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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 w...

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. > >...

...ng it from the CLI? It looks like it is > from an internal extension, if I were guessing, but that side of the > call isn't in your log. > > If it is from an internal extension, I think a SIP trace on that side > would help. > > > On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Please help find the cause of strange behavior res_pjsip. > > Making outgoint call to other sip server (CommuniGatePro), my > asterisk suddenly sends BYE after picking up! > Parti...

...registration will be INVITE sip:siptrunk-in at .... I offer to change res_pjsip_endpoint_identifier_user to realize endpoint identification by sip uri. I think it will be usefull. P.S. i hope issues will be rejected: https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069 Dmitriy Serov -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150306/2793b808/attachment.html>

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 20...

...used on an external RTP proxy? 2. What settings need to be done in pjsip.conf to use this external RTP proxy? Preferably specifies the external RTP proxy to specify a specific endpoint, not globally. If only globally valid, the suit and the decision. I would be grateful for any clues. Dmitriy Serov. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151102/16618c8b/attachment.html>

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. >&...

...rrent 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 attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151005/e85b0ff4/attachment.html>

...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 than the 90 seconds specified during the registration Would be grateful for any clues. Dmitriy Serov. expiration settings: [common-aor](!) type=aor qualify_frequency=60 default_expiration=120 maximum_expiration=600 minimum_expiration=90 log: [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact 'sip:17367 at 46.39.229.18:37910' to AOR '17367' with expiration...

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

...he AGI and there to update the CDR record by unique identifiers. But faced with the fact that there are no needed record in the table yet. To write the data into a separate table and join them may be an option. But do not want to resort to such a decision How do you solve this problem? Dmitriy Serov.

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? >>>> http://lists.digium.com/pipermail/asteris...

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 "match_auth_username=yes" and have nothing in pjsip. > > I have a lot of endpoints and registrations on same SIP server. And it...

Hello. Is there an analog option "outofcall_message_context" for pjsip? or: how to determine that the "call" is an outbound text message? Dmitriy Serov.

...) [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 authorization (peer1) I think asterisk can't do that. Is where work around? Dmitriy Serov.

...pseudodid}" = "086"]?internal,89,1:11) > exten => s,11,GotoIf($["${pseudodid}" = "087"]?internal,36,1:12) > exten => s,12,GotoIf($["${pseudodid}" = "088"]?internal,13,1:13) -Andrew Galdes On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote: > > This is one of the chronic problems. Try this option in sip.conf: > match_auth_username=yes > > Carefully read the description, it is better to test in "after hours". > > 02.04.2015 2:50, Andrew Galdes ?????: > > Hello...

...19.12.2016 11:33, Jean Aunis ?????: > > This means the remote end was not sending any audio stream, or the > audio stream was not received by Asterisk. The problem may have many > different reasons, but often it is a network-related issue. > > > Le 16/12/2016 ? 21:19, Dmitriy Serov a ?crit : >> Today I faced a problem. Please help to solve this problem. >> >> Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware >> v2.06(AAGJ.9)C1 >> >> Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip >> trunk). >&g...

...=userpass username=login password=secret [srv_dev] type=aor contact=sip:sip.example.com:5060 qualify_frequency=5 default_expiration=10 max_contacts=1 remove_existing=yes [srv_dev] type=endpoint from_domain=example.com aors=srv_dev outbound_auth=srv_dev rewrite_contact=yes allow=!all,alaw Dmitriy Serov

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