similar to: pjsip: outofcall_message_context

07.03.2015 0:24, Kevin Harwell ?????: > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in > pjsip. > > I have a

06.09.2016 16:42, George Joseph ?????: > > > On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Several months server working on asterisk 13.7 and pjproject 2.5 > (installed separately). Once a day the server crashes or hangs and > is familiar sores that written

The call initiated from internal extension. I have made two test call: Successful call from device on res_pjsip via endpoint on chan_sip: http://pastebin.com/LWeDYstj Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: http://pastebin.com/hepVb6Nu And ones again i don't see anything that would make asterisk send BYE. I would be grateful for any ideas. 11.02.2016 1:47,

05.10.2015 23:24, Joshua Colp ?????: > On 15-10-05 05:22 PM, Dmitriy Serov wrote: >> Hello. Do I understand correctly that the current implementation >> res_pjsip does not support ZRTP? >> http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html > > ZRTP is not supported in Asterisk itself. > >> Nothing has changed since 2013? P.S. I greatly

Hello. Do I understand correctly that the current implementation res_pjsip does not support ZRTP? http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html Nothing has changed since 2013? P.S. I greatly regret that moved from chan_sip to res_pjsip. Previously used very much lacking, and much of the promise failed. Dmitriy Serov. -------------- next part -------------- An HTML

The asterisk server has a permanent IP address, but the provider cannot ensure stable quality traffic for RTP. There is a desire to use an external server, the address of which shall be specified in the SDP, through which flowing media. I use asterisk 13.6 and res_pjsip. Prompt, please: 1. what software can be used on an external RTP proxy? 2. What settings need to be done in pjsip.conf to use

Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk]

06.10.2015 1:22, Joshua Colp ?????: > On 15-10-05 05:58 PM, Dmitriy Serov wrote: >> 05.10.2015 23:24, Joshua Colp ?????: >>> On 15-10-05 05:22 PM, Dmitriy Serov wrote: >>>> Hello. Do I understand correctly that the current implementation >>>> res_pjsip does not support ZRTP? >>>>

Hello. Voice quality when calling - this is one of the most important in the PBX. You need to record the quality parameters for each call to improve. Because the overall quality of a call can only be determined upon completion, I did it in the HangUp handler and wrote in custom fields of CDR. This worked well in asterisk 11. In asterisk 13 I did not find a handler after the call, but before

Hello. Asterisk 13.2, PJSIP. Problem: I do not get any AMI events when changing the status of the contact. When using chan_sip I got "peerstatus" event. When using res_pjsip and devices (endpoint configuration) I got "peerstatus" event. When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION i got "registry" event. When using

Yes, this means the remote end was not sending any audio stream. But it shouldn't. According to [1] before remote end send OK or ACK there is one way SDP, no any audio stream. PJSIP channel (option rtp_timeout) does not take this one. Isn't it a mistake? What could be workarounds? 19.12.2016 11:33, Jean Aunis ?????: > > This means the remote end was not sending any audio stream,

Good day. Asterisk 13.7.2, res_pjsip. There is a problem of loss of registration of several devices. This happens not on all devices, but problem devices a lot. Below is the log of registration of a contact of one device. Is suspect two things: 1. delete a contact after the contact is added. But, like, it's a feature of code that may already be fixed. 2. deleting a contact much earlier

Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten => s,n,Dial(SIP/peer1/number at domain2.com,60,r) [peer1] type=friend host=domain1.com fromdomain=domain1.com As a result in SIP packet uri: number at domain2.com@domain1.com I need: number at domain2.com I can't use "SIP uri dial", i need

On Wed, 22 Nov 2017, Dmitriy Serov wrote: > ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" > > /var/spool/asterisk/outgoing/${number}-${confnum}) I get: Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/... Unknown keyword 'ActionID' at line 2 of

Hi Dmitriy and others and thanks for your help so far. The option "match_auth_username=yes" seems to have had no effect. From my reading, this option will try to match the username of the incoming SIP account to a section heading. If that is how it must work then i can see a big problem. I'm trying to present the receptionist with a nice display of which line the call came in on.

Hello. I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip. Now i am transfering all from chan_sip to chan_pjsip. And have a lot of questions. First of... system: Asterisk 13.2 on slackware 14.1 Errors on outgoing call: [2015-03-03 00:18:58] ERROR[6794]: chan_pjsip.c:1778 request: Failed to create outgoing session to endpoint 'srv_d228' [2015-03-03 00:18:58]

Hello. After successful compilation 15.6.1 (bundled pjsip) and start asterisk i has error Symbol pjsip_tls_transport_start2 not found. /main/libasteriskpj.exports does not containg pjsip_tls_transport_start2 and pjsip_tls_transport_start. More: * All versions before (including 15.5) has not such error on this computer (ubuntu 18.04). * with 15.6.0, 15.6.1 has error on this computer

Hello. Several months server working on asterisk 13.7 and pjproject 2.5 (installed separately). Once a day the server crashes or hangs and is familiar sores that written watchdogs. Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5). Solved all the problems with compilation I started asterisk several times and each time after 5-7 seconds was seg fault. So I didn't get

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote: > 07.03.2015 0:24, Kevin Harwell ?????: > > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com> > wrote: > >> Hello. >> >> Asterisk 13.2. >> I transfer configs from chan_sip to res_pjsip. >> In chan_sip i have