Re: [OpenSIPS-Users] Opensips as proxy for Asterisk
Hi, I am facing issue with INVITE . > > On 31 May 2024 at 3:46 pm, Bogdan-Andrei Iancuwrote: > > Hi, > > What exact part is not working for you? The REGISTERs? or the INVITEs? > > Regards, > > Bogdan-Andrei Iancu OpenSIPS Founder and Developer > https://www.opensips-solutions.com https://www.siphub.com > > On 11.04.2024 13:58, Sterlin Devanish wrote: > > > > > Hi friends, > > > > > > I am new to opensips. > > > > I am working on handling Background calls for Flutter WebRTC clients using > > Asterisk. > > > > > > > > Since Asterisk doesn't support RFC8599, I am trying to configure opensips > > as a proxy server for Asterisk. > > > > > > > > I am using mid_registrar to forward the registration request from opensips > > to asterisk. > > > > It is perfectly working for SIP signaling, whereas for WebSockets the > > request is not reaching the asterisk from opensips. > > > > > > > > Kindly help me where I am going wrong, or help me handle this scenario. > > > > > > > > > > > > > > Thanks, > > Sterlin Devanish D > > > > > > > > > > ___ Users mailing list > > Users@lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] utimer_ticker warning in opensips 3.2
You should check the opensips logs to check for the ERROR/CRITICAL messages that comes with the failure of OpenSIPS to start. Some similar to the CRITICAL you mentioned in the prev posts. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 31.05.2024 12:01, Sasmita Panda wrote: [opensips] this is a line I am getting in between the services of opensips when opensips is not listening . Attached the logs with mem Debug as you mentioned . */Thanks & Regards/* /Sasmita Panda/ /Senior Network Testing and Software Engineer/ /3CLogic , ph:07827611765/ On Thu, May 30, 2024 at 9:21 PM Bogdan-Andrei Iancu wrote: Hi, Try to start opensips is mem debugging support - add "|-a Q_MALLOC_DBG" to the startup cmd line of OpenSIPS - this will give more hints on that crashing point. Regards, | Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 27.05.2024 12:46, Sasmita Panda wrote: Hi , I am just facing the same issue again . /usr/local/sbin/opensips[3443]: CRITICAL:core:fm_free: freeing already freed shm pointer (0x7f3a59ddb4a0), first free: (null): (null)(0) - aborting! whenever I am starting opensips this is the last line I am getting in the logs and it wont listen on the specified port . As earlier suggested I was trying to run a trap with opensips-cli , but I am facing an issue with that . It says the trap module is not loaded . How do I load the trap module of opesnips-cli specifically ? Please help . This is a kind of blocker for me . */Thanks & Regards/* /Sasmita Panda/ /Senior Network Testing and Software Engineer/ /3CLogic , ph:07827611765/ On Thu, Mar 21, 2024 at 10:01 PM Bogdan-Andrei Iancu wrote: IF (a) it crashes, try to get a backtrace (b) it block on starting, try to do a "trap" via opensips-cli Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 21.03.2024 08:24, Sasmita Panda wrote: Sometimes it crashes and sometimes while starting I get the warings of the timer . Same config shows different issues . */Thanks & Regards/* /Sasmita Panda/ /Senior Network Testing and Software Engineer/ /3CLogic , ph:07827611765/ On Wed, Mar 20, 2024 at 6:45 PM Bogdan-Andrei Iancu wrote: Hi, How the two reports fit together here ? there are completely separate experiences on different runs?? or if you start opensips first you get the warnings and later it crashes ?? For the crash part, I see a core file was generated - could you extract the backtrace and post here ? (see https://opensips.org/Documentation/TroubleShooting-Crash) Regards Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11.03.2024 15:04, Sasmita Panda wrote: Any update on this ? */Thanks & Regards/* /Sasmita Panda/ /Senior Network Testing and Software Engineer/ /3CLogic , ph:07827611765/ On Mon, Mar 11, 2024 at 12:03 PM Sasmita Panda wrote: With the same server configuration and opensips version I am getting below error as well . CRITICAL:core:fm_free: freeing already freed shm pointer (0x7fc110e0b408), first free: (null): (null)(0) - aborting! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips as proxy for Asterisk
Ok, and where the things are getting broken with the INVITE? is an INVITE from the webrtc client? does it get to OpenSIPS? is OpenSIPS forwarding it? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 02.06.2024 09:21, sterlin wrote: Hi, I am facing issue with INVITE . On 31 May 2024 at 3:46 pm, Bogdan-Andrei Iancu wrote: Hi, What exact part is not working for you? The REGISTERs? or the INVITEs? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11.04.2024 13:58, Sterlin Devanish wrote: Hi friends, I am new to opensips. I am working on handling Background calls for Flutter WebRTC clients using Asterisk. Since Asterisk doesn't support RFC8599, I am trying to configure opensips as a proxy server for Asterisk. I am using mid_registrar to forward the registration request from opensips to asterisk. It is perfectly working for SIP signaling, whereas for WebSockets the request is not reaching the asterisk from opensips. Kindly help me where I am going wrong, or help me handle this scenario. /Thanks,/ /*Sterlin Devanish D*/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Call forward at opensips with rtpegine in use - fail to connect media
Hi
I am running one opensips with rtpengine
opensips -V
version: opensips 3.2.9 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
main.c compiled on with gcc 8
And have issue with attendend transfer:
Here is the basic scenario:
A calls B
B puts A on hold (A listens to MOH)
B makes second call C and C answers
So far all good, B and C can talk, but now when I want to transfer A to C,
opensips receives REFER and replies with "202 Accepted"
However now I dont know how to connect media with A and C
A is still listening to MOH and C has silence.
Here is the part that handles REFER and MOH
route[handle_sequential]
{
if(has_totag() &&is_method("REFER") &&loose_route()) {
xlog("Route: handle_sequential | method: $rm | call_transfer | rt $rt");
append_to_reply("Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE,
INFO, UPDATE, REGISTER, REFER, NOTIFY\r\n");
append_to_reply("Supported: timer, path, replaces\r\n");
append_to_reply("Allow-Events: talk, hold, conference, refer\r\n");
append_to_reply("Contact: \r\n");
append_to_reply("Expires: 60\r\n");
append_to_reply("User-Agent: rtpengine\r\n");
send_reply(202, "Accepted");
exit;
}
if(is_method("INVITE") &&has_totag()) {
if(is_audio_on_hold()) {
xlog("Start playing MusicOnHold");
xlog("request code: $rc");
$var(on_hold) ="1";
append_hf("SDP-Rewritten: MOH\r\n");
rtpengine_play_media("from-tag=$tt file=/etc/opensips/music/moh1.wav");
rtpengine_manage("RTP/AVP replace-session-connection replace-origin
trust-address ICE=remove");
}
else{
$var(on_hold) ="0";
xlog("Stop playing MusicOnHold: var(on_hold): $var(on_hold)");
rtpengine_stop_media("from-tag=$tt");
append_hf("SDP-Rewritten: Return from MOH\r\n");
rtpengine_manage("RTP/AVP replace-session-connection replace-origin
trust-address ICE=remove");
}
} else{}...
}
I tried also with b2b_logic but I obviusly dont have enough knowledge to
make it work by myself
Can you please give me a hint how to proceed?
Or if anyone has a working example, what part of code is missing here?
Thank you
BR
Simon
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