From amel.guesmi at sofrecom.com Fri Aug 2 08:06:03 2024 From: amel.guesmi at sofrecom.com (amel.guesmi at sofrecom.com) Date: Fri, 2 Aug 2024 08:06:03 +0000 Subject: [OpenSIPS-Users] [Opensips DB]- issue to keep one database view Message-ID: Hello Community, I have a question regarding the OpenSIPS database setup. I'm currently using a Galera cluster with two OpenSIPS instances, and I'm encountering an issue with maintaining a unified database view. Specifically, in certain tables such as Dispatcher and Location, the Socket column always contains IP addresses that are specific to each VM. This VM-specific data prevents a unified view of the database, causing inconsistencies between the two OpenSIPS instances. How can I ensure that both VMs share a unified database view without relying on VM-specific IP addresses in the Socket column? Thank you in advance for your assistance. On belahf of my colleague Wissal -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Fri Aug 2 14:57:50 2024 From: igorolhovskiy at gmail.com (Ihor Olkhovskyi) Date: Fri, 2 Aug 2024 16:57:50 +0200 Subject: [OpenSIPS-Users] DTMF Issue In-Reply-To: References: Message-ID: Alex, OpenSIPS is a SIP proxy. DTMF method you're mentioning is RTP-based. So, you need to make sure you're passing RTP streams correctly. And on client side this should be configured as well. But just to highlight again, DTMF (unless SIP INFO) is not a question related to OpenSIPS. Le 31/07/2024 à 17:49, Alexander Perkins a écrit : > Hi All.  We have an issue where when we send a call to our client, > they say the DTMF is not working and we've confirmed that to be the > case.  Our carrier is sending over RFC2833.  How can we set the DTMF > mode in OpenSIPS?  What is the recommended approach? > > Thank you, > Alex > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bullehs at gmail.com Mon Aug 5 07:11:55 2024 From: bullehs at gmail.com (HS) Date: Mon, 5 Aug 2024 12:11:55 +0500 Subject: [OpenSIPS-Users] Opensips+RTPEngine Billing Solution on AWS Message-ID: Hi all, I have Opensips 3.0 + RTPengine setup on an AWS instance. I tried CGRates recently, but couldn't get it to work (calls weren't redirected to the GW if allowed, however, were declined if the call wasn't authorised). Does anyone else have any experience with the above issue? Or is some other opensource billing software that works better with Opensips+RTPEngine setup? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Aug 6 08:22:04 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Aug 2024 11:22:04 +0300 Subject: [OpenSIPS-Users] [Opensips DB]- issue to keep one database view In-Reply-To: References: Message-ID: Hi Amel & Wissal, The `location` table should not be shared across multiple OpenSIPS instances, it should be one per OpenSIPS ( similar the `dialog` or presence related tables) For the `dispatcher` table, the `socket` column is optional. But if you have to use it, use the socket definition via socket tag (see [1]) - of course, use the same socket tag in all OpenSIPS cfg instances. [1] https://www.opensips.org/Documentation/Script-CoreParameters-3-4#socket Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 02.08.2024 11:06, amel.guesmi at sofrecom.com wrote: > > Hello Community, > > I have a question regarding the OpenSIPS database setup. I'm currently > using a Galera cluster with two OpenSIPS instances, and I'm > encountering an issue with maintaining a unified database view. > > Specifically, in certain tables such as *Dispatcher* and *Location,* > the *Socket *column always contains IP addresses that are specific to > each VM. This VM-specific data prevents a unified view of the > database, causing inconsistencies between the two OpenSIPS instances. > > How can I ensure that both VMs share a unified database view without > relying on VM-specific IP addresses in the Socket column? > >  Thank you in advance for your assistance. > > On belahf of my colleague Wissal > > > _______________________________________________ > 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 Aug 6 08:25:56 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Aug 2024 11:25:56 +0300 Subject: [OpenSIPS-Users] DIfference between R-URI and Destination URI in openSIPs In-Reply-To: References: <9437C142-614D-400A-9D51-0BEE9C22D0DC@evaristesys.com> Message-ID: <53c54242-ac0d-47dc-86a3-61972a688465@opensips.org> Hi Mohamed, Use t_relay(), the (transactional) statefull way to send out a SIP request. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 31.07.2024 14:46, Mohamed OUALLA wrote: > [...] > >   One last question: What is the most effective way to connect the > caller and callee in a simple VoIP call setup (Caller - SIP Proxy - > Callee)? Should I change the *$ru* and then use *forward()* for > stateless routing or *t_relay()* for stateful routing, or is there > another method you would recommend? > > Mohamed. > > > On Wed, Jul 31, 2024 at 4:00 AM Alex Balashov > wrote: > > Hi, > > The Request URI is a SIP concept, while the destination URI might > be best described as a fictive invention of OpenSIPS configuration > script. It represents the next-hop destination to which the > request will be forwarded on the network and transport layer, as > you correctly surmised, while the request URI is a logical > destination. The destination URI supersedes the request URI, but > if the destination URI is not set, the domain/port/transport > attributes of the request URI are consumed to determine the > forwarding destination. > > An RURI is not the same thing as an Address of Record; an AoR > refers to a logical URI entity in the location service (registrar) > context. The purpose of a registrar is to map an AoR (such as > sip:mohamed at sip.opensips.org > ) to one or more Contact > URIs (e.g. sip:line1 at 192.168.1.100 > ;user=phone), which indicate how > to reach a given device on the network and transport layer. > > Hopefully that helps! > > -- Alex > > > On Jul 30, 2024, at 9:26 PM, Mohamed OUALLA > wrote: > > > > Hello all, > > > >   I have a technical question about the difference between > Request URI and Destination URI in SIP. In my understanding of > SIP, the R-URI (Request URI) is located in the start line of the > SIP request and is also known as the Address of Record (AoR). > However, I am unclear about what the Destination URI is for > openSIPs. Is it the same as the Request URI, or is it related to > an added route header, or the destination address in the transport > protocol, I am not sure about it? > > > >   Additionally, I have observed that when I change the $du > pseudo variable in OpenSIPS, it relays the request to the UAS > without changing the R-URI (change it with the sip uri I gave to > $du pseudo variable). This behavior is the same as using the > t_relay() method, which also does not change the R-URI but sends > the request to the UAS. I guess that changes have been done only > for the destination address in the transport layer. > > > >   Could someone please explain these observations and clarify > the difference between R-URI and Destination URI? > >   And the best way to route calls from UAC to UAS in simple VoIP > call components (Caller - SIP Proxy - Callee), actually I change > the $ru, then I forward() the request stateless or t_relay() stateful. > > > > Thank you. > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Alex Balashov > Principal Consultant > Evariste Systems LLC > Web: https://evaristesys.com > Tel: +1-706-510-6800 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > ============================== > Mohamed OUALLA > *VoIP Technical Solutions and Software Engineer* > Mail: oualla.simohamed at gmail.com > N.Phone: +212 6 29 19 3116 > *SSC Certified Professional* > ============================== > > _______________________________________________ > 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 Aug 6 08:39:02 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Aug 2024 11:39:02 +0300 Subject: [OpenSIPS-Users] I need some info while setiing sethostport on opensips config . In-Reply-To: References: Message-ID: Hi Sasmita, Going back to your original idea, with sethostport(). What you need to do is : 1) strip any potential `transport` param from RURI (as it may force TLS)     ruri_del_param("transport"); 2) set as outbound socket an UDP one     $socket_out = "udp:192.168.0.69:5060"; Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.07.2024 12:50, Sasmita Panda wrote: > Any suggestions on this ? > > > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Fri, Jul 26, 2024 at 4:55 PM Sasmita Panda wrote: > > Hi , Thanks for the reply . > Rather than configuration change , I have done through dynamic > routing . > >  In the dr_gateway table I have added socket information > *udp:x.x.x.x:5060* > mysql> select * from dr_gateways; > +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+ > | id | gwid | type | address | strip | pri_prefix | attrs | > probe_mode | state | socket                | description | > +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+ > |  1 | gw4  |    3 | fs.3c.com:6080 |     > 0 | NULL       | NULL  |          0 |     0 | > udp:192.168.0.69:5060 | NULL    | > +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+ > > I only have a single Invite on a single session on this leg . > There is no Re-Invite at all  . Will I need to set > record_route_preset on this case as well ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Wed, Jul 24, 2024 at 7:06 PM Ben Newlin > wrote: > > Sasmita, > > You need to be using $socket_out. [1] > > By default, OpenSIPS will use the receiving socket as the > sending socket. This means if you receive the message on TLS > and do not change $socket_out then the message will be sent > out the TLS socket. > > Additionally, unless you are using B2BUA or maybe > topology_hiding you will likely need to “remember” this > protocol transition by adding a double Record-Route [2] to the > message. You may also need to change any “transport” params > that exist in the Request-URI and possibly the Contact header. > > [1] > https://www.opensips.org/Documentation/Script-CoreVar-3-2#toc86 > > [2] https://opensips.org/docs/modules/3.2.x/rr.html > > Ben Newlin > > *From: *Users on behalf of > Sasmita Panda > *Date: *Wednesday, July 24, 2024 at 1:54 AM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] I need some info while setiing > sethostport on opensips config . > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > ------------------------------------------------------------------------ > > Hi All , > > I am using openisp version : 3.2 > > I have an opensips config on which I was listening on UDP port > only and hence using sethostport to route calls to a > particular destination . like below . > > if(is_from_gw() || ($rp=~"5505")) >                    { >                   sethostport("freeswitch-test.xyz.com:6080 > "); >                    route(inbound); >                     exit; >                    } > > Now I have to accept a call on TLS and send that to some other > destination on UDP . I have enabled the tls module and also > the dependent modules like tls_openssl, tls_mgm . > > socket=udp:192.168.0.y:5060 > socket=tls:192.168.0.y:5061 > socket=tcp:192.168.0.y:5060 > > if(is_from_gw() || ($rp=~"5505")) >                    { >                   sethostport("freeswitch-test.xyz.com:6080 > "); >                    route(inbound); >                     exit; >                    } > > This above configuration is not working . I am getting "477 > Send Failed " > > */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 bogdan at opensips.org Tue Aug 6 08:42:59 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Aug 2024 11:42:59 +0300 Subject: [OpenSIPS-Users] Users Digest, Vol 192, Issue 26 In-Reply-To: References: Message-ID: Hi, There may be many reason for disconnecting the call (SST may be one of them, indeed) - but better check in the BYE you get if there is any `Reason` hdr, explaining why the disconnect was trigger. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 27.07.2024 20:33, Alexander Perkins wrote: > Thank you!  Is there a way to extend that timer? Or a way to catch it > in a block and ignore it?  Basically, I’m trying to figure out the > best approach to prevent that from happening. > > Thank you, > Alex > > On Sat, Jul 27, 2024 at 08:01 wrote: > > Send Users mailing list submissions to > users at lists.opensips.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request at lists.opensips.org > > You can reach the person managing the list at > users-owner at lists.opensips.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > >    1. Calls Disconnecting after 10 Minutes (Alexander Perkins) >    2. Re: Calls Disconnecting after 10 Minutes (M S) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 27 Jul 2024 06:19:21 -0400 > From: Alexander Perkins > To: OpenSIPS users mailling list > Subject: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes > Message-ID: >         > > Content-Type: text/plain; charset="utf-8" > > Hi All.  We have an interesting situation.  If we send calls to our > OpenSIPS proxy, they will disconnect after 600 seconds. However, if we > send them directly to our switch, this does not happen. Is there a > timer, > or something, I should be aware of?  If so, can someone point me > in the > right direction?  Any help would be appreciated.  I've been trying to > figure this out for around three months. > > Thank you, > Alex > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > > > ------------------------------ > > Message: 2 > Date: Sat, 27 Jul 2024 12:25:47 +0200 > From: M S > To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes > Message-ID: >         > > > Content-Type: text/plain; charset="utf-8" > > It can be Session-Expires header, or reinvites (in-dialog invite > being sent > after 600 seconds) > > On Sat, Jul 27, 2024 at 12:22 PM Alexander Perkins < > alexanderhenryperkins at gmail.com> wrote: > > > Hi All.  We have an interesting situation.  If we send calls to our > > OpenSIPS proxy, they will disconnect after 600 seconds.  > However, if we > > send them directly to our switch, this does not happen. Is there > a timer, > > or something, I should be aware of?  If so, can someone point me > in the > > right direction?  Any help would be appreciated.  I've been > trying to > > figure this out for around three months. > > > > Thank you, > > Alex > > _______________________________________________ > > 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: > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ------------------------------ > > End of Users Digest, Vol 192, Issue 26 > ************************************** > > > _______________________________________________ > 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 Tue Aug 6 10:27:04 2024 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 6 Aug 2024 15:57:04 +0530 Subject: [OpenSIPS-Users] I need some info while setiing sethostport on opensips config . In-Reply-To: References: Message-ID: Thank you . I will try this . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, Aug 6, 2024 at 2:09 PM Bogdan-Andrei Iancu wrote: > Hi Sasmita, > > Going back to your original idea, with sethostport(). What you need to do > is : > > 1) strip any potential `transport` param from RURI (as it may force TLS) > ruri_del_param("transport"); > > 2) set as outbound socket an UDP one > $socket_out = "udp:192.168.0.69:5060"; > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 29.07.2024 12:50, Sasmita Panda wrote: > > Any suggestions on this ? > > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Fri, Jul 26, 2024 at 4:55 PM Sasmita Panda wrote: > >> Hi , Thanks for the reply . >> Rather than configuration change , I have done through dynamic routing . >> >> In the dr_gateway table I have added socket information >> *udp:x.x.x.x:5060* >> mysql> select * from dr_gateways; >> >> +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+ >> | id | gwid | type | address | strip | >> pri_prefix | attrs | probe_mode | state | socket | >> description | >> >> +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+ >> | 1 | gw4 | 3 | fs.3c.com:6080 | 0 | NULL | NULL | >> 0 | 0 | udp:192.168.0.69:5060 | NULL | >> >> +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+ >> >> I only have a single Invite on a single session on this leg . There is no >> Re-Invite at all . Will I need to set record_route_preset on this case as >> well ? >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> On Wed, Jul 24, 2024 at 7:06 PM Ben Newlin >> wrote: >> >>> Sasmita, >>> >>> >>> >>> You need to be using $socket_out. [1] >>> >>> >>> >>> By default, OpenSIPS will use the receiving socket as the sending >>> socket. This means if you receive the message on TLS and do not change >>> $socket_out then the message will be sent out the TLS socket. >>> >>> >>> >>> Additionally, unless you are using B2BUA or maybe topology_hiding you >>> will likely need to “remember” this protocol transition by adding a double >>> Record-Route [2] to the message. You may also need to change any >>> “transport” params that exist in the Request-URI and possibly the Contact >>> header. >>> >>> >>> >>> [1] https://www.opensips.org/Documentation/Script-CoreVar-3-2#toc86 >>> >>> [2] https://opensips.org/docs/modules/3.2.x/rr.html >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> *From: *Users on behalf of Sasmita >>> Panda >>> *Date: *Wednesday, July 24, 2024 at 1:54 AM >>> *To: *OpenSIPS users mailling list >>> *Subject: *[OpenSIPS-Users] I need some info while setiing sethostport >>> on opensips config . >>> >>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> >>> ------------------------------ >>> >>> Hi All , >>> >>> >>> >>> I am using openisp version : 3.2 >>> >>> >>> >>> I have an opensips config on which I was listening on UDP port only and >>> hence using sethostport to route calls to a particular destination . like >>> below . >>> >>> >>> >>> if(is_from_gw() || ($rp=~"5505")) >>> { >>> sethostport("freeswitch-test.xyz.com:6080"); >>> route(inbound); >>> exit; >>> } >>> >>> >>> >>> Now I have to accept a call on TLS and send that to some other >>> destination on UDP . I have enabled the tls module and also the dependent >>> modules like tls_openssl, tls_mgm . >>> >>> >>> >>> socket=udp:192.168.0.y:5060 >>> socket=tls:192.168.0.y:5061 >>> socket=tcp:192.168.0.y:5060 >>> >>> if(is_from_gw() || ($rp=~"5505")) >>> { >>> sethostport("freeswitch-test.xyz.com:6080"); >>> route(inbound); >>> exit; >>> } >>> >>> >>> >>> This above configuration is not working . I am getting "477 Send Failed " >>> >>> >>> >>> *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 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 Tue Aug 6 12:20:46 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Aug 2024 15:20:46 +0300 Subject: [OpenSIPS-Users] CANCEL cross 200OK In-Reply-To: References: Message-ID: Hi Mickael, Once the UAS generated a 200 OK reply, there is not much you can do - you have to send this 200 to UAC. And the UAC will have to ACK it and after that to decide if to keep the call or not - if not, it will send a BYE on the spot. A proxy is not the place to deal with the typical CANCEL/200ok race in SIP, but to let the UAC to deal with it. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 17.07.2024 12:30, Mickael Hubert wrote: > Hi all, > I have an issue in my OpenSIPS proxy (version: opensips 3.3.4 > (x86_64/linux)). > Proxy receives 200OK from UAS, but in the same time, receives CANCEL > from UAC. > Ex: > ...... > UAS --> 200OK (SDP) --> proxy >                         proxy <--  CANCEL <-- UAC >                         proxy --> 200 CANCELING --> UAC >                         proxy --> 200OK  (SDP) --> UAC >                         proxy <-- ACK <-- UAC > UAS <-- ACK <--         proxy > ..... > > I want to find a solution that proxy sends 487 to UAC, and BYE to UAS. > How can I do that please ? There is a function or I have to code all > this scenario ? > > thanks in advance > Have a good day > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Tue Aug 6 12:22:19 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Aug 2024 15:22:19 +0300 Subject: [OpenSIPS-Users] INVITE brings in extra transport=UDP parameter In-Reply-To: References: Message-ID: Hi, Just do ruri_del_param("transport"); Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 12.07.2024 09:20, Saint Michael wrote: > INVITE sip:19206661392 at 38.95.11.250;transport=UDP > when this happens, opensips has a contaminated ru variable > and since it's read-only, I cannot fix it in code > I tried > if ($ru =~ ";transport=UDP") { > $ru = $(ru{s.select,0,-13}); > but it makes no difference, ru does not change > How do I get around this? > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Tue Aug 6 12:24:01 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Aug 2024 15:24:01 +0300 Subject: [OpenSIPS-Users] Maybe this is a bug version 3.4 latest In-Reply-To: References: Message-ID: <582429ea-2f04-48cf-b2cd-ab0f86c5ccd5@opensips.org> Hi, There is not bug, it is most probably just you messing (in script) the RURI to a format that is SIP invalid. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 12.07.2024 08:56, Saint Michael wrote: > (xlog) NOTICE:ru sip:19206661392 at 1.1.1.1;transport=UDP rU 19206661392 > DST= 1.1.1.1 rd= 1.1.1.1 > CRITICAL:core:mk_proxy: could not resolve hostname: ";transport=UDP" > ERROR:tm:uri2proxy: bad host name in URI > ERROR:tm:t_forward_nonack: failure to add branches > ERROR:tm:w_t_relay: t_forward_nonack failed > My box is multihomed, and I am using rttpproxy. > this: socket=udp:*:5060 use_workers 80 > or this: socket=udp:1.1.1.1:5060 use_workers 80 > make no difference > also to use rttpproxy or not makes no difference > The issue seems to be caused by ru > sip:19206661392 at 1.1.1.1;transport=UDP having now the ;transport=UDP, > but I only use RTP, so the tansport= part is irrelevant > > The main problem is that my CDR is useless > because if the call fails, the field dst_ip comes as ";transport=UDP" > when it should have been "1.1.1.1" > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From alexanderhenryperkins at gmail.com Wed Aug 7 12:03:08 2024 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Wed, 7 Aug 2024 08:03:08 -0400 Subject: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes In-Reply-To: References: Message-ID: Hi Bogdan. Thank you very much. We will take a look at it. On Tue, Aug 6, 2024 at 04:43 Bogdan-Andrei Iancu wrote: > Hi, > > There may be many reason for disconnecting the call (SST may be one of > them, indeed) - but better check in the BYE you get if there is any > `Reason` hdr, explaining why the disconnect was trigger. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 27.07.2024 20:33, Alexander Perkins wrote: > > Thank you! Is there a way to extend that timer? Or a way to catch it in > a block and ignore it? Basically, I’m trying to figure out the best > approach to prevent that from happening. > > Thank you, > Alex > > On Sat, Jul 27, 2024 at 08:01 wrote: > >> Send Users mailing list submissions to >> users at lists.opensips.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> or, via email, send a message with subject or body 'help' to >> users-request at lists.opensips.org >> >> You can reach the person managing the list at >> users-owner at lists.opensips.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of Users digest..." >> >> >> Today's Topics: >> >> 1. Calls Disconnecting after 10 Minutes (Alexander Perkins) >> 2. Re: Calls Disconnecting after 10 Minutes (M S) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Sat, 27 Jul 2024 06:19:21 -0400 >> From: Alexander Perkins >> To: OpenSIPS users mailling list >> Subject: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes >> Message-ID: >> < >> CALLkTp0xAb9U9XVdnb3oVBooxNMGt9U3rg5fns1sBqOP-SGM_g at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Hi All. We have an interesting situation. If we send calls to our >> OpenSIPS proxy, they will disconnect after 600 seconds. However, if we >> send them directly to our switch, this does not happen. Is there a timer, >> or something, I should be aware of? If so, can someone point me in the >> right direction? Any help would be appreciated. I've been trying to >> figure this out for around three months. >> >> Thank you, >> Alex >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: < >> http://lists.opensips.org/pipermail/users/attachments/20240727/7613a46e/attachment-0001.html >> > >> >> ------------------------------ >> >> Message: 2 >> Date: Sat, 27 Jul 2024 12:25:47 +0200 >> From: M S >> To: OpenSIPS users mailling list >> Subject: Re: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes >> Message-ID: >> > YN4mE+3cE+9ihUsiOTYp-28xXjiecV0sNG6A at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> It can be Session-Expires header, or reinvites (in-dialog invite being >> sent >> after 600 seconds) >> >> On Sat, Jul 27, 2024 at 12:22 PM Alexander Perkins < >> alexanderhenryperkins at gmail.com> wrote: >> >> > Hi All. We have an interesting situation. If we send calls to our >> > OpenSIPS proxy, they will disconnect after 600 seconds. However, if we >> > send them directly to our switch, this does not happen. Is there a >> timer, >> > or something, I should be aware of? If so, can someone point me in the >> > right direction? Any help would be appreciated. I've been trying to >> > figure this out for around three months. >> > >> > Thank you, >> > Alex >> > _______________________________________________ >> > 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: < >> http://lists.opensips.org/pipermail/users/attachments/20240727/75945c69/attachment-0001.html >> > >> >> ------------------------------ >> >> Subject: Digest Footer >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> ------------------------------ >> >> End of Users Digest, Vol 192, Issue 26 >> ************************************** >> > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexandra.titoc at opensips.org Wed Aug 7 13:37:26 2024 From: alexandra.titoc at opensips.org (Alexandra Titoc) Date: Wed, 7 Aug 2024 16:37:26 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?=5BBlog=5D_Amazon=E2=80=99s_DynamoDB_i?= =?utf-8?q?ntegration_in_OpenSIPS_3=2E6?= Message-ID: <485a959f-33ff-4133-8bf4-c57b55823e1a@opensips.org> Hello! OpenSIPS 3.6 (development version) turns an eye on backends that facilitate AWS Cloud integration, like DynamoDB NoSQL database. DynamoDB is a powerful but scalable database that can be used locally or as DBaaS in AWS. https://blog.opensips.org/2024/08/07/amazons-dynamodb-integration-in-opensips-3-6/ Enjoy, -- Alexandra Titoc OpenSIPS Developer https://www.opensips.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexanderhenryperkins at gmail.com Thu Aug 8 14:35:22 2024 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Thu, 8 Aug 2024 10:35:22 -0400 Subject: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes In-Reply-To: References: Message-ID: Bogdan, we think we found the issue and it appears to be related to an OPTIONS keep-alive from our switch. Basically, our switch would send the keep-alive, but the proxy's server was not set up to allow inbound connections from that switch. So, it would not respond and the switch would kill the call. We are working this theory today, but that appears to be it. Thanks again for all your help with this. On Thu, Aug 8, 2024 at 8:01 AM wrote: > Send Users mailing list submissions to > users at lists.opensips.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request at lists.opensips.org > > You can reach the person managing the list at > users-owner at lists.opensips.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: Calls Disconnecting after 10 Minutes (Alexander Perkins) > 2. [Blog] Amazon’s DynamoDB integration in OpenSIPS 3.6 > (Alexandra Titoc) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 7 Aug 2024 08:03:08 -0400 > From: Alexander Perkins > To: Bogdan-Andrei Iancu > Cc: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes > Message-ID: > dZMw at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi Bogdan. Thank you very much. We will take a look at it. > > On Tue, Aug 6, 2024 at 04:43 Bogdan-Andrei Iancu > wrote: > > > Hi, > > > > There may be many reason for disconnecting the call (SST may be one of > > them, indeed) - but better check in the BYE you get if there is any > > `Reason` hdr, explaining why the disconnect was trigger. > > > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > https://www.siphub.com > > > > On 27.07.2024 20:33, Alexander Perkins wrote: > > > > Thank you! Is there a way to extend that timer? Or a way to catch it in > > a block and ignore it? Basically, I’m trying to figure out the best > > approach to prevent that from happening. > > > > Thank you, > > Alex > > > > On Sat, Jul 27, 2024 at 08:01 wrote: > > > >> Send Users mailing list submissions to > >> users at lists.opensips.org > >> > >> To subscribe or unsubscribe via the World Wide Web, visit > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >> or, via email, send a message with subject or body 'help' to > >> users-request at lists.opensips.org > >> > >> You can reach the person managing the list at > >> users-owner at lists.opensips.org > >> > >> When replying, please edit your Subject line so it is more specific > >> than "Re: Contents of Users digest..." > >> > >> > >> Today's Topics: > >> > >> 1. Calls Disconnecting after 10 Minutes (Alexander Perkins) > >> 2. Re: Calls Disconnecting after 10 Minutes (M S) > >> > >> > >> ---------------------------------------------------------------------- > >> > >> Message: 1 > >> Date: Sat, 27 Jul 2024 06:19:21 -0400 > >> From: Alexander Perkins > >> To: OpenSIPS users mailling list > >> Subject: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes > >> Message-ID: > >> < > >> CALLkTp0xAb9U9XVdnb3oVBooxNMGt9U3rg5fns1sBqOP-SGM_g at mail.gmail.com> > >> Content-Type: text/plain; charset="utf-8" > >> > >> Hi All. We have an interesting situation. If we send calls to our > >> OpenSIPS proxy, they will disconnect after 600 seconds. However, if we > >> send them directly to our switch, this does not happen. Is there a > timer, > >> or something, I should be aware of? If so, can someone point me in the > >> right direction? Any help would be appreciated. I've been trying to > >> figure this out for around three months. > >> > >> Thank you, > >> Alex > >> -------------- next part -------------- > >> An HTML attachment was scrubbed... > >> URL: < > >> > http://lists.opensips.org/pipermail/users/attachments/20240727/7613a46e/attachment-0001.html > >> > > >> > >> ------------------------------ > >> > >> Message: 2 > >> Date: Sat, 27 Jul 2024 12:25:47 +0200 > >> From: M S > >> To: OpenSIPS users mailling list > >> Subject: Re: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes > >> Message-ID: > >> >> YN4mE+3cE+9ihUsiOTYp-28xXjiecV0sNG6A at mail.gmail.com> > >> Content-Type: text/plain; charset="utf-8" > >> > >> It can be Session-Expires header, or reinvites (in-dialog invite being > >> sent > >> after 600 seconds) > >> > >> On Sat, Jul 27, 2024 at 12:22 PM Alexander Perkins < > >> alexanderhenryperkins at gmail.com> wrote: > >> > >> > Hi All. We have an interesting situation. If we send calls to our > >> > OpenSIPS proxy, they will disconnect after 600 seconds. However, if > we > >> > send them directly to our switch, this does not happen. Is there a > >> timer, > >> > or something, I should be aware of? If so, can someone point me in > the > >> > right direction? Any help would be appreciated. I've been trying to > >> > figure this out for around three months. > >> > > >> > Thank you, > >> > Alex > >> > _______________________________________________ > >> > 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: < > >> > http://lists.opensips.org/pipermail/users/attachments/20240727/75945c69/attachment-0001.html > >> > > >> > >> ------------------------------ > >> > >> Subject: Digest Footer > >> > >> _______________________________________________ > >> Users mailing list > >> Users at lists.opensips.org > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >> > >> > >> ------------------------------ > >> > >> End of Users Digest, Vol 192, Issue 26 > >> ************************************** > >> > > > > _______________________________________________ > > Users mailing listUsers at lists.opensips.orghttp:// > lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.opensips.org/pipermail/users/attachments/20240807/0b574453/attachment-0001.html > > > > ------------------------------ > > Message: 2 > Date: Wed, 7 Aug 2024 16:37:26 +0300 > From: Alexandra Titoc > To: users at lists.opensips.org, devel at lists.opensips.org > Subject: [OpenSIPS-Users] [Blog] Amazon’s DynamoDB integration in > OpenSIPS 3.6 > Message-ID: <485a959f-33ff-4133-8bf4-c57b55823e1a at opensips.org> > Content-Type: text/plain; charset="utf-8"; Format="flowed" > > Hello! > > OpenSIPS 3.6 (development version) turns an eye on backends that > facilitate AWS Cloud integration, like DynamoDB NoSQL database. DynamoDB > is a powerful but scalable database that can be used locally or as DBaaS > in AWS. > > > https://blog.opensips.org/2024/08/07/amazons-dynamodb-integration-in-opensips-3-6/ > < > https://blog.opensips.org/2024/08/07/amazons-dynamodb-integration-in-opensips-3-6/ > > > > Enjoy, > > -- > Alexandra Titoc > > OpenSIPS Developer > https://www.opensips.org > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.opensips.org/pipermail/users/attachments/20240807/a89c2bf4/attachment-0001.html > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ------------------------------ > > End of Users Digest, Vol 193, Issue 8 > ************************************* > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexanderhenryperkins at gmail.com Thu Aug 8 14:52:22 2024 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Thu, 8 Aug 2024 10:52:22 -0400 Subject: [OpenSIPS-Users] Variable Question Message-ID: Hi All. If I want to have a variable set in the route block and available to the failure route, which would be the appropriate way to do so? An AVP? Or is there a better way? For example: route { #do something route(DO_INVITE) } route [DO_INVITE] { #set variable $var(x) = 123456; t_on_failure("IT_FAILED"); } failure_route [IT_FAILED] { xlog("L_INFO", "Variable X = $var(x)); } Any guidance is appreciated. Thank you, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From slackway2me at gmail.com Thu Aug 8 17:55:22 2024 From: slackway2me at gmail.com (Alexey) Date: Thu, 8 Aug 2024 22:55:22 +0500 Subject: [OpenSIPS-Users] Variable Question In-Reply-To: References: Message-ID: Hello Alexander I'm not sure if script variables will work as you expect - it's not clear (at least for me) from the documentation [1], but I can say for sure that before assigning a value to the variable you have to 'reset' it: #reset variable $var(x) = NULL; #set variable $var(x) = 123456; Otherwise you may catch an OpenSIPS behavior when the variable sometimes has its value and sometimes does not. [1] https://www.opensips.org/Documentation/Script-CoreVar-3-5 -- best regards, Alexey https://alexeyka.zantsev.com/ From eremina.net at gmail.com Fri Aug 9 07:50:33 2024 From: eremina.net at gmail.com (Pavel Eremin) Date: Fri, 9 Aug 2024 12:50:33 +0500 Subject: [OpenSIPS-Users] Variable Question In-Reply-To: References: Message-ID: Hi, Of course var(x) will not work in that case, be sure in that. Maybe someone can correct me about that, but in case of failure route i am using $dlg_var(x) variables to get variable related to whole call. Also you can use setflag("X") in initial route and check by isflagset("X") on failure route. чт, 8 авг. 2024 г. в 19:53, Alexander Perkins : > > Hi All. If I want to have a variable set in the route block and available to the failure route, which would be the appropriate way to do so? An AVP? Or is there a better way? > > For example: > > route > { > > #do something > route(DO_INVITE) > > } > > route [DO_INVITE] > { > > #set variable > $var(x) = 123456; > t_on_failure("IT_FAILED"); > > } > > failure_route [IT_FAILED] > { > > xlog("L_INFO", "Variable X = $var(x)); > > } > > Any guidance is appreciated. > > Thank you, > Alex > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From liviu at opensips.org Fri Aug 9 14:16:00 2024 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 9 Aug 2024 17:16:00 +0300 Subject: [OpenSIPS-Users] Variable Question In-Reply-To: References: Message-ID: <25b0d046-9b24-056c-d562-6f14694730aa@opensips.org> On 08.08.2024 17:52, Alexander Perkins wrote: > If I want to have a variable set in the route block and available to > the failure route, which would be the appropriate way to do so? An > AVP?  Or is there a better way? > Hello Alexander, AVPs are still lightweight (not as lightweight as *$var*, but still fast), so storing data as *$avp* holders across one or more failure routes seems like the ideal approach. Optimization: If your data is binary ("A"/"B", True/False, etc.), see Pavel's suggestion on using the *message flags*, which will also persist across the initial and re-tried messages, as you perform serial forking using one or more *failure_route*'s. Best regards, -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From pa.ka12 at outlook.com Mon Aug 12 07:00:18 2024 From: pa.ka12 at outlook.com (Pa Ka) Date: Mon, 12 Aug 2024 07:00:18 +0000 Subject: [OpenSIPS-Users] PSTN calls fail due to rtpengine not running Message-ID: Hello, Kindly help me resolve this issue. Calls to pstn are faling (log returns server error). Verified all the possible way I could, and I finally believe the issue is caused by the failing rtpengine. Below is the rtp.log Aug 12 02:54:33.452293 sips rtpengine[199106]: INFO: [crypto] Generating new DTLS certificate Aug 12 02:54:33.452310 sips rtpengine[199106]: DEBUG: [crypto] Using EC-prime256v1 key for DTLS certificate Aug 12 02:54:33.581558 sips rtpengine[199106]: Fatal error: Failed to create nftables chains or rules: error returned from netlink for add rule (No such file or directory) Aug 12 02:54:33.581573 sips rtpengine[199106]: CRIT: [core] Fatal error: Failed to create nftables chains or rules: error returned from netlink for add rule (No such file or directory) Thank you ! -------------- next part -------------- An HTML attachment was scrubbed... URL: From pa.ka12 at outlook.com Mon Aug 12 07:03:03 2024 From: pa.ka12 at outlook.com (Pa Ka) Date: Mon, 12 Aug 2024 07:03:03 +0000 Subject: [OpenSIPS-Users] PSTN calls fail due to rtpengine not running In-Reply-To: References: Message-ID: I’m currently using mr12.3.1 ________________________________ From: Pa Ka Sent: Monday, August 12, 2024 03:00 To: Users at lists.opensips.org Subject: PSTN calls fail due to rtpengine not running Hello, Kindly help me resolve this issue. Calls to pstn are faling (log returns server error). Verified all the possible way I could, and I finally believe the issue is caused by the failing rtpengine. Below is the rtp.log Aug 12 02:54:33.452293 sips rtpengine[199106]: INFO: [crypto] Generating new DTLS certificate Aug 12 02:54:33.452310 sips rtpengine[199106]: DEBUG: [crypto] Using EC-prime256v1 key for DTLS certificate Aug 12 02:54:33.581558 sips rtpengine[199106]: Fatal error: Failed to create nftables chains or rules: error returned from netlink for add rule (No such file or directory) Aug 12 02:54:33.581573 sips rtpengine[199106]: CRIT: [core] Fatal error: Failed to create nftables chains or rules: error returned from netlink for add rule (No such file or directory) Thank you ! -------------- next part -------------- An HTML attachment was scrubbed... URL: From slackway2me at gmail.com Mon Aug 12 08:20:26 2024 From: slackway2me at gmail.com (Alexey) Date: Mon, 12 Aug 2024 13:20:26 +0500 Subject: [OpenSIPS-Users] PSTN calls fail due to rtpengine not running In-Reply-To: References: Message-ID: Hello, better to ask this question here: https://github.com/sipwise/rtpengine/issues as OpenSIPS does not handle RTP. But most likely the problem is related to nftables/iptables. I presume that your operating system uses iptables and RTPEngine 12 uses nftables instead (though I haven't worked with 12). If it is so (and if mr12 does not have any backwards compatibility with older versions and with iptables usage), you have to start using nftables in your system instead of iptables. -- best regards, Alexey https://alexeyka.zantsev.com/ From pa.ka12 at outlook.com Mon Aug 12 08:44:40 2024 From: pa.ka12 at outlook.com (Pa Ka) Date: Mon, 12 Aug 2024 08:44:40 +0000 Subject: [OpenSIPS-Users] PSTN calls fail due to rtpengine not running In-Reply-To: References: Message-ID: Thank you -----Original Message----- From: Users On Behalf Of Alexey Sent: Monday, August 12, 2024 4:20 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] PSTN calls fail due to rtpengine not running Hello, better to ask this question here: https://github.com/sipwise/rtpengine/issues as OpenSIPS does not handle RTP. But most likely the problem is related to nftables/iptables. I presume that your operating system uses iptables and RTPEngine 12 uses nftables instead (though I haven't worked with 12). If it is so (and if mr12 does not have any backwards compatibility with older versions and with iptables usage), you have to start using nftables in your system instead of iptables. -- best regards, Alexey https://alexeyka.zantsev.com/ _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From mickael at winlux.fr Tue Aug 13 13:55:45 2024 From: mickael at winlux.fr (Mickael Hubert) Date: Tue, 13 Aug 2024 15:55:45 +0200 Subject: [OpenSIPS-Users] CANCEL cross 200OK In-Reply-To: References: Message-ID: Hi Bogdan thanks for your answer. After many investigations, I found it's a network latency problem. Indeed, when I send my 200OK from UAS, this 200OK arrives after 6 (or more) seconds to UAC. So during this time slot, UAC reaches a timeout and sends a CANCEL (because it never receives 200OK). I have to find a way on my UAS to kill the call correctly. have a good day Le mar. 6 août 2024 à 14:20, Bogdan-Andrei Iancu a écrit : > Hi Mickael, > > Once the UAS generated a 200 OK reply, there is not much you can do - > you have to send this 200 to UAC. And the UAC will have to ACK it and > after that to decide if to keep the call or not - if not, it will send a > BYE on the spot. > > A proxy is not the place to deal with the typical CANCEL/200ok race in > SIP, but to let the UAC to deal with it. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 17.07.2024 12:30, Mickael Hubert wrote: > > Hi all, > > I have an issue in my OpenSIPS proxy (version: opensips 3.3.4 > > (x86_64/linux)). > > Proxy receives 200OK from UAS, but in the same time, receives CANCEL > > from UAC. > > Ex: > > ...... > > UAS --> 200OK (SDP) --> proxy > > proxy <-- CANCEL <-- UAC > > proxy --> 200 CANCELING --> UAC > > proxy --> 200OK (SDP) --> UAC > > proxy <-- ACK <-- UAC > > UAS <-- ACK <-- proxy > > ..... > > > > I want to find a solution that proxy sends 487 to UAC, and BYE to UAS. > > How can I do that please ? There is a function or I have to code all > > this scenario ? > > > > thanks in advance > > Have a good day > > > > _______________________________________________ > > 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 oualla.simohamed at gmail.com Wed Aug 14 03:46:32 2024 From: oualla.simohamed at gmail.com (Mohamed OUALLA) Date: Wed, 14 Aug 2024 04:46:32 +0100 Subject: [OpenSIPS-Users] Help with Adding Custom SIP Header to Relayed CANCEL Request in OpenSIPS Message-ID: Hello everyone, I am currently working with OpenSIPs in stateful mode, and I am encountering an issue with relaying a CANCEL request to cancel a pending initial SIP INVITE. The CANCEL request is hop-by-hop, so OpenSIPs act on it first "*send 200 canceling*" then it generates a CANCEL SIP Request and relay it to the next hop. My goal is to add a custom SIP header, specifically "*X-Reason*", to this relayed CANCEL request. Unfortunately, I am only able to add the standard "*Reason*" header. The challenge I am facing is that FreeSWITCH, which receives the relayed CANCEL request, removes the "Reason" header due to a parameter I have set to disable adding Q.850 reasons. This parameter was initially enabled to prevent FreeSWITCH from adding the "Reason" header to negative SIP responses, but it's now also affecting the CANCEL request, leading to the removal of the "Reason" header! Here’s my current OpenSIPs configuration that successfully adds the "Reason" header to the CANCEL request: ``` if(!is_present_hf("Reason")){ append_hf("Reason: Q.850;cause=32\r\n", "CSeq"); #the outgoing CANCEL request has the Reason header value now } ``` However, when I try to add headers like "*X-Reason*" or "*TestHeader*" for example, they don't seem to be included in the relayed CANCEL request: ``` if(!is_present_hf("Reason")){ append_hf("X-Reason: Q.850;cause=31\r\n", "CSeq"); # No changes are reflected in the relayed CANCEL request :-/ # or append_hf("TestHeader: Q.850;cause=31\r\n", "CSeq");# No changes are reflected in the relayed CANCEL request :-/ } ``` Has anyone faced a similar issue or have any suggestions on how I can successfully add a custom "*X-Reason*" header or any other custom " *X-Header*" to the relayed CANCEL request? Thanks in advance, Have a beautiful day -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Wed Aug 14 14:40:58 2024 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Wed, 14 Aug 2024 11:40:58 -0300 Subject: [OpenSIPS-Users] Help with Adding Custom SIP Header to Relayed CANCEL Request in OpenSIPS In-Reply-To: References: Message-ID: Where in the routing script are you adding the header? Did you try on branch route? Em qua., 14 de ago. de 2024 00:51, Mohamed OUALLA < oualla.simohamed at gmail.com> escreveu: > Hello everyone, > > I am currently working with OpenSIPs in stateful mode, and I am > encountering an issue with relaying a CANCEL request to cancel a pending > initial SIP INVITE. > > The CANCEL request is hop-by-hop, so OpenSIPs act on it first "*send > 200 canceling*" then it generates a CANCEL SIP Request and relay it to > the next hop. My goal is to add a custom SIP header, specifically " > *X-Reason*", to this relayed CANCEL request. Unfortunately, I am only > able to add the standard "*Reason*" header. > > The challenge I am facing is that FreeSWITCH, which receives the relayed > CANCEL request, removes the "Reason" header due to a parameter I have set > to disable adding Q.850 reasons. This parameter was initially enabled to > prevent FreeSWITCH from adding the "Reason" header to negative SIP > responses, but it's now also affecting the CANCEL request, leading to the > removal of the "Reason" header! > > Here’s my current OpenSIPs configuration that successfully adds the > "Reason" header to the CANCEL request: > > ``` > > if(!is_present_hf("Reason")){ > append_hf("Reason: Q.850;cause=32\r\n", "CSeq"); #the outgoing CANCEL > request has the Reason header value now > } > ``` > > However, when I try to add headers like "*X-Reason*" or "*TestHeader*" > for example, they don't seem to be included in the relayed CANCEL request: > > ``` > > if(!is_present_hf("Reason")){ > append_hf("X-Reason: Q.850;cause=31\r\n", "CSeq"); # No changes are > reflected in the relayed CANCEL request :-/ > # or > append_hf("TestHeader: Q.850;cause=31\r\n", "CSeq");# No changes are > reflected in the relayed CANCEL request :-/ > } > ``` > > Has anyone faced a similar issue or have any suggestions on how I can > successfully add a custom "*X-Reason*" header or any other custom " > *X-Header*" to the relayed CANCEL request? > > Thanks in advance, > > Have a beautiful day > > _______________________________________________ > 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 Aug 14 14:58:10 2024 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Aug 2024 17:58:10 +0300 Subject: [OpenSIPS-Users] Help with Adding Custom SIP Header to Relayed CANCEL Request in OpenSIPS In-Reply-To: References: Message-ID: Hi, Take a look at this 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 https://www.siphub.com On 14.08.2024 06:46, Mohamed OUALLA wrote: > > Hello everyone, > >   I am currently working with OpenSIPs in stateful mode, and I am > encountering an issue with relaying a CANCEL request to cancel a > pending initial SIP INVITE. > >   The CANCEL request is hop-by-hop, so OpenSIPs act on it first > "*/send 200 canceling/*" then it generates a CANCEL SIP Request and > relay it to the next hop. My goal is to add a custom SIP header, > specifically "*X-Reason*", to this relayed CANCEL request. > Unfortunately, I am only able to add the standard "*Reason*" header. > >   The challenge I am facing is that FreeSWITCH, which receives the > relayed CANCEL request, removes the "Reason" header due to a parameter > I have set to disable adding Q.850 reasons. This parameter was > initially enabled to prevent FreeSWITCH from adding the "Reason" > header to negative SIP responses, but it's now also affecting the > CANCEL request, leading to the removal of the "Reason" header! > > Here’s my current OpenSIPs configuration that successfully adds the > "Reason" header to the CANCEL request: > > ``` > > if(!is_present_hf("Reason")){ >     append_hf("Reason: Q.850;cause=32\r\n", "CSeq"); #the outgoing > CANCEL request has the Reason header value now > } > > ``` > >   However, when I try to add headers like "*X-Reason*" or > "*TestHeader*" for example, they don't seem to be included in the > relayed CANCEL request: > > ``` > > if(!is_present_hf("Reason")){ >     append_hf("X-Reason: Q.850;cause=31\r\n", "CSeq"); # No changes > are reflected in the relayed CANCEL request :-/ >     # or >     append_hf("TestHeader: Q.850;cause=31\r\n", "CSeq");# No changes > are reflected in the relayed CANCEL request :-/ > } > > ``` > >   Has anyone faced a similar issue or have any suggestions on how I > can successfully add a custom "*X-Reason*" header or any other custom > "*/X-Header/*" to the relayed CANCEL request? > > Thanks in advance, > > Have a beautiful day > > > _______________________________________________ > 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 ahmedmunir007 at gmail.com Wed Aug 14 15:02:48 2024 From: ahmedmunir007 at gmail.com (Ahmed Chohan) Date: Wed, 14 Aug 2024 11:02:48 -0400 Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in OpenSIPs 3.4 Message-ID: Hi, After installing the dependencies(including radcli and freeradius) on RHEL 9, I've installed OpenSIPs 3.4.X from the epel repository and related packages including opensips-radius-modules. When loading the aaa_radius.so module in opensips script, I get the following error message: module 'aaa_radius.so' not found in '/usr/lib64/opensips/modules/'. I inspected the opensips modules folder, 'aaa_radius.so' not found. Please let me know what other module(s) I may need to install to make it work. -- Regards, Ahmed Munir Chohan -------------- next part -------------- An HTML attachment was scrubbed... URL: From oualla.simohamed at gmail.com Thu Aug 15 16:50:04 2024 From: oualla.simohamed at gmail.com (Mohamed OUALLA) Date: Thu, 15 Aug 2024 17:50:04 +0100 Subject: [OpenSIPS-Users] Help with Adding Custom SIP Header to Relayed CANCEL Request in OpenSIPS In-Reply-To: References: Message-ID: Hi Bogdan-Andrei, This indeed has solved my problem! Here is my new code: ``` # CANCEL processing if ( is_method("CANCEL") ) { if ( t_check_trans() ){ t_add_cancel_reason("X-Reason: Q.850;cause=31\r\n"); t_relay(); } exit; } ``` Thank you very much. Best regards, On Wed, Aug 14, 2024 at 3:58 PM Bogdan-Andrei Iancu wrote: > Hi, > > Take a look at this > 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 > https://www.siphub.com > > On 14.08.2024 06:46, Mohamed OUALLA wrote: > > Hello everyone, > > I am currently working with OpenSIPs in stateful mode, and I am > encountering an issue with relaying a CANCEL request to cancel a pending > initial SIP INVITE. > > The CANCEL request is hop-by-hop, so OpenSIPs act on it first "*send > 200 canceling*" then it generates a CANCEL SIP Request and relay it to > the next hop. My goal is to add a custom SIP header, specifically " > *X-Reason*", to this relayed CANCEL request. Unfortunately, I am only > able to add the standard "*Reason*" header. > > The challenge I am facing is that FreeSWITCH, which receives the relayed > CANCEL request, removes the "Reason" header due to a parameter I have set > to disable adding Q.850 reasons. This parameter was initially enabled to > prevent FreeSWITCH from adding the "Reason" header to negative SIP > responses, but it's now also affecting the CANCEL request, leading to the > removal of the "Reason" header! > > Here’s my current OpenSIPs configuration that successfully adds the > "Reason" header to the CANCEL request: > > ``` > > if(!is_present_hf("Reason")){ > append_hf("Reason: Q.850;cause=32\r\n", "CSeq"); #the outgoing CANCEL > request has the Reason header value now > } > ``` > > However, when I try to add headers like "*X-Reason*" or "*TestHeader*" > for example, they don't seem to be included in the relayed CANCEL request: > > ``` > > if(!is_present_hf("Reason")){ > append_hf("X-Reason: Q.850;cause=31\r\n", "CSeq"); # No changes are > reflected in the relayed CANCEL request :-/ > # or > append_hf("TestHeader: Q.850;cause=31\r\n", "CSeq");# No changes are > reflected in the relayed CANCEL request :-/ > } > ``` > > Has anyone faced a similar issue or have any suggestions on how I can > successfully add a custom "*X-Reason*" header or any other custom " > *X-Header*" to the relayed CANCEL request? > > Thanks in advance, > > Have a beautiful day > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- ============================== Mohamed OUALLA *VoIP Technical Solutions and Software Engineer* Mail: oualla.simohamed at gmail.com N.Phone: +212 6 29 19 3116 *SSC Certified Professional* ============================== -------------- next part -------------- An HTML attachment was scrubbed... URL: From ahmedmunir007 at gmail.com Fri Aug 16 14:54:46 2024 From: ahmedmunir007 at gmail.com (Ahmed Chohan) Date: Fri, 16 Aug 2024 10:54:46 -0400 Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in OpenSIPs 3.4 In-Reply-To: References: Message-ID: Hi Guys, Following up on this. > Message: 3 > Date: Wed, 14 Aug 2024 11:02:48 -0400 > From: Ahmed Chohan > To: OpenSIPs Users > Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in > OpenSIPs 3.4 > Message-ID: > Jc4643HrB60FNHPpC25vNEvTNdkU1eGzxqcg0Op9a1JAA at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > After installing the dependencies(including radcli and freeradius) on RHEL > 9, I've installed OpenSIPs 3.4.X from the epel repository and related > packages including opensips-radius-modules. > > When loading the aaa_radius.so module in opensips script, I get the > following error message: module 'aaa_radius.so' not found in > '/usr/lib64/opensips/modules/'. > > I inspected the opensips modules folder, 'aaa_radius.so' not found. Please > let me know what other module(s) I may need to install to make it work. > > > > -- > Regards, > > Ahmed Munir Chohan > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.opensips.org/pipermail/users/attachments/20240814/9fa4da3d/attachment-0001.html > > > > -- -- Regards, Ahmed Munir Chohan -------------- next part -------------- An HTML attachment was scrubbed... URL: From venefax at gmail.com Fri Aug 16 16:45:55 2024 From: venefax at gmail.com (Saint Michael) Date: Fri, 16 Aug 2024 12:45:55 -0400 Subject: [OpenSIPS-Users] changing port on second leg Message-ID: Using opensips 3.4, I need to change the destination port for an outbound call. The call comes in at port 56000, where I have a socket, and I need to send the second leg to change the destination port to 5060. Is there a way to to this? From pimenta at inatel.br Fri Aug 16 20:20:59 2024 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 16 Aug 2024 20:20:59 +0000 Subject: [OpenSIPS-Users] How to replace a port in SDP text? Message-ID: Hi. I'm using OpenSIPs 2.3 How can I change a port value in SDP text (in INVITE or SIP OK) ? I was reading about Script Transformations and module textops, but I can't figure out how to do that. Any hint will be very helpful! I mean the port in the line like this: m=audio 50270 RTP/AVP 106 9 0 8 3 111 102 110 112 98 101 100 99 I just want to test and see what will happen (with RTP packages) if I change such port with the source port. Best regards. [cid:a604bf77-b5d0-4cba-b35b-b5456300e731] Rodrigo Pimenta Carvalho Inatel Competence Center - PDI www.inatel.br [cid:ac707c5f-7ca9-4d68-902f-af55d082397a] [cid:6464666b-6226-4316-a488-a61e14ff4584] [cid:9e1aa20a-9909-4469-b056-bf2a099cc8cb] [cid:5d87a353-7e49-4295-909b-e3cc7cb956e5] -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Outlook-cid_a604bf.png Type: image/png Size: 5592 bytes Desc: Outlook-cid_a604bf.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Outlook-ehe3sv15.png Type: image/png Size: 675 bytes Desc: Outlook-ehe3sv15.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Outlook-cjaof3xp.png Type: image/png Size: 832 bytes Desc: Outlook-cjaof3xp.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Outlook-cid_9e1aa2.png Type: image/png Size: 2252 bytes Desc: Outlook-cid_9e1aa2.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Outlook-w5a25r52.png Type: image/png Size: 957 bytes Desc: Outlook-w5a25r52.png URL: From oualla.simohamed at gmail.com Fri Aug 16 20:48:13 2024 From: oualla.simohamed at gmail.com (Mohamed OUALLA) Date: Fri, 16 Aug 2024 21:48:13 +0100 Subject: [OpenSIPS-Users] How to replace a port in SDP text? In-Reply-To: References: Message-ID: Hi Rodrigo, I hope this help: ``` #Change sdp port of m line to 12345 if (has_body("application/sdp")) { subst_body("/^m=audio [0-9]+ RTP\/AVP/m=audio 12345 RTP\/AVP/"); } ``` Best, Mohamed On Fri, Aug 16, 2024 at 9:24 PM Rodrigo Pimenta Carvalho wrote: > Hi. > > I'm using OpenSIPs 2.3 > > How can I change a port value in SDP text (in INVITE or SIP OK) ? I was > reading about Script Transformations and module textops, but I can't > figure out how to do that. > > Any hint will be very helpful! > > I mean the port in the line like this: m=audio* 50270 *RTP/AVP 106 9 0 8 > 3 111 102 110 112 98 101 100 99 > I just want to test and see what will happen (with RTP packages) if I > change such port with the source port. > > Best regards. > > [image: cid:a604bf77-b5d0-4cba-b35b-b5456300e731] > > *Rodrigo Pimenta Carvalho* > > Inatel Competence Center - PDI > > > > *www.inatel.br * > > > > > > > > [image: cid:9e1aa20a-9909-4469-b056-bf2a099cc8cb] > > > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- ============================== Mohamed OUALLA *VoIP Technical Solutions and Software Engineer* Mail: oualla.simohamed at gmail.com N.Phone: +212 6 29 19 3116 *SSC Certified Professional* ============================== -------------- next part -------------- An HTML attachment was scrubbed... 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Name: Outlook-w5a25r52.png Type: image/png Size: 957 bytes Desc: not available URL: From oualla.simohamed at gmail.com Sat Aug 17 12:24:34 2024 From: oualla.simohamed at gmail.com (Mohamed OUALLA) Date: Sat, 17 Aug 2024 13:24:34 +0100 Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in OpenSIPs 3.4 In-Reply-To: References: Message-ID: Hi, Could you try to load the module indicating the full path and tets? loadmodule “/usr/lib64/opensips/modules/aaa_radius.so” If this works, else indicate the mpath parameter before loading any module like this: mpath=“ usr/lib64/opensips/modules/” Best On Fri, 16 Aug 2024 at 15:58 Ahmed Chohan wrote: > Hi Guys, > > Following up on this. > > > >> Message: 3 >> Date: Wed, 14 Aug 2024 11:02:48 -0400 >> From: Ahmed Chohan >> To: OpenSIPs Users >> Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in >> OpenSIPs 3.4 >> Message-ID: >> > Jc4643HrB60FNHPpC25vNEvTNdkU1eGzxqcg0Op9a1JAA at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Hi, >> >> After installing the dependencies(including radcli and freeradius) on RHEL >> 9, I've installed OpenSIPs 3.4.X from the epel repository and related >> packages including opensips-radius-modules. >> >> When loading the aaa_radius.so module in opensips script, I get the >> following error message: module 'aaa_radius.so' not found in >> '/usr/lib64/opensips/modules/'. >> >> I inspected the opensips modules folder, 'aaa_radius.so' not found. Please >> let me know what other module(s) I may need to install to make it work. >> >> >> >> -- >> Regards, >> >> Ahmed Munir Chohan >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: < >> http://lists.opensips.org/pipermail/users/attachments/20240814/9fa4da3d/attachment-0001.html >> > > > >> >> -- > > -- > Regards, > > Ahmed Munir Chohan > > _______________________________________________ > 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 oualla.simohamed at gmail.com Sat Aug 17 12:26:59 2024 From: oualla.simohamed at gmail.com (Mohamed OUALLA) Date: Sat, 17 Aug 2024 13:26:59 +0100 Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in OpenSIPs 3.4 In-Reply-To: References: Message-ID: I correct the value of mpath from my previous message: mpath=“ /usr/lib64/opensips/modules/” I hope this helps, Best On Sat, 17 Aug 2024 at 13:24 Mohamed OUALLA wrote: > Hi, > > Could you try to load the module indicating the full path and tets? > loadmodule “/usr/lib64/opensips/modules/aaa_radius.so” > If this works, else indicate the mpath parameter before loading any > module like this: > mpath=“ > usr/lib64/opensips/modules/” > > Best > > > > On Fri, 16 Aug 2024 at 15:58 Ahmed Chohan wrote: > >> Hi Guys, >> >> Following up on this. >> >> >> >>> Message: 3 >>> Date: Wed, 14 Aug 2024 11:02:48 -0400 >>> From: Ahmed Chohan >>> To: OpenSIPs Users >>> Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in >>> OpenSIPs 3.4 >>> Message-ID: >>> >> Jc4643HrB60FNHPpC25vNEvTNdkU1eGzxqcg0Op9a1JAA at mail.gmail.com> >>> Content-Type: text/plain; charset="utf-8" >>> >>> Hi, >>> >>> After installing the dependencies(including radcli and freeradius) on >>> RHEL >>> 9, I've installed OpenSIPs 3.4.X from the epel repository and related >>> packages including opensips-radius-modules. >>> >>> When loading the aaa_radius.so module in opensips script, I get the >>> following error message: module 'aaa_radius.so' not found in >>> '/usr/lib64/opensips/modules/'. >>> >>> I inspected the opensips modules folder, 'aaa_radius.so' not found. >>> Please >>> let me know what other module(s) I may need to install to make it work. >>> >>> >>> >>> -- >>> Regards, >>> >>> Ahmed Munir Chohan >>> -------------- next part -------------- >>> An HTML attachment was scrubbed... >>> URL: < >>> http://lists.opensips.org/pipermail/users/attachments/20240814/9fa4da3d/attachment-0001.html >>> > >> >> >>> >>> -- >> >> -- >> Regards, >> >> Ahmed Munir Chohan >> >> _______________________________________________ >> 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 oualla.simohamed at gmail.com Sat Aug 17 13:13:30 2024 From: oualla.simohamed at gmail.com (Mohamed OUALLA) Date: Sat, 17 Aug 2024 14:13:30 +0100 Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in OpenSIPs 3.4 In-Reply-To: References: Message-ID: Hi, I apologize for the confusion in my previous messages. I missed the part when you couldn't find the module in the path of installed modules of openSIPs. Since the *aaa_radius.so* module is not found in your /usr/lib64/opensips/modules/ directory, it seems the module may not be installed. You may want to take a look here opensips-aaa_radius-3.4.3-3.fc40.x86_64.rpm . Again, I apologize for the mistake, Best, On Sat, Aug 17, 2024 at 1:26 PM Mohamed OUALLA wrote: > I correct the value of mpath from my previous message: > mpath=“ > /usr/lib64/opensips/modules/” > > I hope this helps, > Best > > > On Sat, 17 Aug 2024 at 13:24 Mohamed OUALLA > wrote: > >> Hi, >> >> Could you try to load the module indicating the full path and tets? >> loadmodule “/usr/lib64/opensips/modules/aaa_radius.so” >> If this works, else indicate the mpath parameter before loading any >> module like this: >> mpath=“ >> usr/lib64/opensips/modules/” >> >> Best >> >> >> >> On Fri, 16 Aug 2024 at 15:58 Ahmed Chohan >> wrote: >> >>> Hi Guys, >>> >>> Following up on this. >>> >>> >>> >>>> Message: 3 >>>> Date: Wed, 14 Aug 2024 11:02:48 -0400 >>>> From: Ahmed Chohan >>>> To: OpenSIPs Users >>>> Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in >>>> OpenSIPs 3.4 >>>> Message-ID: >>>> >>> Jc4643HrB60FNHPpC25vNEvTNdkU1eGzxqcg0Op9a1JAA at mail.gmail.com> >>>> Content-Type: text/plain; charset="utf-8" >>>> >>>> Hi, >>>> >>>> After installing the dependencies(including radcli and freeradius) on >>>> RHEL >>>> 9, I've installed OpenSIPs 3.4.X from the epel repository and related >>>> packages including opensips-radius-modules. >>>> >>>> When loading the aaa_radius.so module in opensips script, I get the >>>> following error message: module 'aaa_radius.so' not found in >>>> '/usr/lib64/opensips/modules/'. >>>> >>>> I inspected the opensips modules folder, 'aaa_radius.so' not found. >>>> Please >>>> let me know what other module(s) I may need to install to make it work. >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> Ahmed Munir Chohan >>>> -------------- next part -------------- >>>> An HTML attachment was scrubbed... >>>> URL: < >>>> http://lists.opensips.org/pipermail/users/attachments/20240814/9fa4da3d/attachment-0001.html >>>> > >>> >>> >>>> >>>> -- >>> >>> -- >>> Regards, >>> >>> Ahmed Munir Chohan >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> -- -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Aug 19 14:15:19 2024 From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=) Date: Mon, 19 Aug 2024 17:15:19 +0300 Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in OpenSIPs 3.4 In-Reply-To: References: Message-ID: <852638de-1b94-4c2b-b0ff-d216bc095009@opensips.org> Hi, Ahmed! Apparently aaa_radius is not built for RedHat > 7 and/or Fedora > 23 due to the lack of radiusclient-ng-devel package. I will try to push the radcli-devel package instead, see if it can be found on all supported branches. Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 8/14/24 6:02 PM, Ahmed Chohan wrote: > Hi, > > After installing the dependencies(including radcli and freeradius) on RHEL > 9, I've installed OpenSIPs 3.4.X from the epel repository and related > packages including opensips-radius-modules. > > When loading the aaa_radius.so module in opensips script, I get the > following error message: module 'aaa_radius.so' not found in > '/usr/lib64/opensips/modules/'. > > I inspected the opensips modules folder, 'aaa_radius.so' not found. Please > let me know what other module(s) I may need to install to make it work. > > > > > Hi, > > After installing the dependencies(including radcli and freeradius) on > RHEL 9, I've installed OpenSIPs 3.4.X from the epel repository and > related packages including opensips-radius-modules. > > When loading the aaa_radius.so module in opensips script, I get the > following error message: module 'aaa_radius.so' not found in > '/usr/lib64/opensips/modules/'. > > I inspected the opensips modules folder, 'aaa_radius.so' not found. > Please let me know what other module(s) I may need to install to make it > work. > > > > -- > Regards, > > Ahmed Munir Chohan > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Mon Aug 19 14:16:41 2024 From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=) Date: Mon, 19 Aug 2024 17:16:41 +0300 Subject: [OpenSIPS-Users] changing port on second leg In-Reply-To: References: Message-ID: Yes, simply set the destination port ($dp) to 5060. Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 8/16/24 7:45 PM, Saint Michael wrote: > Using opensips 3.4, I need to change the destination port for an > outbound call. The call comes in at port 56000, where I have a socket, > and I need to send the second leg to change the destination port to > 5060. > Is there a way to to this? > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From ahmedmunir007 at gmail.com Mon Aug 19 15:53:21 2024 From: ahmedmunir007 at gmail.com (Ahmed Chohan) Date: Mon, 19 Aug 2024 11:53:21 -0400 Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in OpenSIPs 3.4 In-Reply-To: References: Message-ID: Hi Mohamed, Thanks for your email. As a workaround, I downloaded the source file for OpenSIPs 3.4, compiled it on aaa_radius module, and placed aaa_radius.so under the directory: /usr/lib64/opensips/modules. Now it is working. Thanks. -- Regards > Date: Sat, 17 Aug 2024 14:13:30 +0100 > From: Mohamed OUALLA > To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Unable to locate aaa_radius module in > OpenSIPs 3.4 > Message-ID: > < > CAGmsPDtfynY3UYawVV-+g4GedCZ8f8FFdAMBZTVtFOPVNCs1ng at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > I apologize for the confusion in my previous messages. I missed the part > when you couldn't find the module in the path of installed modules of > openSIPs. Since the *aaa_radius.so* module is not found in your > /usr/lib64/opensips/modules/ directory, it seems the module may not be > installed. > > You may want to take a look here > opensips-aaa_radius-3.4.3-3.fc40.x86_64.rpm > < > https://fedora.pkgs.org/40/fedora-x86_64/opensips-aaa_radius-3.4.3-3.fc40.x86_64.rpm.html > > > . > > Again, I apologize for the mistake, > > Best, > -- Regards, Ahmed Munir Chohan -------------- next part -------------- An HTML attachment was scrubbed... URL: From venefax at gmail.com Mon Aug 19 17:15:18 2024 From: venefax at gmail.com (Saint Michael) Date: Mon, 19 Aug 2024 13:15:18 -0400 Subject: [OpenSIPS-Users] changing port on second leg In-Reply-To: References: Message-ID: It does not work at all. $dp = 5060; On Mon, Aug 19, 2024 at 10:19 AM Răzvan Crainea wrote: > > Yes, simply set the destination port ($dp) to 5060. > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer / SIPhub CTO > http://www.opensips-solutions.com / https://www.siphub.com > > On 8/16/24 7:45 PM, Saint Michael wrote: > > Using opensips 3.4, I need to change the destination port for an > > outbound call. The call comes in at port 56000, where I have a socket, > > and I need to send the second leg to change the destination port to > > 5060. > > Is there a way to to this? > > > > _______________________________________________ > > 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 daniel.zanutti at gmail.com Mon Aug 19 19:41:45 2024 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Mon, 19 Aug 2024 16:41:45 -0300 Subject: [OpenSIPS-Users] Delay relay of 200 OK Message-ID: Hi We need to delay the delivery of the 200 OK, and send it a few seconds later, after an event is generated internally. If the call is marked as suspicious, we will hangup the call and send 503 to the origin and BYE to destination. Is it possible to do it? Can someone send some ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: From mayamatakeshi at gmail.com Mon Aug 19 23:37:21 2024 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Tue, 20 Aug 2024 08:37:21 +0900 Subject: [OpenSIPS-Users] Delay relay of 200 OK In-Reply-To: References: Message-ID: Hi, I never it but you might want to look at: https://www.opensips.org/Documentation/Script-Async-3-5 On Tue, Aug 20, 2024 at 4:45 AM Daniel Zanutti wrote: > Hi > > We need to delay the delivery of the 200 OK, and send it a few seconds > later, after an event is generated internally. If the call is marked as > suspicious, we will hangup the call and send 503 to the origin and BYE to > destination. > > Is it possible to do it? Can someone send some ideas? > > _______________________________________________ > 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 eremina.net at gmail.com Tue Aug 20 06:46:18 2024 From: eremina.net at gmail.com (Pavel Eremin) Date: Tue, 20 Aug 2024 11:46:18 +0500 Subject: [OpenSIPS-Users] changing port on second leg In-Reply-To: References: Message-ID: just guess... I suppose you may want to send second leg from 5060 port of your server? пт, 16 авг. 2024 г. в 21:48, Saint Michael : > > Using opensips 3.4, I need to change the destination port for an > outbound call. The call comes in at port 56000, where I have a socket, > and I need to send the second leg to change the destination port to > 5060. > Is there a way to to this? > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From venefax at gmail.com Tue Aug 20 10:37:43 2024 From: venefax at gmail.com (Saint Michael) Date: Tue, 20 Aug 2024 06:37:43 -0400 Subject: [OpenSIPS-Users] changing port on second leg In-Reply-To: References: Message-ID: The "from" port is not relevant. I need to manipulate the remote port, the tsrget. On Tue, Aug 20, 2024, 2:50 AM Pavel Eremin wrote: > just guess... I suppose you may want to send second leg from 5060 port > of your server? > > пт, 16 авг. 2024 г. в 21:48, Saint Michael : > > > > Using opensips 3.4, I need to change the destination port for an > > outbound call. The call comes in at port 56000, where I have a socket, > > and I need to send the second leg to change the destination port to > > 5060. > > Is there a way to to this? > > > > _______________________________________________ > > 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 slackway2me at gmail.com Tue Aug 20 10:54:05 2024 From: slackway2me at gmail.com (Alexey) Date: Tue, 20 Aug 2024 15:54:05 +0500 Subject: [OpenSIPS-Users] changing port on second leg In-Reply-To: References: Message-ID: What about setport [1] core function? [1] https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#setport -- best regards, Alexey https://alexeyka.zantsev.com/ From liviu at opensips.org Wed Aug 21 10:09:46 2024 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 21 Aug 2024 13:09:46 +0300 Subject: [OpenSIPS-Users] [Minor Release] OpenSIPS 3.5.1 and 3.4.8 Minor Releases Message-ID: <2396c2e0-e03b-6b86-9ba2-af376de2ee93@opensips.org> Hi, everyone! A new round of stable minor releases is now out: *3.5.1 *and *3.4.8*.  These versions include a series of important fixes done in the past two months. Do make sure to schedule an update as soon as possible -- it is highly recommended. Full changelogs: https://opensips.org/pub/opensips/3.5.1/ChangeLog https://opensips.org/pub/opensips/3.4.8/ChangeLog Please enjoy! OpenSIPS Team -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Aug 22 14:42:52 2024 From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=) Date: Thu, 22 Aug 2024 17:42:52 +0300 Subject: [OpenSIPS-Users] Unable to locate aaa_radius module in OpenSIPs 3.4 In-Reply-To: References: Message-ID: <1692aac6-d8bb-4209-841d-9d2ecd40a426@opensips.org> Hi, Ahmed! The module should be in the opensips-radius-module package now in OpenSIPS 3.4.8. ``` [root at repos x86_64]# rpm -ql opensips-radius-modules-3.4.8-1.el9.x86_64.rpm | grep aaa_radius warning: opensips-radius-modules-3.4.8-1.el9.x86_64.rpm: Header V4 RSA/SHA256 Signature, key ID 049ad65b: NOKEY /usr/lib64/opensips/modules/aaa_radius.so /usr/share/doc/opensips-radius-modules/README.aaa_radius ``` Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 8/19/24 6:53 PM, Ahmed Chohan wrote: > Hi Mohamed, > > Thanks for your email. As a workaround, I downloaded the source file for > OpenSIPs 3.4, compiled it on aaa_radius module, and placed aaa_radius.so > under the directory: /usr/lib64/opensips/modules. > > Now it is working. Thanks. > > -- > Regards > > >> Date: Sat, 17 Aug 2024 14:13:30 +0100 >> From: Mohamed OUALLA >> To: OpenSIPS users mailling list >> Subject: Re: [OpenSIPS-Users] Unable to locate aaa_radius module in >> OpenSIPs 3.4 >> Message-ID: >> < >> CAGmsPDtfynY3UYawVV-+g4GedCZ8f8FFdAMBZTVtFOPVNCs1ng at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Hi, >> >> I apologize for the confusion in my previous messages. I missed the part >> when you couldn't find the module in the path of installed modules of >> openSIPs. Since the *aaa_radius.so* module is not found in your >> /usr/lib64/opensips/modules/ directory, it seems the module may not be >> installed. >> >> You may want to take a look here >> opensips-aaa_radius-3.4.3-3.fc40.x86_64.rpm >> < >> https://fedora.pkgs.org/40/fedora-x86_64/opensips-aaa_radius-3.4.3-3.fc40.x86_64.rpm.html >>> >> . >> >> Again, I apologize for the mistake, >> >> Best, >> >> >> Hi Mohamed, >> >> Thanks for your email. As a workaround, I downloaded the source file >> for OpenSIPs 3.4, compiled it on aaa_radius module, and placed >> aaa_radius.so under the directory: /usr/lib64/opensips/modules. >> >> Now it is working. Thanks. >> >> -- >> Regards >> >> Date: Sat, 17 Aug 2024 14:13:30 +0100 >> From: Mohamed OUALLA > > >> To: OpenSIPS users mailling list > > >> Subject: Re: [OpenSIPS-Users] Unable to locate aaa_radius module in >>         OpenSIPs        3.4 >> Message-ID: >> >> > >> Content-Type: text/plain; charset="utf-8" >> >> Hi, >> >>   I apologize for the confusion in my previous messages. I missed >> the part >> when you couldn't find the module in the path of installed modules of >> openSIPs. Since the *aaa_radius.so* module is not found in your >> /usr/lib64/opensips/modules/ directory, it seems the module may not be >> installed. >> >>   You may want to take a look here >> opensips-aaa_radius-3.4.3-3.fc40.x86_64.rpm >> > >> . >> >> Again, I apologize for the mistake, >> >> Best, >> >> -- >> Regards, >> >> Ahmed Munir Chohan >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From voransoy at gmail.com Sat Aug 24 11:06:04 2024 From: voransoy at gmail.com (Volkan Oransoy) Date: Sat, 24 Aug 2024 12:06:04 +0100 Subject: [OpenSIPS-Users] TCP keepalive problem. Message-ID: Hi all, I have an interesting issue that I am trying to identify the cause of. I am trying to migrate a quite old Kamailio SIP proxy to OpenSIPS 3.4. The new OpenSIPS proxy uses mid_registrar and replicates registrations to the backend boxes. A specific type of UACs (Gigaset AS690) does not keep the TCP connection alive. So on every registration refresh, the traffic comes from another port and a new contact is created. Interestingly, there are UACs from the very same network that have a healthy connection to the OpenSIPS box. Since this is not an issue with the old box, I assume the issue is related to my new setup. When sniffing the traffic, I see that the UAC ends the connection. 11 9.378043 UAC_IP SERVER_IP TCP 60 22719 → 5060 [SYN] Seq=0 Win=1608 Len=0 MSS=536 12 9.378148 SERVER_IP UAC_IP TCP 58 5060 → 22719 [SYN, ACK] Seq=0 Ack=1 Win=64240 Len=0 MSS=1460 13 9.401441 UAC_IP SERVER_IP TCP 60 22719 → 5060 [ACK] Seq=1 Ack=1 Win=1608 Len=0 14 9.405625 UAC_IP SERVER_IP SIP 560 Request: REGISTER sip:c384.bulutfon.net (1 binding) | 15 9.405731 SERVER_IP UAC_IP TCP 54 5060 → 22719 [ACK] Seq=1 Ack=507 Win=63784 Len=0 16 9.405863 SERVER_IP UAC_IP SIP 557 Status: 401 Unauthorized | 17 9.434851 UAC_IP SERVER_IP TCP 60 22719 → 5060 [ACK] Seq=507 Ack=504 Win=1608 Len=0 18 9.464046 UAC_IP SERVER_IP SIP 756 Request: REGISTER sip:c384.bulutfon.net (1 binding) | 19 9.470312 SERVER_IP UAC_IP TCP 590 5060 → 22719 [ACK] Seq=504 Ack=1209 Win=63784 Len=536 [TCP segment of a reassembled PDU] 20 9.470337 SERVER_IP UAC_IP SIP 125 Status: 200 OK (REGISTER) (2 bindings) | 21 9.495625 UAC_IP SERVER_IP TCP 60 22719 → 5060 [ACK] Seq=1209 Ack=1040 Win=1608 Len=0 22 9.501248 UAC_IP SERVER_IP TCP 60 22719 → 5060 [ACK] Seq=1209 Ack=1111 Win=1608 Len=0 23 10.520457 UAC_IP SERVER_IP TCP 60 22719 → 5060 [FIN, ACK] Seq=1209 Ack=1111 Win=1608 Len=0 24 10.520701 SERVER_IP UAC_IP TCP 54 5060 → 22719 [FIN, ACK] Seq=1111 Ack=1210 Win=63784 Len=0 25 10.543913 UAC_IP SERVER_IP TCP 60 22719 → 5060 [ACK] Seq=1210 Ack=1112 Win=1608 Len=0 I use tcp_persistent_flag to flag the TCP connections and I can see the relevant flag on ul_dump output. modparam("mid_registrar", "tcp_persistent_flag", "TCP_PERSISTENT") if ($socket_in(proto) == "tcp" || $socket_in(proto) == "tls") setbflag("TCP_PERSISTENT"); mid_registrar_save("location"); I have tried disabling the config above and giving a fixed TCP lifetime value to the OpenSIPS box, but the issue remains the same. Do you have any hints that I can chase after? Best -- Volkan Oransoy -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Thu Aug 29 07:22:43 2024 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 29 Aug 2024 12:52:43 +0530 Subject: [OpenSIPS-Users] Need some clarification on TLS configuration on opensips 3.2 Message-ID: Hi All , I am using opensips 3.2 from very long time . For TLS connection I was using our domain specific certificate and private key which was authorized by some verified organization . With that my TLS connection with the server is getting established and also I am able to get REGISTER and INVITE request on the connection . Rather than this , when I build opensips with TLS=1 opensips itself creates its own rootCA . If I am using those crt and private key file for TLS connection the connection get established but I am not getting any request . What can be the reason . My configuration is like below . modparam("tls_mgm", "server_domain", "dom3") modparam("tls_mgm", "match_ip_address", "[dom3]20.1.x.y:5061") modparam("tls_mgm", "match_sip_domain", "[dom3]none") # 20.1.x.y this is my servers private IP on which I have configured TLS socket . modparam("tls_mgm", "tls_method", "[dom3]-TLSv1_2") modparam("tls_mgm", "certificate", "[dom3]/etc/opensips/tls/rootCA/cacert.pem") modparam("tls_mgm", "private_key", "[dom3]/etc/opensips/tls/rootCA/private/cakey.pem") modparam("tls_mgm", "ca_list", "[dom3]/etc/opensips/tls/rootCA/certs/01.pem") modparam("tls_mgm", "require_cert", "[dom3]0") modparam("tls_mgm", "verify_cert", "[dom3]1") In the logs I am getting below message *2024-08-29T07:14:59.213460+00:00 ip-20-1-205-63 /sbin/opensips[22895]: INFO:tls_openssl:openssl_tls_accept: New TLS connection from x.x.x.x:20219 accepted2024-08-29T07:14:59.213866+00:00 ip-20-1-205-63 /sbin/opensips[22895]: INFO:tls_openssl:openssl_tls_accept: Client did not present a TLS certificate2024-08-29T07:14:59.214064+00:00 ip-20-1-205-63 /sbin/opensips[22895]: INFO:tls_openssl:tls_dump_cert_info: tls_accept: local TLS server certificate subject: /CN=OpenSIPS/ST=opensips.org/C=IP/emailAddress=team at opensips.org/O=opensips.org , issuer: /CN=OpenSIPS/ST=opensips.org/C=IP/emailAddress=team at opensips.org/O=opensips.org * I have added siptrace and tracing to the DB as well . I am not getting any SIP messages on the 2nd case . What can be the reason for this ? This is quite critical to me . Please do help. *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien at pwlk.fr Thu Aug 29 15:25:59 2024 From: julien at pwlk.fr (Julien Pawlak) Date: Thu, 29 Aug 2024 17:25:59 +0200 Subject: [OpenSIPS-Users] RTP Relay issue Message-ID: <72632a7847a2864754bd7a847ad73834@pwlk.fr> Hello I have a problem with RTP Relay module in the initial invite request. I have 2 rtpengine servers with both 2 interfaces, internal and external. When I add the lien below, there is no problem. Packets come and leave through good interfaces. $rtp_relay = "out-iface=internal in-iface=external" But, when I execute the mi command to switch rtpengine server, packets don't come and leave through good interfaces. I see that I have to put this lines : $rtp_relay(iface) = "external"; $rtp_relay_peer(iface) = "internal"; But, that doesn't work too, I have this in debug mode : août 29 17:04:44 opensips-1 /usr/local/sbin/opensips[844747]: DBG:rtp_relay:rtp_relay_offer: callid=[] ftag=[] ttag=[] type=[] in-iface=[internal] out-iface=[] ctx-flags=[] flags=[] peer-flags=[] août 29 17:04:44 opensips-1 /usr/local/sbin/opensips[844747]: ERROR:rtpengine:parse_flags: in-iface value without out-iface août 29 17:04:44 opensips-1 /usr/local/sbin/opensips[844747]: ERROR:rtpengine:rtpe_function_call: could not parse flags août 29 17:04:44 opensips-1 /usr/local/sbin/opensips[844747]: ERROR:rtp_relay:rtp_relay_offer: could not engage offer! Please help. Thank you -- J. -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Aug 30 11:57:58 2024 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 30 Aug 2024 17:27:58 +0530 Subject: [OpenSIPS-Users] Need some clarification on TLS configuration on opensips 3.2 In-Reply-To: References: Message-ID: Hi , for outbound call to a tls gateway I have below configuration for client_domain modparam("tls_mgm", "client_domain", "dom1") modparam("tls_mgm", "match_ip_address", "[dom1]*") modparam("tls_mgm", "tls_method", "[dom1]-TLSv1_2") modparam("tls_mgm", "certificate", "[dom1]/etc/opensips/tls/3cdomain.crt") modparam("tls_mgm", "private_key", "[dom1]/etc/opensips/tls/3cdomain.key") modparam("tls_mgm", "require_cert", "[dom1]0") modparam("tls_mgm", "verify_cert", "[dom1]0") With this configuration when I place an outbound call I am getting below error in the logs . I don't have the certificate and key of the next party . How can I authorized this certificate the provide on opensips end ? * NOTICE:tls_openssl:verify_callback: depth = 1, verify failure NOTICE:tls_openssl:verify_callback: subject = /C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./OU=http:\/\/certs.godaddy.com \/repository\//CN=Go Daddy Secure Certificate Authority - G2 NOTICE:tls_openssl:verify_callback: issuer = /C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./CN=Go Daddy Root Certificate Authority - G2 NOTICE:tls_openssl:verify_callback: verify error: unable to get local issuer certificate [error=20] INFO:tls_openssl:openssl_tls_connect: New TLS connection to 18.169.x.y:5065 established INFO:tls_openssl:tls_dump_cert_info: tls_connect: server TLS certificate subject: /CN=*.sftel.yyy.cloud, issuer: /C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./OU=http:\/\/certs.godaddy.com \/repository\//CN=Go Daddy Secure Certificate Authority - G2 WARNING:tls_openssl:openssl_tls_connect: TLS server certificate verification failed ERROR:tls_openssl:tls_dump_verification_failure: unable to get local issuer certificate INFO:tls_openssl:tls_dump_cert_info: tls_connect: local TLS client certificate subject: /CN=*.xxx.com , issuer: /C=GB/ST=Greater Manchester/L=Salford/O=Sectigo Limited/CN=xyz RSA Domain Validation Secure Server CA* *What should I do here ? * *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Thu, Aug 29, 2024 at 12:52 PM Sasmita Panda wrote: > Hi All , > > I am using opensips 3.2 from very long time . For TLS connection I was > using our domain specific certificate and private key which was authorized > by some verified organization . With that my TLS connection with the server > is getting established and also I am able to get REGISTER and INVITE > request on the connection . > > > Rather than this , when I build opensips with TLS=1 opensips itself > creates its own rootCA . If I am using those crt and private key file for > TLS connection the connection get established but I am not getting any > request . What can be the reason . > > My configuration is like below . > > modparam("tls_mgm", "server_domain", "dom3") > modparam("tls_mgm", "match_ip_address", "[dom3]20.1.x.y:5061") > modparam("tls_mgm", "match_sip_domain", "[dom3]none") > # 20.1.x.y this is my servers private IP on which I have configured TLS > socket . > modparam("tls_mgm", "tls_method", "[dom3]-TLSv1_2") > > modparam("tls_mgm", "certificate", > "[dom3]/etc/opensips/tls/rootCA/cacert.pem") > modparam("tls_mgm", "private_key", > "[dom3]/etc/opensips/tls/rootCA/private/cakey.pem") > modparam("tls_mgm", "ca_list", > "[dom3]/etc/opensips/tls/rootCA/certs/01.pem") > > modparam("tls_mgm", "require_cert", "[dom3]0") > modparam("tls_mgm", "verify_cert", "[dom3]1") > > In the logs I am getting below message > > > > *2024-08-29T07:14:59.213460+00:00 ip-20-1-205-63 /sbin/opensips[22895]: > INFO:tls_openssl:openssl_tls_accept: New TLS connection from x.x.x.x:20219 > accepted2024-08-29T07:14:59.213866+00:00 ip-20-1-205-63 > /sbin/opensips[22895]: INFO:tls_openssl:openssl_tls_accept: Client did not > present a TLS certificate2024-08-29T07:14:59.214064+00:00 ip-20-1-205-63 > /sbin/opensips[22895]: INFO:tls_openssl:tls_dump_cert_info: tls_accept: > local TLS server certificate subject: > /CN=OpenSIPS/ST=opensips.org/C=IP/emailAddress=team at opensips.org/O=opensips.org > , > issuer: > /CN=OpenSIPS/ST=opensips.org/C=IP/emailAddress=team at opensips.org/O=opensips.org > * > > I have added siptrace and tracing to the DB as well . I am not getting any > SIP messages on the 2nd case . What can be the reason for this ? This is > quite critical to me . Please do help. > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > -------------- next part -------------- An HTML attachment was scrubbed... URL: From JPyle at fusionconnect.com Fri Aug 30 19:54:00 2024 From: JPyle at fusionconnect.com (Pyle, Jeff) Date: Fri, 30 Aug 2024 19:54:00 +0000 Subject: [OpenSIPS-Users] clusterer compatibility between 2.4 and 3.4 Message-ID: Hello, I have an OpenSIPS 2.4 cluster I need to upgrade to 3.4. The cluster is configured as an HA pair, with one active and one standby where keepalived moves the IP between the two. Can instances on 2.4 and 3.4 participate in the same cluster? I'm hoping to update one "half" at a time while maintaining call processing on the side that isn't being upgraded. Regards, Jeff This message is subject to Fusion Connect, Inc.'s email communication policy: www.fusionconnect.com/email-policy -------------- next part -------------- An HTML attachment was scrubbed... URL: From ag at ag-projects.com Fri Aug 30 22:05:40 2024 From: ag at ag-projects.com (Adrian Georgescu) Date: Fri, 30 Aug 2024 19:05:40 -0300 Subject: [OpenSIPS-Users] clusterer compatibility between 2.4 and 3.4 In-Reply-To: References: Message-ID: <08BF7F86-F7E9-470C-ABBB-3DEE283FBE03@ag-projects.com> Because of the database schema differences between the two versions, you will not be able to use the same database to run both. One migration strategy could be: 1. Convert the configuration and migrate the database structure to the new version 3.4 on the current slave 2. Test the slave SIP logic using a separate IP address or port with a SIP client that uses the slave server address as outbound SIP Proxy. When everything works continue to the next step. 3. Switch over the cluster to the newly configured slave running the new OpenSIPS version 4. Copy the configurations from the newly promoted slave to master to the old master machine 5. Switch back to the old master Adrian > On 30. Aug 2024, at 16:54, Pyle, Jeff wrote: > > Hello, > > I have an OpenSIPS 2.4 cluster I need to upgrade to 3.4. The cluster is configured as an HA pair, with one active and one standby where keepalived moves the IP between the two. > > Can instances on 2.4 and 3.4 participate in the same cluster? I'm hoping to update one "half" at a time while maintaining call processing on the side that isn't being upgraded. > > > Regards, > Jeff > > This message is subject to Fusion Connect, Inc.’s email communication policy: www.fusionconnect.com/email-policy _______________________________________________ > 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: