message-ID: Hi I'm sorry for my late answer. I'm talking about drouting table for example. Le mer. 3 oct. 2018 22:14, Bogdan-Andrei Iancu a écrit : > Hi Mickael, > > AFAIK E164 is without '+' (which is actually a breakout code, not part of > the number itself). > > And when you say the numbers are stored in DB - which module/table are you > talking about ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 10/03/2018 11:47 AM, Mickael Hubert wrote: > > Hi all, > I have a non technical question. > why phonenumbers into database are stocked whitout + sign (non E164 > format) ? > is it for a prformance purposes ? > > I'm looking for the best way to store E164 phonenumbers for a personnal > dev. > > thanks in advance > > ++ > Micka > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From basit.engg at gmail.com Sun Jan 27 19:16:51 2019 From: basit.engg at gmail.com (Abdul Basit) Date: Mon, 28 Jan 2019 05:16:51 +0500 Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages In-Reply-To: <6c99641c-279f-6cf1-78cc-f41dcd592263@opensips.org> References: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> <012f01d4b409$3ec9b9a0$bc5d2ce0$@democon.be> <652258ae-5160-921c-f6bb-555bb196ab8b@opensips.org> <135235270.72204.1548431212249.JavaMail.zimbra@skillsearch.ca> <6c99641c-279f-6cf1-78cc-f41dcd592263@opensips.org> Message-ID: nice addition as this will eliminate the kannel integration logic along side with SIP services. How to handle message max. characters length while transmitting the message over SMPP and concatenation on receiving message from SMSC? Is this implementation address multilingual support, i.e Unicode support? -- regards, abdul basit On Fri, 25 Jan 2019 at 20:51, Liviu Chircu wrote: > OpenSIPS GitHub [1] will build it on Sunday :) > > [1]: https://github.com/opensips-github > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 25.01.2019 17:46, Slava Bendersky wrote: > > Hello Răzvan, > > proto_smpp missing README in main tree or just wasn't generated ? > > volga629 > > ------------------------------ > *From: *"Răzvan Crainea" > *To: *"OpenSIPS users mailling list" > > *Sent: *Friday, January 25, 2019 4:34:38 AM > *Subject: *Re: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP > messages > > Unfortunately this is a feature, not a bug, so it won't be backported to > 2.4. > > Cheers, > Răzvan > > On 1/24/19 7:21 PM, johan de clercq wrote: > > Do you backport this to 2.4 ? > > > > BR, > > > > -----Original Message----- > > From: Users > On Behalf Of Razvan Crainea > > Sent: Thursday, January 24, 2019 5:37 PM > > To: OpenSIPS users mailling list > ; OpenSIPS devel mailling list > > > Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages > > > > Hi, Everyone! > > > > Check out the latest OpenSIPS module, proto_smpp[1], that you can use to > create a two-way bridge between SIP and SMPP text messages. Read more about > this on our blog[2]. > > > > [1] https://opensips.org/html/docs/modules/3.0.x/proto_smpp.html > > [2] > > > https://blog.opensips.org/2019/01/24/gateway-between-sip-and-smpp-messages/ > > > > Cheers, > > -- > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.com > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > Meet the OpenSIPS team at the next OpenSIPS Summit: > https://www.opensips.org/events > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Jan 28 03:24:13 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 28 Jan 2019 10:24:13 +0200 Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP messages In-Reply-To: References: <9f3ba4b1-cd78-f390-debe-32c30d705251@opensips.org> <012f01d4b409$3ec9b9a0$bc5d2ce0$@democon.be> <652258ae-5160-921c-f6bb-555bb196ab8b@opensips.org> <135235270.72204.1548431212249.JavaMail.zimbra@skillsearch.ca> <6c99641c-279f-6cf1-78cc-f41dcd592263@opensips.org> Message-ID: <5c0f395e-fbe0-05c4-5fac-b88244c73b98@opensips.org> Hi, Abdul! Unfortunately handling message fragmentation is not yet supported, nor multilingual support, as there was no request for them. But of course, if you need this, you can definitely add a feature request for them! Best regards, Răzvan On 1/28/19 2:16 AM, Abdul Basit wrote: > nice addition as this will eliminate the kannel integration logic along > side with SIP services. > > How to handle message max. characters length while transmitting the > message over SMPP and concatenation on receiving message from SMSC? > Is this implementation address multilingual support, i.e Unicode support? > > -- > regards, > > abdul basit > > On Fri, 25 Jan 2019 at 20:51, Liviu Chircu > wrote: > > OpenSIPS GitHub [1] will build it on Sunday :) > > [1]: https://github.com/opensips-github > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 25.01.2019 17:46, Slava Bendersky wrote: >> Hello Răzvan, >> >> proto_smpp missing README in main tree  or just wasn't generated ? >> >> volga629 >> >> ------------------------------------------------------------------------ >> *From: *"Răzvan Crainea" >> >> *To: *"OpenSIPS users mailling list" >> >> *Sent: *Friday, January 25, 2019 4:34:38 AM >> *Subject: *Re: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP >> messages >> >> Unfortunately this is a feature, not a bug, so it won't be >> backported to >> 2.4. >> >> Cheers, >> Răzvan >> >> On 1/24/19 7:21 PM, johan de clercq wrote: >> > Do you backport this to 2.4 ? >> > >> > BR, >> > >> > -----Original Message----- >> > From: Users >> On Behalf Of Razvan Crainea >> > Sent: Thursday, January 24, 2019 5:37 PM >> > To: OpenSIPS users mailling list >> ; OpenSIPS devel mailling list >> >> > Subject: [OpenSIPS-Users] [NEW] Gateway between SIP and SMPP >> messages >> > >> > Hi, Everyone! >> > >> > Check out the latest OpenSIPS module, proto_smpp[1], that you >> can use to create a two-way bridge between SIP and SMPP text >> messages. Read more about this on our blog[2]. >> > >> > [1] https://opensips.org/html/docs/modules/3.0.x/proto_smpp.html >> > [2] >> > >> https://blog.opensips.org/2019/01/24/gateway-between-sip-and-smpp-messages/ >> > >> > Cheers, >> > -- >> > Răzvan Crainea >> > OpenSIPS Core Developer >> > http://www.opensips-solutions.com >> > >> > _______________________________________________ >> > Users mailing list >> > Users at lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> > >> > _______________________________________________ >> > Users mailing list >> > Users at lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> >> -- >> Răzvan Crainea >> OpenSIPS Core Developer >> http://www.opensips-solutions.com >> Meet the OpenSIPS team at the next OpenSIPS Summit: >> https://www.opensips.org/events >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From bogdan at opensips.org Mon Jan 28 07:18:39 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 28 Jan 2019 14:18:39 +0200 Subject: [OpenSIPS-Users] Early Birds for OpenSIPS Summit Message-ID: Hi all, Heads up - you are in the last days of Early Birds discounts for registering for the OpenSIPS Summit 2019 - do not miss the deals and register today https://www.opensips.org/events/Summit-2019Amsterdam/#pricing See you in Amsterdam, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Tue Jan 29 12:20:56 2019 From: johan at democon.be (johan de clercq) Date: Tue, 29 Jan 2019 18:20:56 +0100 Subject: [OpenSIPS-Users] topology_hiding : ACK has extra via header. Message-ID: <01af01d4b7f6$ff778dd0$fe66a970$@democon.be> Hi, Using opensips 2.4.4, Scenario Phone -> Opensips with topology-hiding -> Provider When the INVITE comes in from the phone, the ip addr of the phone is in the via header. When opensips forwards the request to the provider, the ip addr of opensips is in the via header. When 200 OK comes in from the provider and forwarded / relayed to the phone the via header is correct. When ACK comes in from the phone, opensips adds an extra via header so the provider sees the ip address of the originating phone. Samy Go opened an issue on this a few years ago, but I couldn't find a solution. Johan De Clercq, Managing Director Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke Tel +3256980990 - GSM +32478720104 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 15602 bytes Desc: not available URL: From goup2010 at gmail.com Tue Jan 29 12:39:35 2019 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Tue, 29 Jan 2019 19:39:35 +0200 Subject: [OpenSIPS-Users] more INVITEs Message-ID: Hello Opensips comunity, I must to solve the following case: One SIP device send to OpenSips not one, but three identical INVITEs (with same Call-ID). How do I set the OpenSips to only handle the first one and ignore others? Best regards, Dragomir -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Tue Jan 29 12:49:12 2019 From: johan at democon.be (johan de clercq) Date: Tue, 29 Jan 2019 18:49:12 +0100 Subject: [OpenSIPS-Users] more INVITEs In-Reply-To: References: Message-ID: <01c501d4b7fa$f24bade0$d6e309a0$@democon.be> If there is a gap from 0,5 sec between invite 1 and 2; and a gap of 1 sec between invite 2 and 3 then it are just retransmissioins. In that case, nothing to worry about. From: Users On Behalf Of Dragomir Haralambiev Sent: Tuesday, January 29, 2019 6:40 PM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] more INVITEs Hello Opensips comunity, I must to solve the following case: One SIP device send to OpenSips not one, but three identical INVITEs (with same Call-ID). How do I set the OpenSips to only handle the first one and ignore others? Best regards, Dragomir -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Tue Jan 29 12:57:39 2019 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Tue, 29 Jan 2019 19:57:39 +0200 Subject: [OpenSIPS-Users] more INVITEs In-Reply-To: <01c501d4b7fa$f24bade0$d6e309a0$@democon.be> References: <01c501d4b7fa$f24bade0$d6e309a0$@democon.be> Message-ID: They come very fast in less than 0.5 s На вт, 29.01.2019 г. в 19:52 ч. johan de clercq написа: > If there is a gap from 0,5 sec between invite 1 and 2; and a gap of 1 sec > between invite 2 and 3 then it are just retransmissioins. In that case, > nothing to worry about. > > > > *From:* Users *On Behalf Of *Dragomir > Haralambiev > *Sent:* Tuesday, January 29, 2019 6:40 PM > *To:* OpenSIPS users mailling list > *Subject:* [OpenSIPS-Users] more INVITEs > > > > Hello Opensips comunity, > > > > I must to solve the following case: > > One SIP device send to OpenSips not one, but three identical INVITEs (with > same Call-ID). > > > > How do I set the OpenSips to only handle the first one and ignore others? > > > > Best regards, > > Dragomir > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Jan 29 16:01:32 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 29 Jan 2019 23:01:32 +0200 Subject: [OpenSIPS-Users] OpenSIPS @ ClueCon Weekly Message-ID: <60611ed1-35a6-c193-5d01-e94424251140@opensips.org> Hi there, Join OpenSIPS for a new ClueCon Weekly therapy session to talk about the upcoming 3.0 release and about the OpenSIPS Summit 2019. When? 30th of Jan , 6pm GMT Where? https://youtu.be/DV01wKXV7Bo See you there !! -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ From alexei.vasilyev at gmail.com Wed Jan 30 02:12:55 2019 From: alexei.vasilyev at gmail.com (vasilevalex) Date: Wed, 30 Jan 2019 00:12:55 -0700 (MST) Subject: [OpenSIPS-Users] topology_hiding : ACK has extra via header. In-Reply-To: <01af01d4b7f6$ff778dd0$fe66a970$@democon.be> References: <01af01d4b7f6$ff778dd0$fe66a970$@democon.be> Message-ID: <1548832375464-0.post@n2.nabble.com> Hi Johan, We are using topology_hiding in tm mode. I've checked your case, ACK has only one VIA header. What about BYE from the phone (if phone ends call)? Does it has extra VIA? Do you use dialog or tm mode for topology hiding? ----- --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From johan at democon.be Wed Jan 30 02:43:37 2019 From: johan at democon.be (johan de clercq) Date: Wed, 30 Jan 2019 08:43:37 +0100 Subject: [OpenSIPS-Users] topology_hiding : ACK has extra via header. In-Reply-To: <1548832375464-0.post@n2.nabble.com> References: <01af01d4b7f6$ff778dd0$fe66a970$@democon.be> <1548832375464-0.post@n2.nabble.com> Message-ID: <003b01d4b86f$83a8deb0$8afa9c10$@democon.be> I use with dialog (need profiles in my setup). The incoming ACK from the phone has only one via header whereas opensips goes to the carrier with two. BR, -----Original Message----- From: Users On Behalf Of vasilevalex Sent: Wednesday, January 30, 2019 8:13 AM To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] topology_hiding : ACK has extra via header. Hi Johan, We are using topology_hiding in tm mode. I've checked your case, ACK has only one VIA header. What about BYE from the phone (if phone ends call)? Does it has extra VIA? Do you use dialog or tm mode for topology hiding? ----- --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From johan at democon.be Wed Jan 30 14:27:14 2019 From: johan at democon.be (johan de clercq) Date: Wed, 30 Jan 2019 20:27:14 +0100 Subject: [OpenSIPS-Users] topology_hiding : ACK has extra via header. In-Reply-To: <1548832375464-0.post@n2.nabble.com> References: <01af01d4b7f6$ff778dd0$fe66a970$@democon.be> <1548832375464-0.post@n2.nabble.com> Message-ID: <008d01d4b8d1$ce8cda00$6ba68e00$@democon.be> Hi, I found it : 1. This code results in having only one VIA header: ##initial INVITE if (is_method("INVITE") && !has_totag()) { xlog("callid=$ci: Route[0]: initial INVITE is coming in from A: $fU to B: $rU"); xlog("callid=$ci: Route[0]: let's do accounting first"); do_accounting("db","cdr"); topology_hiding(); 2. The code resulting in 2 VIA headers: ##initial INVITE if (is_method("INVITE") && !has_totag()) { xlog("callid=$ci: Route[0]: initial INVITE is coming in from A: $fU to B: $rU"); xlog("callid=$ci: Route[0]: let's do accounting first"); topology_hiding(); do_accounting("db","cdr"); Is this now a bug or can somebody please explain why the code in 2 results in 2 VIA headers ? -----Original Message----- From: Users On Behalf Of vasilevalex Sent: Wednesday, January 30, 2019 8:13 AM To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] topology_hiding : ACK has extra via header. Hi Johan, We are using topology_hiding in tm mode. I've checked your case, ACK has only one VIA header. What about BYE from the phone (if phone ends call)? Does it has extra VIA? Do you use dialog or tm mode for topology hiding? ----- --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Thu Jan 31 07:18:04 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 31 Jan 2019 14:18:04 +0200 Subject: [OpenSIPS-Users] topology_hiding : ACK has extra via header. In-Reply-To: <008d01d4b8d1$ce8cda00$6ba68e00$@democon.be> References: <01af01d4b7f6$ff778dd0$fe66a970$@democon.be> <1548832375464-0.post@n2.nabble.com> <008d01d4b8d1$ce8cda00$6ba68e00$@democon.be> Message-ID: Hi, Johan! Are there two identical via headers, or only one? What is the OpenSIPS version you are using? Best regards, Răzvan On 1/30/19 9:27 PM, johan de clercq wrote: > Hi, > > I found it : > > 1. This code results in having only one VIA header: > ##initial INVITE > if (is_method("INVITE") && !has_totag()) > { > xlog("callid=$ci: Route[0]: initial INVITE is coming in from A: $fU to B: $rU"); > xlog("callid=$ci: Route[0]: let's do accounting first"); > do_accounting("db","cdr"); > topology_hiding(); > > 2. The code resulting in 2 VIA headers: > ##initial INVITE > if (is_method("INVITE") && !has_totag()) > { > xlog("callid=$ci: Route[0]: initial INVITE is coming in from A: $fU to B: $rU"); > xlog("callid=$ci: Route[0]: let's do accounting first"); > topology_hiding(); > do_accounting("db","cdr"); > > Is this now a bug or can somebody please explain why the code in 2 results in 2 VIA headers ? > > -----Original Message----- > From: Users On Behalf Of vasilevalex > Sent: Wednesday, January 30, 2019 8:13 AM > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] topology_hiding : ACK has extra via header. > > Hi Johan, > > We are using topology_hiding in tm mode. > I've checked your case, ACK has only one VIA header. > > What about BYE from the phone (if phone ends call)? Does it has extra VIA? > Do you use dialog or tm mode for topology hiding? > > > > ----- > --- > Alexey Vasilyev > -- > Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From Johan at democon.be Thu Jan 31 07:21:41 2019 From: Johan at democon.be (Johan De Clercq) Date: Thu, 31 Jan 2019 13:21:41 +0100 Subject: [OpenSIPS-Users] topology_hiding : ACK has extra via header. In-Reply-To: References: <01af01d4b7f6$ff778dd0$fe66a970$@democon.be> <1548832375464-0.post@n2.nabble.com> <008d01d4b8d1$ce8cda00$6ba68e00$@democon.be> Message-ID: 2 different ones and I use 244 On Thu, 31 Jan 2019, 13:20 Răzvan Crainea, wrote: > Hi, Johan! > > Are there two identical via headers, or only one? > What is the OpenSIPS version you are using? > > Best regards, > Răzvan > > On 1/30/19 9:27 PM, johan de clercq wrote: > > Hi, > > > > I found it : > > > > 1. This code results in having only one VIA header: > > ##initial INVITE > > if (is_method("INVITE") && !has_totag()) > > { > > xlog("callid=$ci: Route[0]: initial INVITE is coming in from A: > $fU to B: $rU"); > > xlog("callid=$ci: Route[0]: let's do accounting first"); > > do_accounting("db","cdr"); > > topology_hiding(); > > > > 2. The code resulting in 2 VIA headers: > > ##initial INVITE > > if (is_method("INVITE") && !has_totag()) > > { > > xlog("callid=$ci: Route[0]: initial INVITE is coming in from A: > $fU to B: $rU"); > > xlog("callid=$ci: Route[0]: let's do accounting first"); > > topology_hiding(); > > do_accounting("db","cdr"); > > > > Is this now a bug or can somebody please explain why the code in 2 > results in 2 VIA headers ? > > > > -----Original Message----- > > From: Users On Behalf Of vasilevalex > > Sent: Wednesday, January 30, 2019 8:13 AM > > To: users at lists.opensips.org > > Subject: Re: [OpenSIPS-Users] topology_hiding : ACK has extra via header. > > > > Hi Johan, > > > > We are using topology_hiding in tm mode. > > I've checked your case, ACK has only one VIA header. > > > > What about BYE from the phone (if phone ends call)? Does it has extra > VIA? > > Do you use dialog or tm mode for topology hiding? > > > > > > > > ----- > > --- > > Alexey Vasilyev > > -- > > Sent from: > http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > Meet the OpenSIPS team at the next OpenSIPS Summit: > https://www.opensips.org/events > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Jan 31 07:25:47 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 31 Jan 2019 14:25:47 +0200 Subject: [OpenSIPS-Users] topology_hiding : ACK has extra via header. In-Reply-To: References: <01af01d4b7f6$ff778dd0$fe66a970$@democon.be> <1548832375464-0.post@n2.nabble.com> <008d01d4b8d1$ce8cda00$6ba68e00$@democon.be> Message-ID: Can you upload somewhere a SIP trace, so we can understand the setup and SIP flow. Also, is the VIA appearing only in ACK, or in INVITE too? Best regards, Razvan On 1/31/19 2:21 PM, Johan De Clercq wrote: > 2 different ones and I use 244 > > On Thu, 31 Jan 2019, 13:20 Răzvan Crainea, > wrote: > > Hi, Johan! > > Are there two identical via headers, or only one? > What is the OpenSIPS version you are using? > > Best regards, > Răzvan > > On 1/30/19 9:27 PM, johan de clercq wrote: > > Hi, > > > > I found it : > > > > 1. This code results in having only one VIA header: > >      ##initial INVITE > >      if (is_method("INVITE") && !has_totag()) > >      { > >          xlog("callid=$ci: Route[0]: initial INVITE is coming in > from A: $fU to B: $rU"); > >          xlog("callid=$ci: Route[0]: let's do accounting first"); > >          do_accounting("db","cdr"); > >          topology_hiding(); > > > > 2. The code resulting in 2 VIA headers: > >      ##initial INVITE > >      if (is_method("INVITE") && !has_totag()) > >      { > >          xlog("callid=$ci: Route[0]: initial INVITE is coming in > from A: $fU to B: $rU"); > >          xlog("callid=$ci: Route[0]: let's do accounting first"); > >          topology_hiding(); > >          do_accounting("db","cdr"); > > > > Is this now a bug or can somebody please explain why the code in > 2 results in 2 VIA headers ? > > > > -----Original Message----- > > From: Users > On Behalf Of vasilevalex > > Sent: Wednesday, January 30, 2019 8:13 AM > > To: users at lists.opensips.org > > Subject: Re: [OpenSIPS-Users] topology_hiding : ACK has extra via > header. > > > > Hi Johan, > > > > We are using topology_hiding in tm mode. > > I've checked your case, ACK has only one VIA header. > > > > What about BYE from the phone (if phone ends call)? Does it has > extra VIA? > > Do you use dialog or tm mode for topology hiding? > > > > > > > > ----- > > --- > > Alexey Vasilyev > > -- > > Sent from: > http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > Meet the OpenSIPS team at the next OpenSIPS Summit: > https://www.opensips.org/events > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From razvan at opensips.org Thu Jan 31 07:29:52 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 31 Jan 2019 14:29:52 +0200 Subject: [OpenSIPS-Users] more INVITEs In-Reply-To: References: <01c501d4b7fa$f24bade0$d6e309a0$@democon.be> Message-ID: Hi, Dragomir! You can detect retransmissions of the same message using the t_check_trans() function. But in order to use it properly, make sure you create the transaction as soon as possible. A short snippet that could solve this is: t_newtran(); t_check_status(); For more information please consult the documentation: https://opensips.org/html/docs/modules/2.4.x/tm.html#func_t_newtran Best regards, Răzvan On 1/29/19 7:57 PM, Dragomir Haralambiev wrote: > They come very fast in less than 0.5 s > > На вт, 29.01.2019 г. в 19:52 ч. johan de clercq > написа: > > If there is a gap from 0,5 sec between invite 1 and 2; and a gap of > 1 sec between invite 2 and 3 then it are just retransmissioins.  In > that case, nothing to worry about. ____ > > __ __ > > *From:* Users > *On Behalf Of *Dragomir > Haralambiev > *Sent:* Tuesday, January 29, 2019 6:40 PM > *To:* OpenSIPS users mailling list > > *Subject:* [OpenSIPS-Users] more INVITEs____ > > __ __ > > Hello Opensips comunity,____ > > __ __ > > I must to solve the following case:____ > > One SIP device send to OpenSips not one, but three identical INVITEs > (with same Call-ID).____ > > __ __ > > How do I set the OpenSips to only handle the first one and ignore > others?____ > > __ __ > > Best regards,____ > > Dragomir____ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com Meet the OpenSIPS team at the next OpenSIPS Summit: https://www.opensips.org/events From Johan at democon.be Thu Jan 31 07:30:27 2019 From: Johan at democon.be (Johan De Clercq) Date: Thu, 31 Jan 2019 13:30:27 +0100 Subject: [OpenSIPS-Users] topology_hiding : ACK has extra via header. In-Reply-To: References: <01af01d4b7f6$ff778dd0$fe66a970$@democon.be> <1548832375464-0.post@n2.nabble.com> <008d01d4b8d1$ce8cda00$6ba68e00$@democon.be> Message-ID: I will send this evening a small trace. On Thu, 31 Jan 2019, 13:27 Răzvan Crainea, wrote: > Can you upload somewhere a SIP trace, so we can understand the setup and > SIP flow. > Also, is the VIA appearing only in ACK, or in INVITE too? > > Best regards, > Razvan > > On 1/31/19 2:21 PM, Johan De Clercq wrote: > > 2 different ones and I use 244 > > > > On Thu, 31 Jan 2019, 13:20 Răzvan Crainea, > > wrote: > > > > Hi, Johan! > > > > Are there two identical via headers, or only one? > > What is the OpenSIPS version you are using? > > > > Best regards, > > Răzvan > > > > On 1/30/19 9:27 PM, johan de clercq wrote: > > > Hi, > > > > > > I found it : > > > > > > 1. This code results in having only one VIA header: > > > ##initial INVITE > > > if (is_method("INVITE") && !has_totag()) > > > { > > > xlog("callid=$ci: Route[0]: initial INVITE is coming in > > from A: $fU to B: $rU"); > > > xlog("callid=$ci: Route[0]: let's do accounting first"); > > > do_accounting("db","cdr"); > > > topology_hiding(); > > > > > > 2. The code resulting in 2 VIA headers: > > > ##initial INVITE > > > if (is_method("INVITE") && !has_totag()) > > > { > > > xlog("callid=$ci: Route[0]: initial INVITE is coming in > > from A: $fU to B: $rU"); > > > xlog("callid=$ci: Route[0]: let's do accounting first"); > > > topology_hiding(); > > > do_accounting("db","cdr"); > > > > > > Is this now a bug or can somebody please explain why the code in > > 2 results in 2 VIA headers ? > > > > > > -----Original Message----- > > > From: Users > > On Behalf Of vasilevalex > > > Sent: Wednesday, January 30, 2019 8:13 AM > > > To: users at lists.opensips.org > > > Subject: Re: [OpenSIPS-Users] topology_hiding : ACK has extra via > > header. > > > > > > Hi Johan, > > > > > > We are using topology_hiding in tm mode. > > > I've checked your case, ACK has only one VIA header. > > > > > > What about BYE from the phone (if phone ends call)? Does it has > > extra VIA? > > > Do you use dialog or tm mode for topology hiding? > > > > > > > > > > > > ----- > > > --- > > > Alexey Vasilyev > > > -- > > > Sent from: > > > http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > _______________________________________________ > > > Users mailing list > > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.com > > Meet the OpenSIPS team at the next OpenSIPS Summit: > > https://www.opensips.org/events > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > Meet the OpenSIPS team at the next OpenSIPS Summit: > https://www.opensips.org/events > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Thu Jan 31 14:37:23 2019 From: social at bohboh.info (Social Boh) Date: Thu, 31 Jan 2019 14:37:23 -0500 Subject: [OpenSIPS-Users] OpenSIPs Load_balancer replication Message-ID: Hello, I'm trying to replicate load data about load_balancer module between two OpenSIPs active instance. Table load_balancer use /b after resource name. Script use /b on the load_balance function. On the dialog module parameters: modparam("dialog", "dialog_replication_cluster", 1) modparam("dialog", "profile_replication_cluster", 1) Cluster working correctly. When a call arrive to OpenSIPs1 on the OpenSIPs2: WARNING:dialog:fetch_socket_info: non-local socket ...ignoring WARNING:dialog:fetch_socket_info: non-local socket ...ignoring ERROR:dialog:dlg_replicated_create: Replicated dialog doesn't match any listening sockets ERROR:dialog:receive_dlg_repl: Failed to process a binary packet! How can I solve? Thank you -- --- I'm SoCIaL, MayBe From spanda at 3clogic.com Wed Jan 23 06:17:53 2019 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 23 Jan 2019 16:47:53 +0530 Subject: [OpenSIPS-Users] I deleted an entry from clusterer table , but still opensips try to ping that node . In-Reply-To: References: Message-ID: HI , The data get stored in the cache for use of cluster module won't get deleted . Like , when I am saving something on location table that get deleted when the user get logout . The reference cfg file what I get for cluster module is only SET the data in the cache and GET that from cache . How can I remove the data from the cache ? Attached my config file . Please have a look into it . Let me know if I am doing anything wrong . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Thu, Jan 17, 2019 at 11:18 AM Mohit Sachan wrote: > hi > Can you suggest me how to install oversip for webrtc for opensips-2.4 in > centos. > > On Thu, Jan 3, 2019 at 10:56 PM Vlad Patrascu wrote: > >> Hi Sasmita, >> >> By default, there is no clusterer replication if "replicate_contacts_to" >> parameter is not set in usrloc. Also, even if another node is sending >> replication packets, they will no get processed on the receiving node >> unless "accept_replicated_contacts" is set. On a typical setup, both these >> parameter should be set on all nodes. >> >> Are you getting any other errors in the logs besides that "parameter not >> found" ? >> >> Btw, I strongly suggest updating to 2.4 as it has received major upgrades >> in terms of clustering. >> >> Regards, >> >> Vlad Patrascu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> On 01/02/2019 12:10 PM, Sasmita Panda wrote: >> >> Hi, >> I have another doubt . Please do help me . >> >> When I am reading usrloc module document , its saying in a cluster if we >> want to replicate the contacts across the cluster then we have to set a >> parameter as below . >> >> modparam("usrloc", "replicate_contacts_to", 1) >> >> The default value is 0 , where no cluster id is mentioned . >> >> I have not set this , I have a cluster having 2 node . While I am registering a user , the contact is getting replicated between 2 nodes . >> >> If I am trying to mention this parameter , then opensips is not getting started . Its saying >> >> *Parameter not found in module * >> >> *So , my question is , if this parameter is not set , still how contact replication is happening ? Is this the default behavior of cluster module ? * >> >> *May be my question is foolish ,it will be great if anybody will explain this . * >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> On Wed, Jan 2, 2019 at 12:22 PM Sasmita Panda wrote: >> >>> Hi Sammy, >>> >>> Yes , you are right . I need to reload the cluster data through MI >>> command . After reloading its seems fine . >>> >>> I was not aware about the fact that the cluster data also get shared >>> with all nodes when I am adding that in 1 node only . >>> >>> Thank you for your explanation . Its really helpful . >>> >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Senior Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> >>> On Mon, Dec 31, 2018 at 10:25 PM SamyGo wrote: >>> >>>> Hi, >>>> Did you restart OpenSIPS process on both node1, and 2 simultaneously ? >>>> The way I look at this is one of the two nodes kept the 3rd one in the >>>> memory and restarting both nodes one at a time resulted in both sharing >>>> their node structure and hence node3 stayed visible. >>>> I think possible way to remove a node gracefully would be to disable >>>> the node via the MI command and then remove from DB. I will try doing this >>>> on my test setup as well. >>>> >>>> Regards, >>>> Sammy >>>> >>>> >>>> On Fri, Dec 28, 2018 at 6:40 AM Sasmita Panda >>>> wrote: >>>> >>>>> Hi All, >>>>> >>>>> I have a cluster of 2 nodes . Both in working condition . Then I >>>>> added another node in the same cluster which is down . >>>>> >>>>> I restarted the opensips process , so it starts pinging the new node >>>>> to check its status . As the new node is down , other nodes in the >>>>> cluster wont get any reply for the ping . Then I remove the 3rd node from >>>>> the cluster table and restart the opensips process . >>>>> >>>>> Now what I am getting in logs is , still the 2 working node in the >>>>> cluster try to ping the 3rd node which is not in the DB . >>>>> >>>>> Is this an issue on the cluster module or I am doing something wrong >>>>> ?? Please help me . >>>>> *Thanks & Regards* >>>>> *Sasmita Panda* >>>>> *Senior Network Testing and Software Engineer* >>>>> *3CLogic , ph:07827611765* >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: opensips-cluster-new.cfg Type: application/octet-stream Size: 22368 bytes Desc: not available URL: From Ben.Newlin at genesys.com Fri Jan 25 11:44:32 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Fri, 25 Jan 2019 16:44:32 +0000 Subject: [OpenSIPS-Users] Invalid parameter errors In-Reply-To: References: <3290d7c2-2577-d130-3411-702c153bf41a@opensips.org> Message-ID: Liviu, You’re right, there are quite a lot of them. Almost all of our use of parameterized routes and json access is in a subset of code that performs our logging, and none of that changed when the errors began. Routes: route(is_null, $param(1)) route(is_null, $param(2)) route(is_json, $param(2)) route(is_null, $json(rlog/$param(1))) route(json_rlog, "thread_name", $pp); route(json_rlog, "level", $param(1)); route(json_rlog, "route", $param(2)); route(rlog_start, $param(1), $param(2)); route(is_null, $param(3)) route(json_rlog, "message", $param(3)); route(is_null, $json(rlog_msg)) route(json_rlog, "message", $var(rlog_msg)); route(rlog_v, $param(1)); route(json_rlog, "callID", $ci); route(json_rlog, "requestUri", $ru); route(json_rlog, "fromUri", $fu); route(json_rlog, "toUri", $tu); route(json_rlog, "sipStatus", $rs); route(json_rlog, "sipReason", $rr); route(json_rlog, "mflags", $mf); route(json_rlog, "conversation", $avp(cnv_id)); route(json_rlog, "callState", $avp(call_state)); route(json_rlog, "method", $rm); route(is_null, $json(rlog_msg)) route(is_null, $param($var(add_rlog_i))) route(is_null, $param($var(add_rlog_j))) route(is_json, $param($var(add_rlog_j))) route(is_null, $param(1)) route(is_null, $param($var(i)))) route(add_rlog_msg_data, "function", "t_relay", "retcode", $var(relay_rc)); route(add_rlog_msg_data, "sipStatus", $param(1), "sipReason", $param(2)); route(add_rlog_msg_data, "maxfwd", $hdr(Max-Forwards)); route(is_null, $param(1)) route(is_null, $param(1)) route(is_null, $param(1)) route(is_null, $var(ft_cache_val)) route(is_null, $param(2)) route(is_null, $var(ft_orgs)) route(add_rlog_msg_data, "feature", $param(1), "orgID", $param(2), "allow", $var(ft_allow)); route(add_rlog_msg_data, "state", $shv(state)); route(add_rlog_msg_data, "group", $var(group)); route(add_rlog_msg_data, "dir", $var(dir), "group", $var(group)); route(add_rlog_msg_data, "rr_params", $rr_params); route(add_rlog_msg_data, "rr_params", $rr_params); route(add_rlog_msg_data, "state", $shv(state)); route(add_rlog_msg_data, "domain", $hdr(x-special-header)); JSON: $json(rlog/$param(1)) = NULL; $json(rlog/$param(1)) := $param(2); route(is_null, $json(rlog/$param(1))) $json(rlog/$param(1)) = $(param(2){re.subst,/"/\\"/g}); $json(rlog/message) := $json(rlog_msg); $json(rlog/t) = $(time("%Y-%m-%dT%T."){s.select,1,"}) + $(Tsm{s.fill.left,0,6}) + "Z"; #" $json(rlog_msg/$param($var(add_rlog_i))) = ""; $json(rlog_msg/$param($var(add_rlog_i))) := $param($var(add_rlog_j)); $json(rlog_msg/$param($var(add_rlog_i))) = $(param($var(add_rlog_j)){re.subst,/"/\\"/g}); $json(rlog_msg/function) = $param(1); $json(rlog_msg/retcode) = $var(add_rlog_msg_retcode); $json(rlog_msg/params[]) = $(param($var(i)){re.subst,/"/\\"/g}); $json(rlog_msg/destUri) = $du; $json(rlog_msg/srcIp) = $si; $json(rlog_msg/srcPort) = $sp; $json(rlog_msg/replyCode) = $T_reply_code; $json(rlog_msg/protocol) = $(proto{s.tolower}); $json(rlog/errClass) = $(err.class{s.escape.common}); $json(rlog/errLevel) = $(err.level{s.escape.common}); $json(rlog/errInfo) = $(err.info{s.escape.common}); $json(rlog/errRcode) = $(err.rcode{s.escape.common}); $json(rlog/errRreason) = $(err.rreason{s.escape.common}); $json(rlog/msg) = $(mb{s.escape.common}); We have a few cases where we used parameterized parameter references, if that makes sense. I suspect it may be something related to that? For example, the only error line number that references a line where json access or variable assignment is actually being performed is here: $json(rlog_msg/$param($var(add_rlog_i))) = $(param($var(add_rlog_j)){re.subst,/"/\\"/g}); Ben Newlin From: Users on behalf of Liviu Chircu Reply-To: OpenSIPS users mailling list Date: Thursday, January 24, 2019 at 3:12 PM To: "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] Invalid parameter errors Hi Ben, We are actually dealing with two bugs here, which may or may not be related to one another. Bug #1: bad? variable during a route() call ------------------------------------------------------- For this one, can you enumerate all "route()" calls in your script which pass at least one variable, along with their full parameter call syntax? Example call: route(sequential_requests, $rm, $avp(myinfo)); Bug #2: bad "key variable" during a $json expansion ---------------------------------------------------------------------- For this one, can you enumerate all $json() variable appearances which include at least one parameterized key access? I realize there may be lots of these, but you may group them into "categories" and print out a few ones that might be relevant (i.e. their index may contain an INT-only variable, which is >wrong<). Example appearances: $json(http_body/$var(tag)) $json(http_body/users[0]/$avp(username)) Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 24.01.2019 01:37, Ben Newlin wrote: Liviu, Thank you for the quick response. I do see 2 such errors shortly after startup: ERROR:core:pv_get_param: cannot get spec value ERROR:core:pv_get_param: cannot get spec value However, after that it just continues on with more of the same errors that keep scrolling. There is a variation of the scrolling errors that was I didn’t included before, in case it helps: ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ERROR:json:expand_tag_list: Non string value in key ERROR:json:pv_set_json: Cannot expand variables in path ERROR:core:do_assign: setting PV failed ERROR:core:do_assign: error at opensips.cfg:346 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 There aren’t any other non-repeating errors. I have picked up your commit and will try to gather more information from it, but this issue is primarily happening in our production environment so it may take a bit. Also, I haven’t completely verified this yet, but it seems that enabling TLS has made the errors stop somehow. Continuing to investigate that. Ben Newlin From: Users on behalf of Liviu Chircu Reply-To: OpenSIPS users mailling list Date: Tuesday, January 22, 2019 at 6:08 PM To: "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] Invalid parameter errors Hi, Ben! The strange "...type 1836017711" errors seem to be caused by a poorly handed error condition (a secondary bug), which is now fixed [1]. If this theory holds, you must have a "cannot get spec value" error (or slew of errors) in the earlier section of your OpenSIPS log (possibly right after restart or shortly after starting to process traffic). Could you please confirm/infirm the above? If true, are there any other relevant errors thrown around that initial "cannot get spec value" error message? Those error logs could be key to making progress in understanding the main bug. Best regards, [1]: https://github.com/OpenSIPS/opensips/commit/52ff74af8702a Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 22.01.2019 20:58, Ben Newlin wrote: Hi, Since upgrading to 2.4.4 we are seeing the following logs scrolling nearly continuously on our servers: ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:583 ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value ALERT:core:pv_get_param: BUG: invalid parameter type 1836017711 ERROR:core:comp_scriptvar: cannot get left var value WARNING:core:do_action: error in expression at opensips.cfg:439 It seems to be related to our use of the json module. We often pass json variable types as parameters to other routes and I believe the errors are caused by that. But it’s hard to say as there are a few different script lines referenced in the errors, but some of them point to return statements and other code sections that don’t really make sense. For example, line 583 referenced in the error above is: return(-1); Any ideas? Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: