From spanda at 3clogic.com Fri Apr 1 06:19:31 2022 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 1 Apr 2022 11:49:31 +0530 Subject: [OpenSIPS-Users] Need some help in adding custome header in Cancel Request . In-Reply-To: <4e9d2d8d-a8f6-2cfe-4e04-fe0de7b87319@opensips.org> References: <16d51ba6-aab2-53e8-ff87-9e3566a632bf@opensips.org> <4e9d2d8d-a8f6-2cfe-4e04-fe0de7b87319@opensips.org> Message-ID: I had done this through t_add_cancel_reason[1]. Working for me . Thanks all . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Thu, Mar 31, 2022 at 7:19 PM Bogdan-Andrei Iancu wrote: > Also take a look at this post > https://blog.opensips.org/2016/11/15/cancel-request-and-reason-header/ > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 3/9/22 6:14 PM, Răzvan Crainea wrote: > > Hi, Sasmita! > > > > I actually don't think local_route is run for CANCEL messages. > > You may want to try to add a more complex reason using > > t_add_cancel_reason[1]. > > > > [1] https://opensips.org/docs/modules/3.2.x/tm.html#idp6205808 > > > > Best regards, > > > > Răzvan Crainea > > OpenSIPS Core Developer > > http://www.opensips-solutions.com > > > > On 3/9/22 12:07, Sasmita Panda wrote: > >> My call flow is like below . > >> > >> A -- > INVITE TO OPENSIPS -- > B > >> A -- > CANCEL TO OPENSIPS -- > B > >> > >> While A sends Cancel to Opensips (adds a custom header ) . When > >> Opensips generates Cancel for B it won't add the custom header . > >> > >> This can be done by local_route ? > >> > >> */Thanks & Regards/* > >> /Sasmita Panda/ > >> /Senior Network Testing and Software Engineer/ > >> /3CLogic , ph:07827611765/ > >> > >> > >> On Wed, Mar 9, 2022 at 3:27 PM Nick Altmann >> > wrote: > >> > >> Hi, > >> > >> If cancel request generated by opensips, then you can control it > >> from local_route. > >> > >> -- > >> Nick > >> > >> ср, 9 мар. 2022 г. в 10:54, Sasmita Panda >> >: > >> > >> Hi All, > >> > >> Cancel is generated Hop by Hop . When the Opensips server > >> receives a Cancel , Then it generates Cancel for the next > >> party . > >> > >> I am adding a custom header in the Cancel request , but when the > >> next Hop Cancel is getting generated that custom header is not > >> getting added . How will I pass the custom header in the Cancel > >> request to the destination ? > >> > >> */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 kwem at gmx.de Fri Apr 1 11:40:51 2022 From: kwem at gmx.de (Karsten Wemheuer) Date: Fri, 01 Apr 2022 13:40:51 +0200 Subject: [OpenSIPS-Users] Call forking, branches, Record-routing In-Reply-To: <45a50a6c-0aec-6862-3710-8dbf2085ec8e@opensips.org> References: <45a50a6c-0aec-6862-3710-8dbf2085ec8e@opensips.org> Message-ID: <196889f56f647f3f44e1c34c9b30f21752ca19f8.camel@gmx.de> Hi Bogdan-Andrei, I tried with record_route and do some corrections. The part for calls from the PBX via the proxy to the phones now looks like this: route[TOPHONES] { record_route(); lookup("location"); t_on_branch("AST2PHONE"); if (!t_relay()) { sl_reply_error(); } } branch_route[AST2PHONE] { if ( isbflagset("NAT") ) { set_advertised_address(PUBLIC-IP); set_advertised_port(PUBLIC-PORT); } else { set_advertised_address("10.0.2.3"); set_advertised_port("5061"); } t_on_reply("AST2PHONE_REPlY"); } This works with calls to phones on the LAN and also with calls to phones behind NAT. If a call is to go to one local and one NAT accessible phone in parallel, an error occurs after the call was taken on the local phone. The CANCEL to the phone behind NAT is sent by the proxy without routing information and the phone ignores this CANCEL. A call to a telephone behind NAT can also not be terminated with CANCEL. Where is my mistake? Thanks in advance Best regards, Karsten Am Donnerstag, dem 31.03.2022 um 17:44 +0300 schrieb Bogdan-Andrei Iancu: > Hi Karsten, > > See my prev email, just to record_route() before the t_relay() for > the > initial INVITE. And the loose_route() stuff for whatever > sequential/in-dialog requests. > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 3/31/22 2:50 PM, Karsten Wemheuer wrote: > > Hi*, > > > > I have a understanding problem regarding branches and call forking. > > A call from a PBX is to be routed to phone(s) via OpenSIPS. The > > phones > > are registered to OpenSIPs. > > > > INVITE --> lookup ----> 1. Destination > > | > > \--> 2. Destination > > > > When the call is terminated by the caller, the BYE request shall > > take > > the same path. Currently, the BYE is sent from the PBX directly to > > the > > Contact URI (which is not reachable by the PBX). > > > > Is it possible to use record_route in the branch_route so that > > different record route headers are used? Or is there another way? > > > > Thanks in advance, > > > > Karsten > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From kurgan-rus at inbox.ru Fri Apr 1 12:19:29 2022 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Fri, 01 Apr 2022 15:19:29 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?RTP_Engine_rpms_for_Centos_7?= Message-ID: <1648815569.15737670@f187.i.mail.ru> Hi list,   a small contribution to the community, not as a yum-repository yet, but still.   https://alexeyka.zantsev.com/rpm/   Often creating rpm packages of RTPEngine becomes a headache, so I decided to share.   Hope this will be useful!   ----------------------------------------------- BR, Alexey https://alexeyka.zantsev.com/   -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 1 13:48:08 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 1 Apr 2022 16:48:08 +0300 Subject: [OpenSIPS-Users] OpenSIPS timers In-Reply-To: References: Message-ID: <5225093f-65d7-f259-c9eb-9ec4e2c55123@opensips.org> Hi Ovidiu, Originally you mentioned tm-utimer, now tm-timer....which one is ? As it is very important. When increasing the timer_partitions, what you mean by "instability" of the system? Yes, in the reactor, the UDP workers may handle timer jobs also beside the UDP traffic. While the timer procs are 100% dedicated to the timer jobs only. So yes, if the workers are idle, they can act as any timer procs also. Increasing the TM_TABLE_ENTRIES should not impact too much, at the performance over the timer lists (in TM) has nothing to do with the size of the hash table. I will check the mentioned ticket, but if what you are saying is true on the HP malloc, it means the bottle neck is actually in the ops on the shared memory. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/1/22 12:31 AM, Ovidiu Sas wrote: > Hello Bogdan, > > Thank you for looking into this! > > I get warnings mostly from tm-timer. I've seen warnings from > blcore-expire, dlg-options-pinger, dlg-reinvite-pinger, dlg-timer (in > the logs, but not during my testing). > While testing, I saw only the tm-timer warnings. > > I took a superficial look at the "timer_partitions" and your > explanation matches my findings. However, increasing the > "timer_partitions" makes the system unstable (doesn't matter how many > timer procs we have). > I found that I can get the most out of the system if one > "timer_partiton" is used along with one timer_proc. > > With the reactor scheme, a UDP receiver can handle timer jobs, is that > right? If yes, if the UDP workers are idle, there are enough resources > to handle timer jobs, correct? > > I was also increasing the TM_TABLE_ENTRIES to (1<<18) and there was a > little bit of performance increase, but I will need to test more to > come up with a valid conclusion. > > On the other hand, I noticed a strange behavior on timer handling. > Take a look at: > https://github.com/OpenSIPS/opensips/issues/2797 > Not sure if this is related to the warnings that I'm seeing. > > The biggest performance improvement was switching to HP_MALLOC for > both pkg and shm memory. > > I will keep you posted with my findings, > Ovidiu > > On Thu, Mar 31, 2022 at 10:28 AM Bogdan-Andrei Iancu > wrote: >> Hi Ovidiu, >> >> As warnings from the timer_ticker, do you get only for the tm-utimer >> task ? I'm asking as the key question here is where the bottleneck is : >> in the whole "timer" subsystem, or in the tm-utimer task only? >> >> The TM "timer_partitions" creates multiple parallel timer lists, to >> avoid having large "amounts" of transactions handled at a moment in a >> single tm-utimer task (but rather split/partition the whole of amount of >> handled transactions into smaller chunks, to be handled one at a time in >> the timer task. >> >> The "timer_workers" creates more than one dedicated processes for >> handling the timer tasks (so scales up the timer sub-system). >> >> If you get warnings only on tm-utimer, I suspect the bottleneck is TM >> related, mainly on performing re-transmissions (that's what that task is >> doing). So the increasing the timer-partitions should be the way to help. >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 3/24/22 12:54 AM, Ovidiu Sas wrote: >>> Hello all, >>> >>> I'm working on tuning an opensips server. I get this pesky: >>> WARNING:core:utimer_ticker: utimer task already scheduled >>> I was trying to get rid of them by playing with the tm >>> timer_partitions parameter and the timer_workers core param. >>> By increasing any of them doesn't increase performance. >>> By increasing both of them, it actually decreases performance. >>> The server is not at limit, the load on the UDP workers is around >>> 50-60 with some spikes. >>> I have around 3500+ cps sipp traffic. >>> >>> My understanding is that by increasing the number of timer_partitions, >>> we will have more procs walking in parallel over the timer structures. >>> If we have on timer structure, we have one proc walking over it. >>> How is this working for two timer structures? What is the difference >>> between the first and the second timer structure? Should we expect >>> less work for each proc? >>> >>> For now, to reduce the occurrence of the warning log, I increased the >>> timer interval for tm-utimer from 100ms to 200ms. This should be ok as >>> the timer has the TIMER_FLAG_DELAY_ON_DELAY flag set. >>> >>> Thanks, >>> Ovidiu >>> > From bogdan at opensips.org Fri Apr 1 14:04:24 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 1 Apr 2022 17:04:24 +0300 Subject: [OpenSIPS-Users] Call forking, branches, Record-routing In-Reply-To: <80730f412ba7559ccd1de5459fdf7ce154992cf4.camel@gmx.de> References: <45a50a6c-0aec-6862-3710-8dbf2085ec8e@opensips.org> <0239fc41a5d0265deeb86562ccc7e57d0a819050.camel@gmx.de> <80730f412ba7559ccd1de5459fdf7ce154992cf4.camel@gmx.de> Message-ID: <8d36a05b-6534-094a-82c3-56a040fe5ea6@opensips.org> Oh, so you have traffic both from public and private network, right ? If so, you have 2 options: 1) use a single socket, without advertise and use the script advertise function depending on the source of the call - see set_advertised_address() [1] 2) use 2 sockets, one for public traffic, with advertise and one for the internal traffic, without advertise. For each call, control the outbound interface via the $socket_out variable [2] My 2 cents, (1) may be too complicated as the set_advertised_address() function as a message level scope, so you need to take care and do it for each and every single request of the call. With (2) you need to do it only for the initial request (INVITE) as the sockets are remembered for the whole duration of the call by the Record Route mechanism. [1] https://www.opensips.org/Documentation/Script-CoreFunctions-3-2#set_advertised_address [2] https://opensips.org/Documentation/Script-CoreVar-3-2#socket_out Best Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 3/31/22 7:27 PM, Karsten Wemheuer wrote: > Hi Bogdan-Andrei, > > in case of global advertising is active and set to the natted address > the advertised address is used, but this leads to problems using phones > in the LAN. > > As written in my other post: Without setting the advertise address and > port, I have a problem with the phones behind NAT. Is it possible to > manipulate the route before in a branch or something like that? > > Regards, > > Karsten > > Am Donnerstag, dem 31.03.2022 um 18:53 +0300 schrieb Bogdan-Andrei > Iancu: >> Hi Karsten, >> >> You say the record_route() does not take into consideration the >> global >> advertising ?? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 3/31/22 6:22 PM, Karsten Wemheuer wrote: >>> Hi Bogdan-Andrei, >>> >>> many thanks for Your help. >>> >>> I tried with record_route. It doesn't work for me, as I set >>> "advertised_address" and "advertised_port" to the natted address of >>> the >>> (only) interface. I wasn't able to avoid this. It seemed to be >>> required >>> to be able to reflect the path "phone -> proxy -> pbx". >>> >>> I removed the "advertised"-stuff and checked again the call with >>> record_route. Now this seems to work. >>> >>> I think, I have to fix the other call flow to avoid the global >>> setting >>> of the advertised address and port. >>> >>> Best regards, >>> >>> Karsten >>> >>> Am Donnerstag, dem 31.03.2022 um 17:44 +0300 schrieb Bogdan-Andrei >>> Iancu: >>>> Hi Karsten, >>>> >>>> See my prev email, just to record_route() before the t_relay() >>>> for >>>> the >>>> initial INVITE. And the loose_route() stuff for whatever >>>> sequential/in-dialog requests. >>>> >>>> Best regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>>> >>>> On 3/31/22 2:50 PM, Karsten Wemheuer wrote: >>>>> Hi*, >>>>> >>>>> I have a understanding problem regarding branches and call >>>>> forking. >>>>> A call from a PBX is to be routed to phone(s) via OpenSIPS. The >>>>> phones >>>>> are registered to OpenSIPs. >>>>> >>>>> INVITE --> lookup ----> 1. Destination >>>>> | >>>>> \--> 2. Destination >>>>> >>>>> When the call is terminated by the caller, the BYE request >>>>> shall >>>>> take >>>>> the same path. Currently, the BYE is sent from the PBX directly >>>>> to >>>>> the >>>>> Contact URI (which is not reachable by the PBX). >>>>> >>>>> Is it possible to use record_route in the branch_route so that >>>>> different record route headers are used? Or is there another >>>>> way? >>>>> >>>>> Thanks in advance, >>>>> >>>>> Karsten >>>>> >>>>> >>>>> _______________________________________________ >>>>> 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 From osas at voipembedded.com Fri Apr 1 14:10:29 2022 From: osas at voipembedded.com (Ovidiu Sas) Date: Fri, 1 Apr 2022 10:10:29 -0400 Subject: [OpenSIPS-Users] OpenSIPS timers In-Reply-To: <5225093f-65d7-f259-c9eb-9ec4e2c55123@opensips.org> References: <5225093f-65d7-f259-c9eb-9ec4e2c55123@opensips.org> Message-ID: Hello Bogdan, During my test, it was tm-utimer only. It was a typo on my side. I also see in the logs from time to time the other timers too, including tm-timer. What I noticed in my tests is that as soon as I increase the timer_partitions, the system is able to handle less cps until workers are becoming 100% loaded and calls starts failing (due to retransmissions and udp queue being full - the udp queue is quite big to accommodate spikes). Is there a way to make the timer lists more efficient (in terms of ops in shared memory)? Please take a look at the mentioned ticket as it makes the ratelimit module unusable (and maybe with side effects for other modules that require accurate timeslots). Basically, for a timer that is supposed to fire every second, the observed behaviour is that the timer fires at approx 1s (or less by a few ms) and then from time to time it fires at 1.8s and the cycle repeats. Thanks, Ovidiu On Fri, Apr 1, 2022 at 9:48 AM Bogdan-Andrei Iancu wrote: > > Hi Ovidiu, > > Originally you mentioned tm-utimer, now tm-timer....which one is ? As it > is very important. > > When increasing the timer_partitions, what you mean by "instability" of > the system? > > Yes, in the reactor, the UDP workers may handle timer jobs also beside > the UDP traffic. While the timer procs are 100% dedicated to the timer > jobs only. So yes, if the workers are idle, they can act as any timer > procs also. > > Increasing the TM_TABLE_ENTRIES should not impact too much, at the > performance over the timer lists (in TM) has nothing to do with the size > of the hash table. > > I will check the mentioned ticket, but if what you are saying is true on > the HP malloc, it means the bottle neck is actually in the ops on the > shared memory. > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/1/22 12:31 AM, Ovidiu Sas wrote: > > Hello Bogdan, > > > > Thank you for looking into this! > > > > I get warnings mostly from tm-timer. I've seen warnings from > > blcore-expire, dlg-options-pinger, dlg-reinvite-pinger, dlg-timer (in > > the logs, but not during my testing). > > While testing, I saw only the tm-timer warnings. > > > > I took a superficial look at the "timer_partitions" and your > > explanation matches my findings. However, increasing the > > "timer_partitions" makes the system unstable (doesn't matter how many > > timer procs we have). > > I found that I can get the most out of the system if one > > "timer_partiton" is used along with one timer_proc. > > > > With the reactor scheme, a UDP receiver can handle timer jobs, is that > > right? If yes, if the UDP workers are idle, there are enough resources > > to handle timer jobs, correct? > > > > I was also increasing the TM_TABLE_ENTRIES to (1<<18) and there was a > > little bit of performance increase, but I will need to test more to > > come up with a valid conclusion. > > > > On the other hand, I noticed a strange behavior on timer handling. > > Take a look at: > > https://github.com/OpenSIPS/opensips/issues/2797 > > Not sure if this is related to the warnings that I'm seeing. > > > > The biggest performance improvement was switching to HP_MALLOC for > > both pkg and shm memory. > > > > I will keep you posted with my findings, > > Ovidiu > > > > On Thu, Mar 31, 2022 at 10:28 AM Bogdan-Andrei Iancu > > wrote: > >> Hi Ovidiu, > >> > >> As warnings from the timer_ticker, do you get only for the tm-utimer > >> task ? I'm asking as the key question here is where the bottleneck is : > >> in the whole "timer" subsystem, or in the tm-utimer task only? > >> > >> The TM "timer_partitions" creates multiple parallel timer lists, to > >> avoid having large "amounts" of transactions handled at a moment in a > >> single tm-utimer task (but rather split/partition the whole of amount of > >> handled transactions into smaller chunks, to be handled one at a time in > >> the timer task. > >> > >> The "timer_workers" creates more than one dedicated processes for > >> handling the timer tasks (so scales up the timer sub-system). > >> > >> If you get warnings only on tm-utimer, I suspect the bottleneck is TM > >> related, mainly on performing re-transmissions (that's what that task is > >> doing). So the increasing the timer-partitions should be the way to help. > >> > >> Best regards, > >> > >> Bogdan-Andrei Iancu > >> > >> OpenSIPS Founder and Developer > >> https://www.opensips-solutions.com > >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 > >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > >> > >> On 3/24/22 12:54 AM, Ovidiu Sas wrote: > >>> Hello all, > >>> > >>> I'm working on tuning an opensips server. I get this pesky: > >>> WARNING:core:utimer_ticker: utimer task already scheduled > >>> I was trying to get rid of them by playing with the tm > >>> timer_partitions parameter and the timer_workers core param. > >>> By increasing any of them doesn't increase performance. > >>> By increasing both of them, it actually decreases performance. > >>> The server is not at limit, the load on the UDP workers is around > >>> 50-60 with some spikes. > >>> I have around 3500+ cps sipp traffic. > >>> > >>> My understanding is that by increasing the number of timer_partitions, > >>> we will have more procs walking in parallel over the timer structures. > >>> If we have on timer structure, we have one proc walking over it. > >>> How is this working for two timer structures? What is the difference > >>> between the first and the second timer structure? Should we expect > >>> less work for each proc? > >>> > >>> For now, to reduce the occurrence of the warning log, I increased the > >>> timer interval for tm-utimer from 100ms to 200ms. This should be ok as > >>> the timer has the TIMER_FLAG_DELAY_ON_DELAY flag set. > >>> > >>> Thanks, > >>> Ovidiu > >>> > > > -- VoIP Embedded, Inc. http://www.voipembedded.com From kwem at gmx.de Fri Apr 1 15:13:14 2022 From: kwem at gmx.de (Karsten Wemheuer) Date: Fri, 01 Apr 2022 17:13:14 +0200 Subject: [OpenSIPS-Users] Call forking, branches, Record-routing In-Reply-To: <735693e7-ffe7-07c4-86fe-271de5d5dc32@opensips.org> References: <45a50a6c-0aec-6862-3710-8dbf2085ec8e@opensips.org> <0239fc41a5d0265deeb86562ccc7e57d0a819050.camel@gmx.de> <80730f412ba7559ccd1de5459fdf7ce154992cf4.camel@gmx.de> <8d36a05b-6534-094a-82c3-56a040fe5ea6@opensips.org> <735693e7-ffe7-07c4-86fe-271de5d5dc32@opensips.org> Message-ID: <697e552aa5651c1ea9f0ab78e1dd5adb9aded7d0.camel@gmx.de> Yes, I originally assumed that I could not use two TLS sockets on one interface. My last sentence in the last post showed that I understood it now. As you said, I now have port forwarding to another port and two separate sockets for public and private phones. socket = udp:10.0.2.3 socket = tls:10.0.2.3:5061 socket = tls:10.0.2.3:5062 as PUBLIC-IP:PUBLIC-PORT It seems to work now. I now have to check the other cases. Many thanks for the help and patience. Have a nice weekend! Best regards, Karsten Am Freitag, dem 01.04.2022 um 17:55 +0300 schrieb Bogdan-Andrei Iancu: > Just to be sure, by sockets I mean OpenSIPS listeners. Even with one > IP, > you can have something like PUB:5060->NAT->PRIV:5070 and have > OpenSIPS > listening on PRIV:5070 (with advertise for external traffic) and > PRIV:5060 for internal. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/1/22 5:38 PM, Karsten Wemheuer wrote: > > Hi Bogdan-Andrei, > > > > unfortunately, only one interface is available for communication to > > the > > phones. > > > > For the single interface I have two sockets, one for UDP and one > > for > > TLS. TLS is used to access local phones and those over NAT. > > > > listen = udp:10.0.2.3 > > listen = tls:10.0.2.3:5061 > > > > The way that doesn't work is a call coming from the PBX via UDP and > > going to a phone via TLS behind a NAT. How can I influence the > > CANCEL > > request here, especially if it comes internally from OpenSIPs? > > > > Maybe I should use another port for the public side. Than it should > > be > > possible to use two sockets for tls. > > > > Thanks again. > > > > Best regards, > > Karsten > > > > Am Freitag, dem 01.04.2022 um 17:04 +0300 schrieb Bogdan-Andrei > > Iancu: > > > Oh, so you have traffic both from public and private network, > > > right ? > > > If > > > so, you have 2 options: > > > > > > 1) use a single socket, without advertise and use the script > > > advertise > > > function depending on the source of the call - see > > > set_advertised_address() [1] > > > > > > 2) use 2 sockets, one for public traffic, with advertise and one > > > for > > > the > > > internal traffic, without advertise. For each call, control the > > > outbound > > > interface via the $socket_out variable [2] > > > > > > My 2 cents, (1) may be too complicated as the > > > set_advertised_address() > > > function as a message level scope, so you need to take care and > > > do > > > it > > > for each and every single request of the call. With (2) you need > > > to > > > do > > > it only for the initial request (INVITE) as the sockets are > > > remembered > > > for the whole duration of the call by the Record Route mechanism. > > > > > > [1] > > > https://www.opensips.org/Documentation/Script-CoreFunctions-3-2#set_advertised_address > > > [2] > > > https://opensips.org/Documentation/Script-CoreVar-3-2#socket_out > > > > > > Best Regards, > > > > > > Bogdan-Andrei Iancu > > > > > > OpenSIPS Founder and Developer > > > https://www.opensips-solutions.com > > > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > > > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > > > > > On 3/31/22 7:27 PM, Karsten Wemheuer wrote: > > > > Hi Bogdan-Andrei, > > > > > > > > in case of global advertising is active and set to the natted > > > > address > > > > the advertised address is used, but this leads to problems > > > > using > > > > phones > > > > in the LAN. > > > > > > > > As written in my other post: Without setting the advertise > > > > address > > > > and > > > > port, I have a problem with the phones behind NAT. Is it > > > > possible > > > > to > > > > manipulate the route before in a branch or something like that? > > > > > > > > Regards, > > > > > > > > Karsten > > > > > > > > Am Donnerstag, dem 31.03.2022 um 18:53 +0300 schrieb Bogdan- > > > > Andrei > > > > Iancu: > > > > > Hi Karsten, > > > > > > > > > > You say the record_route() does not take into consideration > > > > > the > > > > > global > > > > > advertising ?? > > > > > > > > > > Regards, > > > > > > > > > > Bogdan-Andrei Iancu > > > > > > > > > > OpenSIPS Founder and Developer > > > > > https://www.opensips-solutions.com > > > > > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > > > > > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > > > > > > > > > On 3/31/22 6:22 PM, Karsten Wemheuer wrote: > > > > > > Hi Bogdan-Andrei, > > > > > > > > > > > > many thanks for Your help. > > > > > > > > > > > > I tried with record_route. It doesn't work for me, as I set > > > > > > "advertised_address" and "advertised_port" to the natted > > > > > > address of > > > > > > the > > > > > > (only) interface. I wasn't able to avoid this. It seemed to > > > > > > be > > > > > > required > > > > > > to be able to reflect the path "phone -> proxy -> pbx". > > > > > > > > > > > > I removed the "advertised"-stuff and checked again the call > > > > > > with > > > > > > record_route. Now this seems to work. > > > > > > > > > > > > I think, I have to fix the other call flow to avoid the > > > > > > global > > > > > > setting > > > > > > of the advertised address and port. > > > > > > > > > > > > Best regards, > > > > > > > > > > > > Karsten > > > > > > > > > > > > Am Donnerstag, dem 31.03.2022 um 17:44 +0300 schrieb > > > > > > Bogdan- > > > > > > Andrei > > > > > > Iancu: > > > > > > > Hi Karsten, > > > > > > > > > > > > > > See my prev email, just to record_route() before the > > > > > > > t_relay() > > > > > > > for > > > > > > > the > > > > > > > initial INVITE. And the loose_route() stuff for whatever > > > > > > > sequential/in-dialog requests. > > > > > > > > > > > > > > Best regards, > > > > > > > > > > > > > > Bogdan-Andrei Iancu > > > > > > > > > > > > > > OpenSIPS Founder and Developer > > > > > > > https://www.opensips-solutions.com > > > > > > > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > > > > > > > > > > > > > > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > > > > > > > > > > > > > On 3/31/22 2:50 PM, Karsten Wemheuer wrote: > > > > > > > > Hi*, > > > > > > > > > > > > > > > > I have a understanding problem regarding branches and > > > > > > > > call > > > > > > > > forking. > > > > > > > > A call from a PBX is to be routed to phone(s) via > > > > > > > > OpenSIPS. > > > > > > > > The > > > > > > > > phones > > > > > > > > are registered to OpenSIPs. > > > > > > > > > > > > > > > > INVITE --> lookup ----> 1. Destination > > > > > > > > | > > > > > > > > \--> 2. Destination > > > > > > > > > > > > > > > > When the call is terminated by the caller, the BYE > > > > > > > > request > > > > > > > > shall > > > > > > > > take > > > > > > > > the same path. Currently, the BYE is sent from the PBX > > > > > > > > directly > > > > > > > > to > > > > > > > > the > > > > > > > > Contact URI (which is not reachable by the PBX). > > > > > > > > > > > > > > > > Is it possible to use record_route in the branch_route > > > > > > > > so > > > > > > > > that > > > > > > > > different record route headers are used? Or is there > > > > > > > > another > > > > > > > > way? > > > > > > > > > > > > > > > > Thanks in advance, > > > > > > > > > > > > > > > > Karsten > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > > 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 From ahrongreenberg at gmail.com Sun Apr 3 22:55:54 2022 From: ahrongreenberg at gmail.com (Ahron Greenberg) Date: Sun, 3 Apr 2022 18:55:54 -0400 Subject: [OpenSIPS-Users] Replace part of Remote-Party-ID header Message-ID: <7AB98086-9C2A-45E7-B309-C78F129E5E4B@hxcore.ol> An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 4 16:30:56 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 4 Apr 2022 19:30:56 +0300 Subject: [OpenSIPS-Users] Replace part of Remote-Party-ID header In-Reply-To: <7AB98086-9C2A-45E7-B309-C78F129E5E4B@hxcore.ol> References: <7AB98086-9C2A-45E7-B309-C78F129E5E4B@hxcore.ol> Message-ID: <0e599d90-8767-c6e9-9b07-97490b2fb9cd@opensips.org> Hi Ahron, The easiest way is to extract the whole RPID hdr body via $hdr(Remote-Party-Id) and run a regexp [1] on it to replace the `privacy=full` with `privacy=off` textually. After replace the old value with the new one by using the remove_hf() and append_hf() functions. [1] https://www.opensips.org/Documentation/Script-Tran-3-2#re.subst Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/4/22 1:55 AM, Ahron Greenberg wrote: > > Hi all, > > How would I replace just the privacy value of the RPID header? > > Ahron Greenberg > > > _______________________________________________ > 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 hobe69 at hotmail.com Tue Apr 5 20:31:27 2022 From: hobe69 at hotmail.com (Bela H) Date: Tue, 5 Apr 2022 20:31:27 +0000 Subject: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? Message-ID: Hello, I want to set up a call forwarding and followed the instructions from the "Building telephony systems with OpenSIPS". However, this is a little bit outdated, the old opensipsctl was replaced by opensips-cli. How can I add an avp data into a usr_preferences table from opensips-cli? This was the old format: opensipsctl avp add A_number callfwd 0 C_number Cheers, Bela -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 6 06:25:16 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 6 Apr 2022 09:25:16 +0300 Subject: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? In-Reply-To: References: Message-ID: <9de95be3-c24d-f7ab-114a-4e5cb6693787@opensips.org> Hi Bela, The opensips-cli does not offer anymore the pure DB oriented ops (like the opensipsctl did). The actually /opensipsctl avp add A_number callfwd 0 C_number/ translates into the query: INSERT INTO  usr_preferences ( uuid, username, domain, attribute, type, value, last_modified) VALUES ('A_number','','','callfwd', 0,'C_number',NOW()); Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/5/22 11:31 PM, Bela H wrote: > Hello, > > I want to set up a call forwarding and followed the instructions from > the "Building telephony systems with OpenSIPS". However, this is a > little bit outdated, the old opensipsctl was replaced by opensips-cli. > How can I add an avp data into a usr_preferences table from opensips-cli? > This was the old format: /opensipsctl avp add A_number callfwd 0 C_number/ > > Cheers, > Bela > > > _______________________________________________ > 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 hobe69 at hotmail.com Wed Apr 6 08:48:55 2022 From: hobe69 at hotmail.com (Bela H) Date: Wed, 6 Apr 2022 08:48:55 +0000 Subject: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? In-Reply-To: <9de95be3-c24d-f7ab-114a-4e5cb6693787@opensips.org> References: <9de95be3-c24d-f7ab-114a-4e5cb6693787@opensips.org> Message-ID: Hi Bogdan, Great thanks for your quick answer. I got that workaround as well, however that would have been my next question what you already answered: using the A number in uuid field instead of username 😉 Cheers, Bela From: Bogdan-Andrei Iancu Sent: Wednesday, 6 April 2022 18:25 To: OpenSIPS users mailling list; Bela H Subject: Re: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? Hi Bela, The opensips-cli does not offer anymore the pure DB oriented ops (like the opensipsctl did). The actually opensipsctl avp add A_number callfwd 0 C_number translates into the query: INSERT INTO usr_preferences ( uuid, username, domain, attribute, type, value, last_modified) VALUES ('A_number','','','callfwd', 0,'C_number',NOW()); Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/5/22 11:31 PM, Bela H wrote: Hello, I want to set up a call forwarding and followed the instructions from the "Building telephony systems with OpenSIPS". However, this is a little bit outdated, the old opensipsctl was replaced by opensips-cli. How can I add an avp data into a usr_preferences table from opensips-cli? This was the old format: opensipsctl avp add A_number callfwd 0 C_number Cheers, Bela _______________________________________________ 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 Apr 6 09:48:48 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 6 Apr 2022 12:48:48 +0300 Subject: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? In-Reply-To: References: <9de95be3-c24d-f7ab-114a-4e5cb6693787@opensips.org> Message-ID: <15f1c1ee-210c-38d3-a904-f87a4b98b237@opensips.org> Hi Bela, If you should put the A number in UUID or user+domain columns depends 100% on how you use the avp_db_load() from script, mainly what is that first param - an uuid or user. See https://opensips.org/html/docs/modules/3.2.x/avpops.html#func_avp_db_load, the "source" param Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/6/22 11:48 AM, Bela H wrote: > > Hi Bogdan, > > Great thanks for your quick answer. > > I got that workaround as well, however that would have been my next > question what you already answered: using the A number in uuid field > instead of  username 😉 > > Cheers, > > Bela > > *From: *Bogdan-Andrei Iancu > *Sent: *Wednesday, 6 April 2022 18:25 > *To: *OpenSIPS users mailling list ; > Bela H > *Subject: *Re: [OpenSIPS-Users] How to use old "opensipsctl avp add" > with opensips-cli? > > Hi Bela, > > The opensips-cli does not offer anymore the pure DB oriented ops (like > the opensipsctl did). The actually /opensipsctl avp add A_number > callfwd 0 C_number/ translates into the query: > > INSERT INTO  usr_preferences > ( uuid, username, domain, attribute, type, value, last_modified) > VALUES > ('A_number','','','callfwd', 0,'C_number',NOW()); > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/5/22 11:31 PM, Bela H wrote: > > Hello, > > I want to set up a call forwarding and followed the instructions > from the "Building telephony systems with OpenSIPS". However, this > is a little bit outdated, the old opensipsctl was replaced by > opensips-cli. How can I add an avp data into a usr_preferences > table from opensips-cli? > > This was the old format: /opensipsctl avp add A_number callfwd 0 > C_number/ > > Cheers, > Bela > > > > _______________________________________________ > > 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 Apr 6 13:54:47 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 6 Apr 2022 16:54:47 +0300 Subject: [OpenSIPS-Users] OpenSIPS 3.3 - interim update Message-ID: <109e8f8c-bead-7b40-7580-ef13d49403ee@opensips.org> The time for releasing 3.3 is rapidly approaching! Learn what was already done, what is left to be done and when it will be ready ! https://blog.opensips.org/2022/04/06/opensips-3-3-interim-update/ Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ From hobe69 at hotmail.com Wed Apr 6 21:27:43 2022 From: hobe69 at hotmail.com (Bela H) Date: Wed, 6 Apr 2022 21:27:43 +0000 Subject: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? In-Reply-To: <15f1c1ee-210c-38d3-a904-f87a4b98b237@opensips.org> References: <9de95be3-c24d-f7ab-114a-4e5cb6693787@opensips.org> <15f1c1ee-210c-38d3-a904-f87a4b98b237@opensips.org> Message-ID: Thanks Bogdan again! I am using that function but somehow didn't work with avp_db_load("$rU",$avp(callfwd)). With uuid it is all good! Cheers, Bela ________________________________ From: Bogdan-Andrei Iancu Sent: Wednesday, 6 April 2022 21:48 To: Bela H ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? Hi Bela, If you should put the A number in UUID or user+domain columns depends 100% on how you use the avp_db_load() from script, mainly what is that first param - an uuid or user. See https://opensips.org/html/docs/modules/3.2.x/avpops.html#func_avp_db_load, the "source" param Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/6/22 11:48 AM, Bela H wrote: Hi Bogdan, Great thanks for your quick answer. I got that workaround as well, however that would have been my next question what you already answered: using the A number in uuid field instead of username 😉 Cheers, Bela From: Bogdan-Andrei Iancu Sent: Wednesday, 6 April 2022 18:25 To: OpenSIPS users mailling list; Bela H Subject: Re: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? Hi Bela, The opensips-cli does not offer anymore the pure DB oriented ops (like the opensipsctl did). The actually opensipsctl avp add A_number callfwd 0 C_number translates into the query: INSERT INTO usr_preferences ( uuid, username, domain, attribute, type, value, last_modified) VALUES ('A_number','','','callfwd', 0,'C_number',NOW()); Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/5/22 11:31 PM, Bela H wrote: Hello, I want to set up a call forwarding and followed the instructions from the "Building telephony systems with OpenSIPS". However, this is a little bit outdated, the old opensipsctl was replaced by opensips-cli. How can I add an avp data into a usr_preferences table from opensips-cli? This was the old format: opensipsctl avp add A_number callfwd 0 C_number Cheers, Bela _______________________________________________ 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 Mon Apr 11 12:40:36 2022 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 11 Apr 2022 15:40:36 +0300 Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 3.1.9 and 3.2.6 minor releases Message-ID: <14e04ac8-e54e-2fe1-22c6-8854e74ad712@opensips.org> Hi, everyone! The 3.1.9 and 3.2.6 OpenSIPS minor versions are scheduled for release on Wed, April 20th -- roughly a week and a half from now. In preparation for the release, we have already imposed a freeze on any significant fixes (as volume) on these two stable branches, in order to ensure a safe window for testing in the days ahead. Best regards, -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS eBootcamp May 23 - June 3 | www.opensips.org/training From osas at voipembedded.com Tue Apr 12 03:43:04 2022 From: osas at voipembedded.com (Ovidiu Sas) Date: Mon, 11 Apr 2022 23:43:04 -0400 Subject: [OpenSIPS-Users] OpenSIPS timers In-Reply-To: References: <5225093f-65d7-f259-c9eb-9ec4e2c55123@opensips.org> Message-ID: Just to conclude the thread. The issue here was high load combined with the fact that tm has two timers (a second based timer *tm-timer* that runs every second and millisecond based timer *tm-utimer* that runs every 200ms). Both timers are protected by the same lock and the timers cannot run in parallel. The second based timer *tm-timer* sometimes takes more then 200ms to complete which prevents the millisecond based timer *tm-utimer* to be executed in its 200ms window. -ovidiu On Fri, Apr 1, 2022 at 10:10 AM Ovidiu Sas wrote: > > Hello Bogdan, > > During my test, it was tm-utimer only. It was a typo on my side. > > I also see in the logs from time to time the other timers too, > including tm-timer. > > What I noticed in my tests is that as soon as I increase the > timer_partitions, the system is able to handle less cps until workers > are becoming 100% loaded and calls starts failing (due to > retransmissions and udp queue being full - the udp queue is quite big > to accommodate spikes). > > Is there a way to make the timer lists more efficient (in terms of ops > in shared memory)? > > Please take a look at the mentioned ticket as it makes the ratelimit > module unusable (and maybe with side effects for other modules that > require accurate timeslots). > Basically, for a timer that is supposed to fire every second, the > observed behaviour is that the timer fires at approx 1s (or less by a > few ms) and then from time to time it fires at 1.8s and the cycle > repeats. > > Thanks, > Ovidiu > > On Fri, Apr 1, 2022 at 9:48 AM Bogdan-Andrei Iancu wrote: > > > > Hi Ovidiu, > > > > Originally you mentioned tm-utimer, now tm-timer....which one is ? As it > > is very important. > > > > When increasing the timer_partitions, what you mean by "instability" of > > the system? > > > > Yes, in the reactor, the UDP workers may handle timer jobs also beside > > the UDP traffic. While the timer procs are 100% dedicated to the timer > > jobs only. So yes, if the workers are idle, they can act as any timer > > procs also. > > > > Increasing the TM_TABLE_ENTRIES should not impact too much, at the > > performance over the timer lists (in TM) has nothing to do with the size > > of the hash table. > > > > I will check the mentioned ticket, but if what you are saying is true on > > the HP malloc, it means the bottle neck is actually in the ops on the > > shared memory. > > > > Best regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > > > On 4/1/22 12:31 AM, Ovidiu Sas wrote: > > > Hello Bogdan, > > > > > > Thank you for looking into this! > > > > > > I get warnings mostly from tm-timer. I've seen warnings from > > > blcore-expire, dlg-options-pinger, dlg-reinvite-pinger, dlg-timer (in > > > the logs, but not during my testing). > > > While testing, I saw only the tm-timer warnings. > > > > > > I took a superficial look at the "timer_partitions" and your > > > explanation matches my findings. However, increasing the > > > "timer_partitions" makes the system unstable (doesn't matter how many > > > timer procs we have). > > > I found that I can get the most out of the system if one > > > "timer_partiton" is used along with one timer_proc. > > > > > > With the reactor scheme, a UDP receiver can handle timer jobs, is that > > > right? If yes, if the UDP workers are idle, there are enough resources > > > to handle timer jobs, correct? > > > > > > I was also increasing the TM_TABLE_ENTRIES to (1<<18) and there was a > > > little bit of performance increase, but I will need to test more to > > > come up with a valid conclusion. > > > > > > On the other hand, I noticed a strange behavior on timer handling. > > > Take a look at: > > > https://github.com/OpenSIPS/opensips/issues/2797 > > > Not sure if this is related to the warnings that I'm seeing. > > > > > > The biggest performance improvement was switching to HP_MALLOC for > > > both pkg and shm memory. > > > > > > I will keep you posted with my findings, > > > Ovidiu > > > > > > On Thu, Mar 31, 2022 at 10:28 AM Bogdan-Andrei Iancu > > > wrote: > > >> Hi Ovidiu, > > >> > > >> As warnings from the timer_ticker, do you get only for the tm-utimer > > >> task ? I'm asking as the key question here is where the bottleneck is : > > >> in the whole "timer" subsystem, or in the tm-utimer task only? > > >> > > >> The TM "timer_partitions" creates multiple parallel timer lists, to > > >> avoid having large "amounts" of transactions handled at a moment in a > > >> single tm-utimer task (but rather split/partition the whole of amount of > > >> handled transactions into smaller chunks, to be handled one at a time in > > >> the timer task. > > >> > > >> The "timer_workers" creates more than one dedicated processes for > > >> handling the timer tasks (so scales up the timer sub-system). > > >> > > >> If you get warnings only on tm-utimer, I suspect the bottleneck is TM > > >> related, mainly on performing re-transmissions (that's what that task is > > >> doing). So the increasing the timer-partitions should be the way to help. > > >> > > >> Best regards, > > >> > > >> Bogdan-Andrei Iancu > > >> > > >> OpenSIPS Founder and Developer > > >> https://www.opensips-solutions.com > > >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 > > >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > >> > > >> On 3/24/22 12:54 AM, Ovidiu Sas wrote: > > >>> Hello all, > > >>> > > >>> I'm working on tuning an opensips server. I get this pesky: > > >>> WARNING:core:utimer_ticker: utimer task already scheduled > > >>> I was trying to get rid of them by playing with the tm > > >>> timer_partitions parameter and the timer_workers core param. > > >>> By increasing any of them doesn't increase performance. > > >>> By increasing both of them, it actually decreases performance. > > >>> The server is not at limit, the load on the UDP workers is around > > >>> 50-60 with some spikes. > > >>> I have around 3500+ cps sipp traffic. > > >>> > > >>> My understanding is that by increasing the number of timer_partitions, > > >>> we will have more procs walking in parallel over the timer structures. > > >>> If we have on timer structure, we have one proc walking over it. > > >>> How is this working for two timer structures? What is the difference > > >>> between the first and the second timer structure? Should we expect > > >>> less work for each proc? > > >>> > > >>> For now, to reduce the occurrence of the warning log, I increased the > > >>> timer interval for tm-utimer from 100ms to 200ms. This should be ok as > > >>> the timer has the TIMER_FLAG_DELAY_ON_DELAY flag set. > > >>> > > >>> Thanks, > > >>> Ovidiu > > >>> > > > > > > > > -- > VoIP Embedded, Inc. > http://www.voipembedded.com -- VoIP Embedded, Inc. http://www.voipembedded.com From alain.bieuzent at free.fr Tue Apr 12 07:16:59 2022 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Tue, 12 Apr 2022 09:16:59 +0200 Subject: [OpenSIPS-Users] avp_db_query retcode value Message-ID: Hi All, Where can i found the different return value of $retcode after an avp_db_query. I need to distinguish a null result from a failed request Regards PS : I’m using opensips 3.2 -------------- next part -------------- An HTML attachment was scrubbed... URL: From efes99999 at gmail.com Tue Apr 12 16:50:08 2022 From: efes99999 at gmail.com (L S) Date: Tue, 12 Apr 2022 12:50:08 -0400 Subject: [OpenSIPS-Users] TLS issue Message-ID: We have been using Opensips 1.11 tls successfully for a while. In our setup, Opensips is used as a dispatcher to a couple of Asterisk servers. Some of the sip clients (mostly Cisco SPA phones) communicate with Opensips server over TLS, some don't. Once in a while internet goes down and the sip clients go offline as a result. When the connection is restored, the phones that are not on TLS recover whereas the ones on TLS stay offline (not registering). We see the traffic between the those phones with the Opensips server, but that traffic doesnt look ok (encrypted so hard to tell). In order to get those phones online again, we need to reboot them. Same issue happens when the sip clients move from one internet circuit to the other (from primary connection to secondary). Phones on TLS go offline and do not recover unless rebooted. I know our version is older, but I think the issue is with our setup and not related to a bug on Opensips. I appreciate some direction to resolve this issue. Thanks, Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 12 16:55:20 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 12 Apr 2022 19:55:20 +0300 Subject: [OpenSIPS-Users] avp_db_query retcode value In-Reply-To: References: Message-ID: <0a0a8167-1524-8a67-12bc-f3f48abb9557@opensips.org> Hi Alain, -2 is empty return set -1 generic failure  1 data returned Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/12/22 10:16 AM, Alain Bieuzent wrote: > > Hi All, > > Where can i found the different return value of $retcode after an > avp_db_query. > > I need to distinguish a null result from a failed request > > Regards > > PS : I’m using opensips 3.2 > > > _______________________________________________ > 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 Apr 12 16:58:11 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 12 Apr 2022 19:58:11 +0300 Subject: [OpenSIPS-Users] TLS issue In-Reply-To: References: Message-ID: <1a3444e1-134c-126e-e8e4-3b95ea53390c@opensips.org> Hi MAtt, HAve you tried to see if the TLS devices (1) are able to reconnect to OpenSIPS and (2) send SIP traffic to OpenSIPS after such no-Internet event ? Maybe you can use the siptrace module to try to capture the SIP traffic sent to OpenSIPS via TLS. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/12/22 7:50 PM, L S wrote: > We have been using Opensips 1.11 tls successfully for a while. In our > setup, Opensips is used as a dispatcher to a couple of Asterisk servers. > Some of the sip clients (mostly Cisco SPA phones) communicate with > Opensips server over TLS, some don't. > > Once in a while internet goes down and the sip clients go offline as a > result. When the connection is restored, the phones that are not on > TLS recover whereas the ones on TLS stay offline (not registering). We > see the traffic between the those phones with the Opensips server, but > that traffic doesnt look ok (encrypted so hard to tell). > > In order to get those phones online again, we need to reboot them. > > Same issue happens when the sip clients move from one internet circuit > to the other (from primary connection to secondary). Phones on TLS go > offline and do not recover unless rebooted. > > I know our version is older, but I think the issue is with our setup > and not related to a bug on Opensips. > > I appreciate some direction to resolve this issue. > > Thanks, > Matt > > _______________________________________________ > 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 efes99999 at gmail.com Tue Apr 12 17:21:43 2022 From: efes99999 at gmail.com (L S) Date: Tue, 12 Apr 2022 13:21:43 -0400 Subject: [OpenSIPS-Users] TLS issue In-Reply-To: <1a3444e1-134c-126e-e8e4-3b95ea53390c@opensips.org> References: <1a3444e1-134c-126e-e8e4-3b95ea53390c@opensips.org> Message-ID: Hi Bogdan, There is definitely traffic between the phones and Opensips, but they dont seem to be establishing a working connection - handshake issue maybe? As I mentioned the phones come back up if they are rebooted. Maybe the old connections are somehow messing it up? I will look into the siptrace module. Thanks, Matt On Tue, Apr 12, 2022, 12:58 PM Bogdan-Andrei Iancu wrote: > Hi MAtt, > > HAve you tried to see if the TLS devices (1) are able to reconnect to > OpenSIPS and (2) send SIP traffic to OpenSIPS after such no-Internet event > ? Maybe you can use the siptrace module to try to capture the SIP traffic > sent to OpenSIPS via TLS. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/12/22 7:50 PM, L S wrote: > > We have been using Opensips 1.11 tls successfully for a while. In our > setup, Opensips is used as a dispatcher to a couple of Asterisk servers. > Some of the sip clients (mostly Cisco SPA phones) communicate with > Opensips server over TLS, some don't. > > Once in a while internet goes down and the sip clients go offline as a > result. When the connection is restored, the phones that are not on TLS > recover whereas the ones on TLS stay offline (not registering). We see the > traffic between the those phones with the Opensips server, but that traffic > doesnt look ok (encrypted so hard to tell). > > In order to get those phones online again, we need to reboot them. > > Same issue happens when the sip clients move from one internet circuit to > the other (from primary connection to secondary). Phones on TLS go offline > and do not recover unless rebooted. > > I know our version is older, but I think the issue is with our setup and > not related to a bug on Opensips. > > I appreciate some direction to resolve this issue. > > Thanks, > Matt > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From hunterj91 at hotmail.com Wed Apr 13 15:08:59 2022 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Wed, 13 Apr 2022 15:08:59 +0000 Subject: [OpenSIPS-Users] Clustering Presence opensips 3.2/3.3 in K8s environment Message-ID: Hi All, Has anyone managed to get presence working when using an active/active opensips setup with k8s ? Everything works apart from presence, In particular when a websocket user disconnects due to a client crash. I ideally would want to use event_route[E_CORE_TCP_DISCONNECT] to then grab the disconnect when it comes in via websockets/tcp, however I cant seem to get it to trigger. Could this be due to the underlying hooks OpenSIPS uses to interact with with OS with TCP or something else? As I would use the event route, to then remove the registration from the location table, as otherwise I have duplicate entries in both location and the presentity list. Is this something anyone else has encountered? I have tried using clustering with both presence and pua and have same issues, whereby after an unwanted disconnect subsequent NOTIFY messages contain more than one id per entity; openopen I need to stop this occurring ideally, any help much appreciated. Many thanks Jon Sent from Mail for Windows -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 14 13:30:45 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 14 Apr 2022 16:30:45 +0300 Subject: [OpenSIPS-Users] OpenSIPS Control Panel version 9 just released Message-ID: <666ee55e-17a0-0c61-5a87-653d591d35e4@opensips.org> OpenSIPS Control Panel version 9 is the next major step in the evolution of this faithful and useful companion of OpenSIPS. It brings major additions and enhancements, like full configuration via WEB and DB, per tool settings, a new charting engine - and these are just couple of them. https://blog.opensips.org/2022/04/14/opensips-control-panel-9-stepping-into-a-new-era/ Enjoy it, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ From bullehs at gmail.com Mon Apr 18 06:57:29 2022 From: bullehs at gmail.com (HS) Date: Mon, 18 Apr 2022 11:57:29 +0500 Subject: [OpenSIPS-Users] Billing And Invoicing for Opensips 3.0 Message-ID: Hi all. I am looking to add billing + invoicing to the setup. Came across CGrates for rate cards etc, but is there a billing + invoicing setup that works great with Opensips CDRs? I have looked at astpp so far and searched quite a bit, but nothing much available. Any suggestions? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alberto.rinaudo at gmail.com Mon Apr 18 07:33:45 2022 From: alberto.rinaudo at gmail.com (Alberto) Date: Mon, 18 Apr 2022 08:33:45 +0100 Subject: [OpenSIPS-Users] Billing And Invoicing for Opensips 3.0 In-Reply-To: References: Message-ID: Stay the hell away from astpp, version 4 it's so full of bugs you won't believe it. Maybe cdrtools? On Mon, 18 Apr 2022, 08:00 HS, wrote: > Hi all. > > I am looking to add billing + invoicing to the setup. Came across CGrates > for rate cards etc, but is there a billing + invoicing setup that works > great with Opensips CDRs? > > I have looked at astpp so far and searched quite a bit, but nothing much > available. > > Any suggestions? > > Thanks. > _______________________________________________ > 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 bullehs at gmail.com Mon Apr 18 09:29:38 2022 From: bullehs at gmail.com (HS) Date: Mon, 18 Apr 2022 14:29:38 +0500 Subject: [OpenSIPS-Users] Billing And Invoicing for Opensips 3.0 In-Reply-To: References: Message-ID: Thanks Alberto. Just looked at CDRTool - don't think it has invoicing. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 18 11:00:31 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 18 Apr 2022 14:00:31 +0300 Subject: [OpenSIPS-Users] OpenSIPS Summit 2022 - where to take place? Message-ID: <21cf8abe-4feb-a46b-0bef-cd2e7e77c80c@opensips.org> We are exploring the option of having an *in-person meeting* for this year Summit (of course, with the online support too). And right now we evaluate the possible options as location for the *OpenSIPS Summit, September 2022*. Here is a *quick survey* we run *by 22nd of April* to see what are the community's preferences as locations. So please let us know, so we can make it possible for you all 😎. https://bit.ly/3vrWDqd Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From alberto.rinaudo at gmail.com Mon Apr 18 12:15:54 2022 From: alberto.rinaudo at gmail.com (Alberto) Date: Mon, 18 Apr 2022 13:15:54 +0100 Subject: [OpenSIPS-Users] Billing And Invoicing for Opensips 3.0 In-Reply-To: References: Message-ID: If you need just pdf bills and reports you could use a jasper server. If you need to manage invoices and payments I don't know, but let me know if you find something decent. I landed on astpp a couple of years ago and regret it ever since. On Mon, 18 Apr 2022, 10:32 HS, wrote: > Thanks Alberto. > > Just looked at CDRTool - don't think it has invoicing. > _______________________________________________ > 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 Mon Apr 18 12:58:50 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 18 Apr 2022 15:58:50 +0300 Subject: [OpenSIPS-Users] Clustering Presence opensips 3.2/3.3 in K8s environment In-Reply-To: References: Message-ID: <551985e3-ed35-9589-8624-54c1efe1247e@opensips.org> Hi Jonathan, Maybe the k8s layer (the ingress ??) sticks its tails in there - could you check at opensips level if the TCP conn is still seen as up ? Use the mi list_tcp_conns MI function https://www.opensips.org/Documentation/Interface-CoreMI-3-2#toc4 Best regard, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/13/22 6:08 PM, Jonathan Hunter wrote: > > Hi All, > > Has anyone managed to get presence working when using an active/active > opensips setup with k8s ? > > Everything works apart from presence, In particular when a websocket > user disconnects due to a client crash. > > I ideally would want to use event_route[E_CORE_TCP_DISCONNECT] to then > grab the disconnect when it comes in via websockets/tcp, however I > cant seem to get it to trigger. Could this be due to the underlying > hooks OpenSIPS uses to interact with with OS with TCP or something else? > > As I would use the event route, to then remove the registration from > the location table, as otherwise I have duplicate entries in both > location and the presentity list. > > Is this something anyone else has encountered? > > I have tried using clustering with both presence and pua and have same > issues, whereby after an unwanted disconnect subsequent NOTIFY > messages contain more than one id per entity; > > entity="sip:61067470a372a031a7495a1a at domain"> > > id="0x7ffe75896760">open xmlns="urn:ietf:params:xml:ns:pidf" > id="0x7ffd0716b390">open > > I need to stop this occurring ideally, any help much appreciated. > > Many thanks > > Jon > > Sent from Mail for > Windows > > > _______________________________________________ > 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 couto.a at gmail.com Mon Apr 18 13:58:34 2022 From: couto.a at gmail.com (Antonio Couto) Date: Mon, 18 Apr 2022 10:58:34 -0300 Subject: [OpenSIPS-Users] B2BUA append_hf Message-ID: It is possible to insert header on outgoing INVITE from a B2BUA scenario start by mi command via UDP like this? let obj = { "jsonrpc":"2.0", "id":"7684", "method":"b2b_trigger_scenario", "params":[ "marketing", "operator,sip:9000 at 192.168.0.213:5060", "customer,sip:microsip at 192.168.0.44:5060", ] } I tried to use append_hf but no effect. thanks in advance. Antonio Couto -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 18 14:16:46 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 18 Apr 2022 17:16:46 +0300 Subject: [OpenSIPS-Users] B2BUA append_hf In-Reply-To: References: Message-ID: <687cc7ab-068f-228c-4b67-f1730fee3043@opensips.org> Hi Antonio, not sure where you placed the append_hf(), but you need to place it in the local_route, for original INVITEs. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/18/22 4:58 PM, Antonio Couto wrote: > It is possible to insert header on outgoing INVITE from a B2BUA > scenario start by mi command > via UDP like this? > > letobj =  { > "jsonrpc":"2.0", > "id":"7684", > "method":"b2b_trigger_scenario", > "params":[ > "marketing", > "operator,sip:9000 at 192.168.0.213:5060 > ", > "customer,sip:microsip at 192.168.0.44:5060 > ", > ] >   } > I tried to use append_hf but no effect. > thanks in advance. > Antonio Couto > > > _______________________________________________ > 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 couto.a at gmail.com Mon Apr 18 14:35:57 2022 From: couto.a at gmail.com (Antonio Couto) Date: Mon, 18 Apr 2022 11:35:57 -0300 Subject: [OpenSIPS-Users] B2BUA append_hf In-Reply-To: <687cc7ab-068f-228c-4b67-f1730fee3043@opensips.org> References: <687cc7ab-068f-228c-4b67-f1730fee3043@opensips.org> Message-ID: Thanks for quick response. I put on route[b2b_request] and route[b2b_reply] routes, but I think this model I am trying to use will not work like you said. i think I will need to send a complete INVITE from another application to start b2b from a INVITE message, so I can preserve this original header on b2b logic module. thanks a lot. On Mon, Apr 18, 2022 at 11:16 AM Bogdan-Andrei Iancu wrote: > Hi Antonio, > > not sure where you placed the append_hf(), but you need to place it in the > local_route, for original INVITEs. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/18/22 4:58 PM, Antonio Couto wrote: > > It is possible to insert header on outgoing INVITE from a B2BUA scenario > start by mi command > via UDP like this? > > let obj = { > "jsonrpc":"2.0", > "id":"7684", > "method":"b2b_trigger_scenario", > "params":[ > "marketing", > "operator,sip:9000 at 192.168.0.213:5060", > "customer,sip:microsip at 192.168.0.44:5060", > ] > } > I tried to use append_hf but no effect. > thanks in advance. > Antonio Couto > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 18 15:03:21 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 18 Apr 2022 18:03:21 +0300 Subject: [OpenSIPS-Users] B2BUA append_hf In-Reply-To: References: <687cc7ab-068f-228c-4b67-f1730fee3043@opensips.org> Message-ID: As per my suggestion, try the local_route Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/18/22 5:35 PM, Antonio Couto wrote: > Thanks for quick response. > > I put on route[b2b_request] and route[b2b_reply] routes, but I think > this model I am trying to use will not work > like you said. > > i think I will need to send a complete INVITE from another application > to start b2b from a INVITE message, > so I can preserve this original header on b2b logic module. > > thanks a lot. > > > On Mon, Apr 18, 2022 at 11:16 AM Bogdan-Andrei Iancu > > wrote: > > Hi Antonio, > > not sure where you placed the append_hf(), but you need to place > it in the local_route, for original INVITEs. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/18/22 4:58 PM, Antonio Couto wrote: >> It is possible to insert header on outgoing INVITE from a B2BUA >> scenario start by mi command >> via UDP like this? >> >> letobj =  { >> "jsonrpc":"2.0", >> "id":"7684", >> "method":"b2b_trigger_scenario", >> "params":[ >> "marketing", >> "operator,sip:9000 at 192.168.0.213:5060 >> ", >> "customer,sip:microsip at 192.168.0.44:5060 >> ", >> ] >>   } >> I tried to use append_hf but no effect. >> thanks in advance. >> Antonio Couto >> >> >> _______________________________________________ >> 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 couto.a at gmail.com Mon Apr 18 15:20:58 2022 From: couto.a at gmail.com (Antonio Couto) Date: Mon, 18 Apr 2022 12:20:58 -0300 Subject: [OpenSIPS-Users] B2BUA append_hf In-Reply-To: References: <687cc7ab-068f-228c-4b67-f1730fee3043@opensips.org> Message-ID: OK, but using the request on json that I inform on may email send ny mi command, all messages are handle on these routes only: route[b2b_request] route[b2b_reply] there no messages captured on main *route {* thanks On Mon, Apr 18, 2022 at 12:03 PM Bogdan-Andrei Iancu wrote: > As per my suggestion, try the local_route > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/18/22 5:35 PM, Antonio Couto wrote: > > Thanks for quick response. > > I put on route[b2b_request] and route[b2b_reply] routes, but I think this > model I am trying to use will not work > like you said. > > i think I will need to send a complete INVITE from another application to > start b2b from a INVITE message, > so I can preserve this original header on b2b logic module. > > thanks a lot. > > > On Mon, Apr 18, 2022 at 11:16 AM Bogdan-Andrei Iancu > wrote: > >> Hi Antonio, >> >> not sure where you placed the append_hf(), but you need to place it in >> the local_route, for original INVITEs. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 4/18/22 4:58 PM, Antonio Couto wrote: >> >> It is possible to insert header on outgoing INVITE from a B2BUA scenario >> start by mi command >> via UDP like this? >> >> let obj = { >> "jsonrpc":"2.0", >> "id":"7684", >> "method":"b2b_trigger_scenario", >> "params":[ >> "marketing", >> "operator,sip:9000 at 192.168.0.213:5060", >> "customer,sip:microsip at 192.168.0.44:5060", >> ] >> } >> I tried to use append_hf but no effect. >> thanks in advance. >> Antonio Couto >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 18 15:49:53 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 18 Apr 2022 18:49:53 +0300 Subject: [OpenSIPS-Users] B2BUA append_hf In-Reply-To: References: <687cc7ab-068f-228c-4b67-f1730fee3043@opensips.org> Message-ID: https://www.opensips.org/Documentation/Script-Routes-3-2#local_route Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/18/22 6:20 PM, Antonio Couto wrote: > OK, but using the request on json that I inform on may email send ny > mi command, > all messages are handle on these routes only: > > route[b2b_request] > route[b2b_reply] > > there no messages captured on main *route {* > > thanks > > > > > On Mon, Apr 18, 2022 at 12:03 PM Bogdan-Andrei Iancu > > wrote: > > As per my suggestion, try the local_route > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/18/22 5:35 PM, Antonio Couto wrote: >> Thanks for quick response. >> >> I put on route[b2b_request] and route[b2b_reply] routes, but I >> think this model I am trying to use will not work >> like you said. >> >> i think I will need to send a complete INVITE from another >> application to start b2b from a INVITE message, >> so I can preserve this original header on b2b logic module. >> >> thanks a lot. >> >> >> On Mon, Apr 18, 2022 at 11:16 AM Bogdan-Andrei Iancu >> > wrote: >> >> Hi Antonio, >> >> not sure where you placed the append_hf(), but you need to >> place it in the local_route, for original INVITEs. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 4/18/22 4:58 PM, Antonio Couto wrote: >>> It is possible to insert header on outgoing INVITE from a >>> B2BUA scenario start by mi command >>> via UDP like this? >>> >>> letobj =  { >>> "jsonrpc":"2.0", >>> "id":"7684", >>> "method":"b2b_trigger_scenario", >>> "params":[ >>> "marketing", >>> "operator,sip:9000 at 192.168.0.213:5060 >>> ", >>> "customer,sip:microsip at 192.168.0.44:5060 >>> ", >>> ] >>>   } >>> I tried to use append_hf but no effect. >>> thanks in advance. >>> Antonio Couto >>> >>> >>> _______________________________________________ >>> 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 couto.a at gmail.com Mon Apr 18 15:58:18 2022 From: couto.a at gmail.com (Antonio Couto) Date: Mon, 18 Apr 2022 12:58:18 -0300 Subject: [OpenSIPS-Users] B2BUA append_hf In-Reply-To: References: <687cc7ab-068f-228c-4b67-f1730fee3043@opensips.org> Message-ID: Sorry, I am new on Opensips, I did understand when you said "local_route", now its working. Thank very so much for your time. best regards. 2022/04/18 12:56:29.681936 192.168.0.215:5070 -> 192.168.0.213:5060 INVITE sip:9000 at 192.168.0.213:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.215:5070;branch=z9hG4bKd8e9.12db2f97.0 To: sip:9000 at 192.168.0.213:5060 From: ;tag=c6cf02786cfbb0099acf67a3ce978fec CSeq: 2 INVITE Call-ID: B2B.332.605219.1650297388 Max-Forwards: 70 Content-Length: 0 User-Agent: OpenSIPS (3.2.5 (x86_64/linux)) Contact: *X-TESTE: hasta la vista baby* On Mon, Apr 18, 2022 at 12:50 PM Bogdan-Andrei Iancu wrote: > https://www.opensips.org/Documentation/Script-Routes-3-2#local_route > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/18/22 6:20 PM, Antonio Couto wrote: > > OK, but using the request on json that I inform on may email send ny mi > command, > all messages are handle on these routes only: > > route[b2b_request] > route[b2b_reply] > > there no messages captured on main *route {* > > thanks > > > > > On Mon, Apr 18, 2022 at 12:03 PM Bogdan-Andrei Iancu > wrote: > >> As per my suggestion, try the local_route >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 4/18/22 5:35 PM, Antonio Couto wrote: >> >> Thanks for quick response. >> >> I put on route[b2b_request] and route[b2b_reply] routes, but I think this >> model I am trying to use will not work >> like you said. >> >> i think I will need to send a complete INVITE from another application to >> start b2b from a INVITE message, >> so I can preserve this original header on b2b logic module. >> >> thanks a lot. >> >> >> On Mon, Apr 18, 2022 at 11:16 AM Bogdan-Andrei Iancu >> wrote: >> >>> Hi Antonio, >>> >>> not sure where you placed the append_hf(), but you need to place it in >>> the local_route, for original INVITEs. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>> >>> On 4/18/22 4:58 PM, Antonio Couto wrote: >>> >>> It is possible to insert header on outgoing INVITE from a B2BUA scenario >>> start by mi command >>> via UDP like this? >>> >>> let obj = { >>> "jsonrpc":"2.0", >>> "id":"7684", >>> "method":"b2b_trigger_scenario", >>> "params":[ >>> "marketing", >>> "operator,sip:9000 at 192.168.0.213:5060", >>> "customer,sip:microsip at 192.168.0.44:5060", >>> ] >>> } >>> I tried to use append_hf but no effect. >>> thanks in advance. >>> Antonio Couto >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 18 16:05:06 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 18 Apr 2022 19:05:06 +0300 Subject: [OpenSIPS-Users] B2BUA append_hf In-Reply-To: References: <687cc7ab-068f-228c-4b67-f1730fee3043@opensips.org> Message-ID: <466b9888-3a7f-50b3-53fb-ae78c720dfe1@opensips.org> Glad it worked :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/18/22 6:58 PM, Antonio Couto wrote: > Sorry, I am new on Opensips, I did understand when you said > "local_route", now its working. > Thank very so much for your time. > best regards. > > 2022/04/18 12:56:29.681936 192.168.0.215:5070 > -> 192.168.0.213:5060 > > INVITE sip:9000 at 192.168.0.213:5060 > SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.215:5070;branch=z9hG4bKd8e9.12db2f97.0 > To: sip:9000 at 192.168.0.213:5060 > From: >;tag=c6cf02786cfbb0099acf67a3ce978fec > CSeq: 2 INVITE > Call-ID: B2B.332.605219.1650297388 > Max-Forwards: 70 > Content-Length: 0 > User-Agent: OpenSIPS (3.2.5 (x86_64/linux)) > Contact: > > *X-TESTE: hasta la vista baby* > > > > > On Mon, Apr 18, 2022 at 12:50 PM Bogdan-Andrei Iancu > > wrote: > > https://www.opensips.org/Documentation/Script-Routes-3-2#local_route > > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/18/22 6:20 PM, Antonio Couto wrote: >> OK, but using the request on json that I inform on may email send >> ny mi command, >> all messages are handle on these routes only: >> >> route[b2b_request] >> route[b2b_reply] >> >> there no messages captured on main *route {* >> >> thanks >> >> >> >> >> On Mon, Apr 18, 2022 at 12:03 PM Bogdan-Andrei Iancu >> > wrote: >> >> As per my suggestion, try the local_route >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 4/18/22 5:35 PM, Antonio Couto wrote: >>> Thanks for quick response. >>> >>> I put on route[b2b_request] and route[b2b_reply] routes, but >>> I think this model I am trying to use will not work >>> like you said. >>> >>> i think I will need to send a complete INVITE from another >>> application to start b2b from a INVITE message, >>> so I can preserve this original header on b2b logic module. >>> >>> thanks a lot. >>> >>> >>> On Mon, Apr 18, 2022 at 11:16 AM Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi Antonio, >>> >>> not sure where you placed the append_hf(), but you need >>> to place it in the local_route, for original INVITEs. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>> >>> On 4/18/22 4:58 PM, Antonio Couto wrote: >>>> It is possible to insert header on outgoing INVITE from >>>> a B2BUA scenario start by mi command >>>> via UDP like this? >>>> >>>> letobj =  { >>>> "jsonrpc":"2.0", >>>> "id":"7684", >>>> "method":"b2b_trigger_scenario", >>>> "params":[ >>>> "marketing", >>>> "operator,sip:9000 at 192.168.0.213:5060 >>>> ", >>>> "customer,sip:microsip at 192.168.0.44:5060 >>>> ", >>>> ] >>>>   } >>>> I tried to use append_hf but no effect. >>>> thanks in advance. >>>> Antonio Couto >>>> >>>> >>>> _______________________________________________ >>>> 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 Wed Apr 20 12:13:18 2022 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 20 Apr 2022 15:13:18 +0300 Subject: [OpenSIPS-Users] [Blog] Fine-Grained TCP Configuration Using the "tcp_mgm" Module Message-ID: Hi all, OpenSIPS 3.3 just got better with the addition of the long-awaited "tcp_mgm" module!  Together with some TCP processing throughput enhancements, they are all discussed in the following blog post: https://blog.opensips.org/2022/04/20/fine-grained-tcp-configuration-using-the-tcp_mgm-module/ Happy Testing, -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS eBootcamp May 23 - June 3 | www.opensips.org/training From liviu at opensips.org Wed Apr 20 16:47:37 2022 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 20 Apr 2022 19:47:37 +0300 Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 3.1.9 and 3.2.6 minor releases In-Reply-To: <14e04ac8-e54e-2fe1-22c6-8854e74ad712@opensips.org> References: <14e04ac8-e54e-2fe1-22c6-8854e74ad712@opensips.org> Message-ID: On 11.04.2022 15:40, Liviu Chircu wrote: > Hi, everyone! > > The 3.1.9 and 3.2.6 OpenSIPS minor versions are scheduled for release > on Wed, April 20th -- roughly a week and a half from now. > > In preparation for the release, we have already imposed a freeze on > any significant fixes (as volume) on these two stable branches, in > order to ensure a safe window for testing in the days ahead. > > Best regards, > Hello, As scheduled, the OpenSIPS 3.1.9 and 3.2.6 minor releases are already out there for the taking, packing important fixes done in the last two months. Detailed changelogs are available on the website^[1][2] . [1]: https://opensips.org/pub/opensips/3.1.9/ChangeLog [2]: https://opensips.org/pub/opensips/3.2.6/ChangeLog Happy Hacking! -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com OpenSIPS eBootcamp May 23 - June 3 |www.opensips.org/training -------------- next part -------------- An HTML attachment was scrubbed... URL: From vinayak.makwana at ecosmob.com Fri Apr 22 08:15:16 2022 From: vinayak.makwana at ecosmob.com (Vinayak Makwana) Date: Fri, 22 Apr 2022 13:45:16 +0530 Subject: [OpenSIPS-Users] Dispatcher issue in Active -> Probing State. Message-ID: Hi all, I having issue with dispatcher module in that when i execute "opensipsctl dispatcher dump" command at that time i am getting like this "URI:: sip:1.2.3.4:5060 state=Active first_hit_counter=0" but the trunk doesn't respond to the OPTIONS packet and it needs to change state from Active to probing as per below configuration but it doesn't work. Why does it not get into probing state after ds_ping_interval timeout and ds_probing_threshhold reached? I am using OpenSIPS Version : 2.4.11 Here's my configuration for dispatcher module loadmodule "dispatcher.so" modparam("dispatcher","db_url","URL") modparam("dispatcher", "ds_ping_from", "sip:opensips at pinger.com") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_sock", "udp:6.7.8.9:5060") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "persistent_state", 1) Please Provide any suggestion regarding this Regards Vinayak Makwana -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexanderhenryperkins at gmail.com Sat Apr 23 17:41:56 2022 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Sat, 23 Apr 2022 13:41:56 -0400 Subject: [OpenSIPS-Users] Media IP of Caller Message-ID: Hi All. I have an interesting question - how can I get the media IP of the caller? Not the signaling IP, but the media IP. Is there a variable for that? I've tried this: $var(aline) = $(rb{sdp.line,c,1}), but it seems to be bringing on the caller's private IP address. Any help is appreciated. Thank you, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Sat Apr 23 18:14:05 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Sat, 23 Apr 2022 21:14:05 +0300 Subject: [OpenSIPS-Users] OpenSIPS Summit 2022 - where to take place? In-Reply-To: <21cf8abe-4feb-a46b-0bef-cd2e7e77c80c@opensips.org> References: <21cf8abe-4feb-a46b-0bef-cd2e7e77c80c@opensips.org> Message-ID: The community has spoken, the Summit will take place in Athens, Greece Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/18/22 2:00 PM, Bogdan-Andrei Iancu wrote: > > We are exploring the option of having an *in-person meeting* for this > year Summit (of course, with the online support too). > And right now we evaluate the possible options as location for the > *OpenSIPS Summit, September 2022*. > > Here is a *quick survey* we run *by 22nd of April* to see what are the > community's preferences as locations. So please let us know, so we can > make it possible for you all 😎. > > https://bit.ly/3vrWDqd > > Best regards, > -- > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > _______________________________________________ > 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: dihblflpalpmgjda.png Type: image/png Size: 6754 bytes Desc: not available URL: From dovid at telecurve.com Sun Apr 24 06:45:16 2022 From: dovid at telecurve.com (Dovid Bender) Date: Sun, 24 Apr 2022 02:45:16 -0400 Subject: [OpenSIPS-Users] OpenSIPS Summit 2022 - where to take place? In-Reply-To: References: <21cf8abe-4feb-a46b-0bef-cd2e7e77c80c@opensips.org> Message-ID: Bogdan, Is there any idea on dates? On Sat, Apr 23, 2022 at 2:16 PM Bogdan-Andrei Iancu wrote: > > The community has spoken, the Summit will take place in Athens, Greece > > > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/18/22 2:00 PM, Bogdan-Andrei Iancu wrote: > > > We are exploring the option of having an *in-person meeting* for this > year Summit (of course, with the online support too). > And right now we evaluate the possible options as location for the *OpenSIPS > Summit, September 2022*. > > Here is a *quick survey* we run *by 22nd of April* to see what are the > community's preferences as locations. So please let us know, so we can make > it possible for you all 😎. > > https://bit.ly/3vrWDqd > > Best regards, > > -- > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > > _______________________________________________ > 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: -------------- next part -------------- A non-text attachment was scrubbed... Name: dihblflpalpmgjda.png Type: image/png Size: 6754 bytes Desc: not available URL: From devang.dhandhalya at ecosmob.com Mon Apr 25 12:47:47 2022 From: devang.dhandhalya at ecosmob.com (Devang Dhandhalya) Date: Mon, 25 Apr 2022 18:17:47 +0530 Subject: [OpenSIPS-Users] Stir Shaken Verification issue Message-ID: Hello All I am testing STIR/SHAKEN calls using two servers. calls originating to the first server adding identity header and when sending calls to the second server for verification service at the time of verification service i am getting below error . error :437 , Unsupported Credential , Verification Fails with Return code :-8 INFO:stir_shaken:verify_callback: certificate validation failed: self signed certificate INFO:stir_shaken:w_stir_verify: Invalid certificate OpenSIPS Version : 3.2.2 I generate certificate using domain which mapped with those 2 server : https://github.com/OpenSIPIt/OpenSIPIt_00/blob/master/STIR_SHAKEN/Certgen/gencert.sh When the same server generates an identity header and verifies it at that time not getting an issue call is working fine but when the identity header generated by server 1 and going to verify it by server 2 we get this above error. Is it related to the URL which is in the info param ? When I open that URL in the browser I am able to see the certificate. Please suggest a solution for this issue. Regards Devang Dhandhalya -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From hunterj91 at hotmail.com Mon Apr 25 14:12:55 2022 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Mon, 25 Apr 2022 14:12:55 +0000 Subject: [OpenSIPS-Users] Clustering Presence opensips 3.2/3.3 in K8s environment In-Reply-To: <551985e3-ed35-9589-8624-54c1efe1247e@opensips.org> References: <551985e3-ed35-9589-8624-54c1efe1247e@opensips.org> Message-ID: Hi Bogdan, Thank you for the reply I can see there are tcp connections but I don’t seem to get anything. I assume it may well be k8s related?. I also cant seem to get NAT ping working, I assume this should work out to websocket connections as long as flags are set? The outputs are; "ID": 1317975555, "Type": "ws", "State": 0, "Remote": "10.10.51.228:35462", "Local": "10.10.2.91:8081", "Lifetime": "2022-04-25 14:11:31", "Alias port": 35462 } ] And location shows; "AORs": [ { "AOR": "61067470a372a031a7495a1a@", "Contacts": [ { "Contact": "sip:c0r0d0i7 at b0ek39eabrvf.invalid;transport=wss", "ContactID": "4544061655272656153", "Expires": 519, "Q": "", "Callid": "jbbvtjp2l7bujcom73s3", "Cseq": 2, "User-agent": "SIP.js/0.20.0", "Received": "sip:10.10.51.228:35462;transport=ws", "State": "CS_NEW", "Flags": 0, "Cflags": "WS_DEVICE SIPPING_RTO SIPPING_ENABLE", "Socket": "ws::8081", "Methods": 5439 } ] } ] However I cant seem to also get opensips to send SIP keepalive with these settings; #### NAThelper module loadmodule "nathelper.so" modparam("nathelper", "received_avp", "$avp(rcv)") modparam("nathelper", "natping_tcp",1) modparam("nathelper", "natping_interval", 5) modparam("nathelper", "sipping_bflag", "SIPPING_ENABLE") modparam("nathelper", "remove_on_timeout_bflag", "SIPPING_RTO") modparam("nathelper", "sipping_from", "sip:pinger at DOMAIN") modparam("nathelper", "max_pings_lost", 2) modparam("nathelper", "cluster_id", 9) modparam("nathelper", "cluster_sharing_tag", "node/2=active") Am I missing something here? Many thanks Jon Sent from Mail for Windows From: Bogdan-Andrei Iancu Sent: 18 April 2022 13:59 To: OpenSIPS users mailling list; Jonathan Hunter Subject: Re: [OpenSIPS-Users] Clustering Presence opensips 3.2/3.3 in K8s environment Hi Jonathan, Maybe the k8s layer (the ingress ??) sticks its tails in there - could you check at opensips level if the TCP conn is still seen as up ? Use the mi list_tcp_conns MI function https://www.opensips.org/Documentation/Interface-CoreMI-3-2#toc4 Best regard, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/13/22 6:08 PM, Jonathan Hunter wrote: Hi All, Has anyone managed to get presence working when using an active/active opensips setup with k8s ? Everything works apart from presence, In particular when a websocket user disconnects due to a client crash. I ideally would want to use event_route[E_CORE_TCP_DISCONNECT] to then grab the disconnect when it comes in via websockets/tcp, however I cant seem to get it to trigger. Could this be due to the underlying hooks OpenSIPS uses to interact with with OS with TCP or something else? As I would use the event route, to then remove the registration from the location table, as otherwise I have duplicate entries in both location and the presentity list. Is this something anyone else has encountered? I have tried using clustering with both presence and pua and have same issues, whereby after an unwanted disconnect subsequent NOTIFY messages contain more than one id per entity; > openopen I need to stop this occurring ideally, any help much appreciated. Many thanks Jon Sent from Mail for Windows _______________________________________________ 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 Apr 26 08:23:49 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 26 Apr 2022 11:23:49 +0300 Subject: [OpenSIPS-Users] OpenSIPS Summit 2022 - where to take place? In-Reply-To: References: <21cf8abe-4feb-a46b-0bef-cd2e7e77c80c@opensips.org> Message-ID: <69be63fc-8e41-0299-9dbc-3d853f64fcc8@opensips.org> Hi Dovid, We are targeting mid / end of September, depends on the hotel arrangements (work in progress) - as soon as we know, we will announce it. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/24/22 9:45 AM, Dovid Bender wrote: > Bogdan, > > Is there any idea on dates? > > > On Sat, Apr 23, 2022 at 2:16 PM Bogdan-Andrei Iancu > > wrote: > > > The community has spoken, the Summit will take place in Athens, Greece > > > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/18/22 2:00 PM, Bogdan-Andrei Iancu wrote: >> >> We are exploring the option of having an *in-person meeting* for >> this year Summit (of course, with the online support too). >> And right now we evaluate the possible options as location for >> the *OpenSIPS Summit, September 2022*. >> >> Here is a *quick survey* we run *by 22nd of April* to see what >> are the community's preferences as locations. So please let us >> know, so we can make it possible for you all 😎. >> >> https://bit.ly/3vrWDqd >> >> Best regards, >> -- >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> _______________________________________________ >> 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: -------------- next part -------------- A non-text attachment was scrubbed... Name: dihblflpalpmgjda.png Type: image/png Size: 6754 bytes Desc: not available URL: From spanda at 3clogic.com Tue Apr 26 13:55:20 2022 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 26 Apr 2022 19:25:20 +0530 Subject: [OpenSIPS-Users] Need some suggestion in full-sharing-cachedb-cluster . Message-ID: Hi All , I am using opensips 3.2 . I Want to build a cluster with redis as cachedb . I am using full-sharing-cachedb-cluster mode . I have tested with MongoDB . It's working perfectly fine . I am curious to know if this works with Redis or not . In the document it mentions about *Cassandra* . But I am not getting how to configure it with Redis . loadmodule "db_cachedb.so" loadmodule "cachedb_redis.so" modparam("cachedb_redis", "cachedb_url","redis:group1://10.4.128.144:6379/") modparam("db_cachedb","cachedb_url", "redis:group1://10.4.128.144:6379/") modparam("usrloc", "cluster_mode", "full-sharing-cachedb") modparam("usrloc", "cachedb_url", "redis:group1:// 10.4.128.144:6379/opensipsDB.userlocation") opensips gives some error while starting . ERROR:usrloc:ul_check_db: not enough capabilities for cachedb_url redis:group1://10.4.128.144:6379/opensipsDB.userlocation ERROR:usrloc:mod_init: DB support check failed ERROR:core:init_mod: failed to initialize module usrloc ERROR:core:main: error while initializing modules Can anyone please help ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From vasilios.tzanoudakis at voiceland.gr Tue Apr 26 14:47:38 2022 From: vasilios.tzanoudakis at voiceland.gr (Vasilios Tzanoudakis) Date: Tue, 26 Apr 2022 17:47:38 +0300 Subject: [OpenSIPS-Users] Need some suggestion in full-sharing-cachedb-cluster . In-Reply-To: References: Message-ID: Dear Sasmita, Cachedb supports only mongodb or cassandra backends for clustering. Vasilios Tzanoudakis Στις Τρί, 26 Απρ 2022, 16:57 ο χρήστης Sasmita Panda έγραψε: > Hi All , > > I am using opensips 3.2 . > I Want to build a cluster with redis as cachedb . I am > using full-sharing-cachedb-cluster mode . I have tested with MongoDB . > It's working perfectly fine . > > I am curious to know if this works with Redis or not . In the document it > mentions about *Cassandra* . But I am not getting how to configure it > with Redis . > > loadmodule "db_cachedb.so" > loadmodule "cachedb_redis.so" > modparam("cachedb_redis", "cachedb_url","redis:group1://10.4.128.144:6379/ > ") > modparam("db_cachedb","cachedb_url", "redis:group1://10.4.128.144:6379/") > > modparam("usrloc", "cluster_mode", "full-sharing-cachedb") > modparam("usrloc", "cachedb_url", "redis:group1:// > 10.4.128.144:6379/opensipsDB.userlocation") > > > opensips gives some error while starting . > > ERROR:usrloc:ul_check_db: not enough capabilities for cachedb_url > redis:group1://10.4.128.144:6379/opensipsDB.userlocation > ERROR:usrloc:mod_init: DB support check failed > ERROR:core:init_mod: failed to initialize module usrloc > ERROR:core:main: error while initializing modules > > > Can anyone please help ? > > *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 hunterj91 at hotmail.com Tue Apr 26 21:30:14 2022 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Tue, 26 Apr 2022 21:30:14 +0000 Subject: [OpenSIPS-Users] Clustering Presence opensips 3.2/3.3 in K8s environment In-Reply-To: References: <551985e3-ed35-9589-8624-54c1efe1247e@opensips.org> Message-ID: Hi Bogdan. Just to follow up I have been trying to make openSIPS send SIP options in the same environment to websocket registered users, and I just cant get it to send even though the branch flags are set. I now have these settings defined for NAThelper; loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", "SIPPING_ENABLE") modparam("nathelper", "sipping_from", "sip:pinger@") modparam("nathelper", "received_avp", "$avp(rcv)") modparam("nathelper", "ping_threshold", 5) modparam("nathelper", "max_pings_lost", 3) modparam("nathelper", "natping_partitions", 4) modparam("nathelper", "remove_on_timeout_bflag", "SIPPING_RTO") modparam("nathelper", "natping_tcp", 1) modparam("nathelper", "cluster_id", 1) modparam("nathelper", "cluster_sharing_tag", "node/2=active") Also have pinging_mode set for usrloc, and this is in a federation-cachedb-cluster with 2 opensips containers running active/active. If I use t_new_request I can send a SIP OPTIONS message out, so I assume I am missing a parameter or its mis configuration? Or could it be environment, this is a k8s setup. Any advice/tips would be great. Thank you . Jon Sent from Mail for Windows From: Jonathan Hunter Sent: 25 April 2022 15:28 To: Bogdan-Andrei Iancu; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Clustering Presence opensips 3.2/3.3 in K8s environment Hi Bogdan, Thank you for the reply I can see there are tcp connections but I don’t seem to get anything. I assume it may well be k8s related?. I also cant seem to get NAT ping working, I assume this should work out to websocket connections as long as flags are set? The outputs are; "ID": 1317975555, "Type": "ws", "State": 0, "Remote": "10.10.51.228:35462", "Local": "10.10.2.91:8081", "Lifetime": "2022-04-25 14:11:31", "Alias port": 35462 } ] And location shows; "AORs": [ { "AOR": "61067470a372a031a7495a1a@", "Contacts": [ { "Contact": "sip:c0r0d0i7 at b0ek39eabrvf.invalid;transport=wss", "ContactID": "4544061655272656153", "Expires": 519, "Q": "", "Callid": "jbbvtjp2l7bujcom73s3", "Cseq": 2, "User-agent": "SIP.js/0.20.0", "Received": "sip:10.10.51.228:35462;transport=ws", "State": "CS_NEW", "Flags": 0, "Cflags": "WS_DEVICE SIPPING_RTO SIPPING_ENABLE", "Socket": "ws::8081", "Methods": 5439 } ] } ] However I cant seem to also get opensips to send SIP keepalive with these settings; #### NAThelper module loadmodule "nathelper.so" modparam("nathelper", "received_avp", "$avp(rcv)") modparam("nathelper", "natping_tcp",1) modparam("nathelper", "natping_interval", 5) modparam("nathelper", "sipping_bflag", "SIPPING_ENABLE") modparam("nathelper", "remove_on_timeout_bflag", "SIPPING_RTO") modparam("nathelper", "sipping_from", "sip:pinger at DOMAIN") modparam("nathelper", "max_pings_lost", 2) modparam("nathelper", "cluster_id", 9) modparam("nathelper", "cluster_sharing_tag", "node/2=active") Am I missing something here? Many thanks Jon Sent from Mail for Windows From: Bogdan-Andrei Iancu Sent: 18 April 2022 13:59 To: OpenSIPS users mailling list; Jonathan Hunter Subject: Re: [OpenSIPS-Users] Clustering Presence opensips 3.2/3.3 in K8s environment Hi Jonathan, Maybe the k8s layer (the ingress ??) sticks its tails in there - could you check at opensips level if the TCP conn is still seen as up ? Use the mi list_tcp_conns MI function https://www.opensips.org/Documentation/Interface-CoreMI-3-2#toc4 Best regard, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/13/22 6:08 PM, Jonathan Hunter wrote: Hi All, Has anyone managed to get presence working when using an active/active opensips setup with k8s ? Everything works apart from presence, In particular when a websocket user disconnects due to a client crash. I ideally would want to use event_route[E_CORE_TCP_DISCONNECT] to then grab the disconnect when it comes in via websockets/tcp, however I cant seem to get it to trigger. Could this be due to the underlying hooks OpenSIPS uses to interact with with OS with TCP or something else? As I would use the event route, to then remove the registration from the location table, as otherwise I have duplicate entries in both location and the presentity list. Is this something anyone else has encountered? I have tried using clustering with both presence and pua and have same issues, whereby after an unwanted disconnect subsequent NOTIFY messages contain more than one id per entity; > openopen I need to stop this occurring ideally, any help much appreciated. Many thanks Jon Sent from Mail for Windows _______________________________________________ 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 Apr 27 05:09:39 2022 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 27 Apr 2022 10:39:39 +0530 Subject: [OpenSIPS-Users] Need some suggestion in full-sharing-cachedb-cluster . In-Reply-To: References: Message-ID: Hi , Is Document DB of AWS supported ? Document DB is mongodb compatible . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, Apr 26, 2022 at 8:28 PM Vasilios Tzanoudakis < vasilios.tzanoudakis at voiceland.gr> wrote: > Dear Sasmita, > > Cachedb supports only mongodb or cassandra backends for clustering. > > Vasilios Tzanoudakis > > Στις Τρί, 26 Απρ 2022, 16:57 ο χρήστης Sasmita Panda > έγραψε: > >> Hi All , >> >> I am using opensips 3.2 . >> I Want to build a cluster with redis as cachedb . I am >> using full-sharing-cachedb-cluster mode . I have tested with MongoDB . >> It's working perfectly fine . >> >> I am curious to know if this works with Redis or not . In the document it >> mentions about *Cassandra* . But I am not getting how to configure it >> with Redis . >> >> loadmodule "db_cachedb.so" >> loadmodule "cachedb_redis.so" >> modparam("cachedb_redis", "cachedb_url","redis:group1:// >> 10.4.128.144:6379/") >> modparam("db_cachedb","cachedb_url", "redis:group1://10.4.128.144:6379/") >> >> modparam("usrloc", "cluster_mode", "full-sharing-cachedb") >> modparam("usrloc", "cachedb_url", "redis:group1:// >> 10.4.128.144:6379/opensipsDB.userlocation") >> >> >> opensips gives some error while starting . >> >> ERROR:usrloc:ul_check_db: not enough capabilities for cachedb_url >> redis:group1://10.4.128.144:6379/opensipsDB.userlocation >> ERROR:usrloc:mod_init: DB support check failed >> ERROR:core:init_mod: failed to initialize module usrloc >> ERROR:core:main: error while initializing modules >> >> >> Can anyone please help ? >> >> *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 Apr 27 06:51:30 2022 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 27 Apr 2022 12:21:30 +0530 Subject: [OpenSIPS-Users] Some query regarding Opensips cluster . Message-ID: Hi All , As mention on all documents , opensips says the clustering support is added for cachedb . It means the data in the cacheDb is shared in the cluster . So there is 2 types of full sharing cluster , 1 . full sharing when all data is stored in opensips memory 2. Full sharing when all data is stored in NoSQL cachedb . Why is cluster support not in SQL? I mean if all nodes will keep the data in a SQL database (like db_mode 3 ) and that data will be shared within nodes then what will be the problem ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 28 14:42:53 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 28 Apr 2022 17:42:53 +0300 Subject: [OpenSIPS-Users] OpenSIPS Security Audit - mission accomplished Message-ID: <777b90db-bf92-b32f-388a-43aed7088743@opensips.org> OpenSIPS 3.2 Security Audit is now completed, with facts and results. A lot of fuzzing, testing and of course fixing. Everything just to make OpenSIPS more secure. Thank you @enablesecurity ! https://blog.opensips.org/2022/04/28/opensips-security-audit-facts-and-results/ Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Fri Apr 29 10:21:00 2022 From: johan at democon.be (johan) Date: Fri, 29 Apr 2022 12:21:00 +0200 Subject: [OpenSIPS-Users] mongo for dialog replication. Message-ID: <2f941803-a124-a351-7c06-93a1e41e7da1@democon.be> Hi, is there any datafill to do on mongodb when you use mongodb in usrloc modparam("usrloc", "cachedb_url", "mongodb://10.0.0.4:27017/opensipsDB.userlocation") ? How do I create opensipsDB.userlocation ? I know :-) I don't have used mongodb ever before. wkr, From liviu at opensips.org Fri Apr 29 10:26:29 2022 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 29 Apr 2022 13:26:29 +0300 Subject: [OpenSIPS-Users] mongo for dialog replication. In-Reply-To: <2f941803-a124-a351-7c06-93a1e41e7da1@democon.be> References: <2f941803-a124-a351-7c06-93a1e41e7da1@democon.be> Message-ID: On 29.04.2022 13:21, johan wrote: > How do I create opensipsDB.userlocation ? > > I know:-) I don't have used mongodb ever before. Hi johan, That's the beautiful thing about the JavaScript-like MongoDB: *it **always**works*. Even if the collection doesn't exist.  Even if the DB doesn't exist... Now, "how good it works" is a topic best left for another day... Best Regards, -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com OpenSIPS eBootcamp May 23 - June 3 |www.opensips.org/training -------------- next part -------------- An HTML attachment was scrubbed... URL: From y.kirsanov at gmail.com Fri Apr 29 10:43:57 2022 From: y.kirsanov at gmail.com (Yury Kirsanov) Date: Fri, 29 Apr 2022 20:43:57 +1000 Subject: [OpenSIPS-Users] Problem with fix_nated_contact Message-ID: Hi, I'm using OpenSIPS 3.2.4 and recently run into following issue: Imagine simplest proxy setup - OpenSIPS just accepts new packet, for example INVITE, changes destination using 'sethostport(....)' and then issues 't_relay()' to forward the packet. Let's ignore replies and so on. If I'm doing a 'fix_nated_contact()' before sending this packet I'm expecting Contact: field to be replaced with a source IP:port as per manual. And this works if the Contact is in simple form like ' sip:7777777 at 192.168.29.106:65033'. But if following Contact comes in OpenSIPS doesn't change it leaving private IP in the contact: 'Contact: sip:7777777 at 192.168.29.106:65033;rinstance=2f59b175103f1088' Can you please let me know why is that happening? Thanks! Best regards, Yury. -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Apr 29 13:23:12 2022 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 29 Apr 2022 18:53:12 +0530 Subject: [OpenSIPS-Users] Query regarding AWS document BD using through Opensips . Message-ID: Hi All , I was exploring fullsharing-cachedb-cluster in opensips 3.2 . I have tested this with single stand alone mongo db instance . Its working perfectly fine . I know that AWS document DB is mongodb compatible . So I want to explore that . Because we are using AWS cloud for our deployment . I have created a single instance of Document DB cluster . I want to connect to that from the opensips script . There was no error while starting opensips . But when opensips tried to write data in the db its threw an error . * ERROR:usrloc:release_urecord: failed to flush AoR Default_Line_118_6 at p2p-cachedb.xyz.com ERROR:cachedb_mongodb:mongo_con_update: last error: 15.13053: No suitable servers found (`serverselectiontryonce` set): [Failed to resolve 'docdb-2022-04-27-10-26-28.cluster-cryhhicuxgzu.us-east-1.docdb.amdocdb-2022-04-27-10-26-28.cluster-cryhhicuxgzu.us-east-1.docdb.amazona'] ERROR:usrloc:cdb_flush_urecord: cache update query for AoR hynode2_CallDefault at p2p-cachedb.xyz.com failed!* My configuration file looks like below . same for usrloc and db_cachedb *modparam("cachedb_mongodb", "cachedb_url","mongodb://?master:opensips3 at docdb-2022.cluster-cryhhicuxgzu.us-east-1.docdb.amazonaws.com:27017/db.test/ ?ssl=true&ssl_ca_certs=/usr/local/src/etc/opensips/rds-combined-ca-bundle.pem&replicaSet=rs0&readPreference=secondaryPreferred&retryWrites=false")* In document db it also its says the same way we can connect to docdb from an application . I have tried so many ways to resolve this . But without any luck . Please help me out if anybody has used a document db through any application can also reply . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From kworm at missouri-telecom.com Fri Apr 29 13:33:10 2022 From: kworm at missouri-telecom.com (Kevin Wormington) Date: Fri, 29 Apr 2022 08:33:10 -0500 Subject: [OpenSIPS-Users] Query regarding AWS document BD using through Opensips . In-Reply-To: References: Message-ID: <47454D85-990A-4965-8A79-48F3D63CC4A1@missouri-telecom.com> Hi Samita, I don’t have any experience with AWS but from the error message OpenSIPS is logging the hostname of the server cannot be resolved. The hostname also appears to be truncated. Have you tried using the IP address instead of hostname or making a CNAME dns entry for the host that is shorter? Perhaps this is some parameter length limit. Kevin > On Apr 29, 2022, at 8:23 AM, Sasmita Panda wrote: > > Hi All , > > > I was exploring fullsharing-cachedb-cluster in opensips 3.2 . I have tested this with single stand alone mongo db instance . Its working perfectly fine . > > I know that AWS document DB is mongodb compatible . So I want to explore that . Because we are using AWS cloud for our deployment . > > I have created a single instance of Document DB cluster . I want to connect to that from the opensips script . There was no error while starting opensips . But when opensips tried to write data in the db its threw an error . > > > ERROR:usrloc:release_urecord: failed to flush AoR Default_Line_118_6 at p2p-cachedb.xyz.com > ERROR:cachedb_mongodb:mongo_con_update: last error: 15.13053: No suitable servers found (`serverselectiontryonce` set): [Failed to resolve 'docdb-2022-04-27-10-26-28.cluster-cryhhicuxgzu.us-east-1.docdb.amdocdb-2022-04-27-10-26-28.cluster-cryhhicuxgzu.us-east-1.docdb.amazona'] > ERROR:usrloc:cdb_flush_urecord: cache update query for AoR hynode2_CallDefault at p2p-cachedb.xyz.com failed! > > > My configuration file looks like below . same for usrloc and db_cachedb > > modparam("cachedb_mongodb", "cachedb_url","mongodb:// > ?master:opensips3 at docdb-2022.cluster-cryhhicuxgzu.us-east-1.docdb.amazonaws.com:27017/db.test/ > ?ssl=true&ssl_ca_certs=/usr/local/src/etc/opensips/rds-combined-ca-bundle.pem&replicaSet=rs0&readPreference=secondaryPreferred&retryWrites=false") > > In document db it also its says the same way we can connect to docdb from an application . > > I have tried so many ways to resolve this . But without any luck . Please help me out if anybody has used a document db through any application can also reply . > > > 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 From yannick.lecoent at nexcom.fr Fri Apr 29 22:43:15 2022 From: yannick.lecoent at nexcom.fr (Yannick LE COENT) Date: Sat, 30 Apr 2022 00:43:15 +0200 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission Message-ID: <278b4e3e-c412-8b45-fd71-d4ce319ce049@nexcom.fr> Hello, I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy is in charge of authenticating the request. This is the callflow: Alice           OpenSIPS          Proxy#2   | INVITE         |                |   |--------------->| INVITE         |   |      100 Tring |--------------->|   |<---------------|            407 |   |                |<---------------|   |                | ACK            |   |                |--------------->|   |            407 |                |   |     X<---------|                |   |                |                | Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. What can I do ? If I set auto_100trying=1, that works, but this increases the number of INVITE retransmissions since 180Ringing are not received instantly. Do you have any suggestion ? Thanks, Yannick From Ben.Newlin at genesys.com Sat Apr 30 00:59:44 2022 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Sat, 30 Apr 2022 00:59:44 +0000 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission In-Reply-To: <278b4e3e-c412-8b45-fd71-d4ce319ce049@nexcom.fr> References: <278b4e3e-c412-8b45-fd71-d4ce319ce049@nexcom.fr> Message-ID: Yannick, The default behavior of OpenSIPS is to relay any received responses back upstream. If it is not doing that it would have to be because you are stopping it in the script. Take a look at the documentation for failure_route [1] which explains this. Check your own failure_route in your script; you must be doing something there that is telling OpenSIPS not to relay the 401/407 back upstream. [1] https://www.opensips.org/Documentation/Script-Routes-2-4#toc3 Ben Newlin From: Users on behalf of Yannick LE COENT Date: Friday, April 29, 2022 at 6:44 PM To: users at lists.opensips.org Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission EXTERNAL EMAIL - Please use caution with links and attachments Hello, I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy is in charge of authenticating the request. This is the callflow: Alice OpenSIPS Proxy#2 | INVITE | | |--------------->| INVITE | | 100 Tring |--------------->| |<---------------| 407 | | |<---------------| | | ACK | | |--------------->| | 407 | | | X<---------| | | | | Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. What can I do ? If I set auto_100trying=1, that works, but this increases the number of INVITE retransmissions since 180Ringing are not received instantly. Do you have any suggestion ? Thanks, Yannick _______________________________________________ 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 yannick.lecoent at nexcom.fr Sat Apr 30 09:45:18 2022 From: yannick.lecoent at nexcom.fr (Yannick LE COENT) Date: Sat, 30 Apr 2022 11:45:18 +0200 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission Message-ID: <05c3bf91-3081-8815-e092-513035d82308@nexcom.fr> Hello Ben, The 407 is sent upstream, but when it is lost, it is not retransmitted by OpenSIPS. I do not have this problem with other negative status codes (e.g. 486). This is clearly explained in https://opensips.org/pub/opensips/1.8.6/src/ChangeLog 2012-03-21 18:36:58 Bogdan-Andrei Iancu,     * [8811] :     TM will no longer do retransmission for the 407/401 replies (if no ACK is received) for both local or proxied replies.     According to RFC 3261, retransmitting 407s/401s is probably a bad idea:     26.3.2.4 DoS Protection At the moment, my only solution is to use forward() instead of t_relay() in order to use the stateless mode. Yannick > Yannick, > > The default behavior of OpenSIPS is to relay any received responses back upstream. If it is not doing that it would have to be because you are stopping it in the script. Take a look at the documentation for failure_route [1] which explains this. Check your own failure_route in your script; you must be doing something there that is telling OpenSIPS not to relay the 401/407 back upstream. > > [1]https://www.opensips.org/Documentation/Script-Routes-2-4#toc3 > > Ben Newlin > > From: Users on behalf of Yannick LE COENT > Date: Friday, April 29, 2022 at 6:44 PM > To:users at lists.opensips.org > Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission > EXTERNAL EMAIL - Please use caution with links and attachments > > Hello, > > I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy > is in charge of authenticating the request. > > This is the callflow: > > Alice OpenSIPS Proxy#2 > | INVITE | | > |--------------->| INVITE | > | 100 Tring |--------------->| > |<---------------| 407 | > | |<---------------| > | | ACK | > | |--------------->| > | 407 | | > | X<---------| | > | | | > > Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. > > What can I do ? > If I set auto_100trying=1, that works, but this increases the number of > INVITE retransmissions since 180Ringing are not received instantly. > > Do you have any suggestion ? > > Thanks, > Yannick -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Sat Apr 30 14:15:38 2022 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Sat, 30 Apr 2022 14:15:38 +0000 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission In-Reply-To: <05c3bf91-3081-8815-e092-513035d82308@nexcom.fr> References: <05c3bf91-3081-8815-e092-513035d82308@nexcom.fr> Message-ID: I see. Apologies, I misunderstood the problem scenario. Ben Newlin From: Users on behalf of Yannick LE COENT Date: Saturday, April 30, 2022 at 5:46 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hello Ben, The 407 is sent upstream, but when it is lost, it is not retransmitted by OpenSIPS. I do not have this problem with other negative status codes (e.g. 486). This is clearly explained in https://opensips.org/pub/opensips/1.8.6/src/ChangeLog 2012-03-21 18:36:58 Bogdan-Andrei Iancu, * [8811] : TM will no longer do retransmission for the 407/401 replies (if no ACK is received) for both local or proxied replies. According to RFC 3261, retransmitting 407s/401s is probably a bad idea: 26.3.2.4 DoS Protection At the moment, my only solution is to use forward() instead of t_relay() in order to use the stateless mode. Yannick Yannick, The default behavior of OpenSIPS is to relay any received responses back upstream. If it is not doing that it would have to be because you are stopping it in the script. Take a look at the documentation for failure_route [1] which explains this. Check your own failure_route in your script; you must be doing something there that is telling OpenSIPS not to relay the 401/407 back upstream. [1] https://www.opensips.org/Documentation/Script-Routes-2-4#toc3 Ben Newlin From: Users on behalf of Yannick LE COENT Date: Friday, April 29, 2022 at 6:44 PM To: users at lists.opensips.org Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission EXTERNAL EMAIL - Please use caution with links and attachments Hello, I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy is in charge of authenticating the request. This is the callflow: Alice OpenSIPS Proxy#2 | INVITE | | |--------------->| INVITE | | 100 Tring |--------------->| |<---------------| 407 | | |<---------------| | | ACK | | |--------------->| | 407 | | | X<---------| | | | | Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. What can I do ? If I set auto_100trying=1, that works, but this increases the number of INVITE retransmissions since 180Ringing are not received instantly. Do you have any suggestion ? Thanks, Yannick -------------- next part -------------- An HTML attachment was scrubbed... URL: From yannick.lecoent at nexcom.fr Sat Apr 30 16:14:48 2022 From: yannick.lecoent at nexcom.fr (Yannick LE COENT) Date: Sat, 30 Apr 2022 18:14:48 +0200 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission In-Reply-To: References: <05c3bf91-3081-8815-e092-513035d82308@nexcom.fr> Message-ID: <93b1545a-b0ed-0826-863b-3dfbf90371c4@nexcom.fr> Hello Ben, Thanks for your answer. This problem occurs when OpenSIPS is not in charge of authenticating the INVITE request, but this is done downstream. I've sent this question to know if somebody has already solved this kind of problem. Best regards, Yannick Le 30/04/2022 à 16:15, Ben Newlin a écrit : > > I see. Apologies, I misunderstood the problem scenario. > > Ben Newlin > > *From: *Users on behalf of Yannick > LE COENT > *Date: *Saturday, April 30, 2022 at 5:46 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission > > *EXTERNAL EMAIL - Please use caution with links and attachments * > > ------------------------------------------------------------------------ > > Hello Ben, > > The 407 is sent upstream, but when it is lost, it is not retransmitted > by OpenSIPS. > I do not have this problem with other negative status codes (e.g. 486). > > This is clearly explained in > https://opensips.org/pub/opensips/1.8.6/src/ChangeLog > > 2012-03-21 18:36:58  Bogdan-Andrei Iancu, org> >     * [8811] : > >     TM will no longer do retransmission for the 407/401 replies > (if no ACK is received) for both local or proxied replies. > > According to RFC 3261, retransmitting 407s/401s is probably a bad > idea: > >  26.3.2.4 DoS Protection > > At the moment, my only solution is to use forward() instead of > t_relay() in order to use the stateless mode. > > Yannick > > > Yannick, > > The default behavior of OpenSIPS is to relay any received responses back upstream. If it is not doing that it would have to be because you are stopping it in the script. Take a look at the documentation for failure_route [1] which explains this. Check your own failure_route in your script; you must be doing something there that is telling OpenSIPS not to relay the 401/407 back upstream. > > [1]https://www.opensips.org/Documentation/Script-Routes-2-4#toc3 > > Ben Newlin > > From: Users on behalf of Yannick LE COENT > > Date: Friday, April 29, 2022 at 6:44 PM > > To:users at lists.opensips.org > > Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission > > EXTERNAL EMAIL - Please use caution with links and attachments > > Hello, > > I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy > > is in charge of authenticating the request. > > This is the callflow: > > Alice           OpenSIPS          Proxy#2 > >    | INVITE         |                | > >    |--------------->| INVITE         | > >    |      100 Tring |--------------->| > >    |<---------------|            407 | > >    |                |<---------------| > >    |                | ACK            | > >    |                |--------------->| > >    |            407 |                | > >    |     X<---------|                | > >    |                |                | > > Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. > > What can I do ? > > If I set auto_100trying=1, that works, but this increases the number of > > INVITE retransmissions since 180Ringing are not received instantly. > > Do you have any suggestion ? > > Thanks, > > Yannick > > > _______________________________________________ > 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 hobe69 at hotmail.com Tue Apr 5 02:29:11 2022 From: hobe69 at hotmail.com (Bela H) Date: Tue, 05 Apr 2022 02:29:11 -0000 Subject: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? Message-ID: Hello, I want to set up a call forwarding and followed the instructions from the "Building telephony systems with OpenSIPS". However, this is a little bit outdated, the old opensipsctl was replaced by opensips-cli. How can I add an avp data into a usr_preferences table from opensips-cli? This was the old format: opensipsctl avp add A_number callfwd 0 C_number Cheers, Bela -------------- next part -------------- An HTML attachment was scrubbed... URL: From di-shi.sun at transnexus.com Tue Apr 26 13:01:25 2022 From: di-shi.sun at transnexus.com (Di-Shi Sun) Date: Tue, 26 Apr 2022 13:01:25 -0000 Subject: [OpenSIPS-Users] pjsip patch supporting sip 3xx embedded headers Message-ID: All, I committed a pjsip patch pull request in pjsip project to support SIP 3xx embedded header. It inserts stand-alone SIP headers based on SIP 3xx Contact header URI header parameters into the redirect SIP INVITE messages. It is a common feature supported by any SIP implementations, such as Metaswitch, Oracle SBC, Ribbon, etc. It should solve the issue that it is difficult to pass useful information using SIP 3xx in Asterisk. If anybody wants to try, the patch link is attached. https://github.com/pjsip/pjproject/pull/3085/commits/b065eef9913df144445757808b352c2ce9054809 To use this patch, please follow the steps 1. Download Asterisk 18 source code 2. Run “./configure --with-pjproject-bundled --with-jansson-bundled”. 3. Apply the patch to ./third-party/pjproject/source/pjsip/src/pjsip-ua/sip_inv.c 4. Build/install/run Asterisk 5. In pjsip.conf, the redirect server endpoint should be configured with “redirect_method=uri_pjsip” 6. When the redirect server returns a SIP 3xx with Contact header, say “Contact: ;q=0.99”, the redirected SIP INVITE message should contain “Test: test” header. Please me know if you have any questions. Sincerely, Di-Shi Sun. -------------- next part -------------- An HTML attachment was scrubbed... URL: