search for: res_pjsip_outbound_registration

...o in this case). So, after a number of Asterisk restarts, FreePBX will reject new registrations, as max_contacts is reached. Unfortunately, "line" does not save random between restarts. It's also unable to specify "random" value in pjsip.conf. I'm thinking to patch res_pjsip_outbound_registration to add this feature. Am I wrong and  there is another way ? It's also a security hole, as anybody can generate INVITE with ";line=random" from any IP address ! res_pjsip_outbound_registration will only match "line", but will not take care about source IP, ... Is there...

...se actually getting to Asterisk? Does it show up in > "pjsip set logger on"? Is the REGISTER a retransmission and thus > expected to be the same? It become stranger and stranger: on one of the register peer we receive in asterisk: *CLI> [2020-01-19 15:23:18] WARNING[17469]: res_pjsip_outbound_registration.c:1021 handle_registration_response: Fatal response '401' received from 'sip:<myprovider>' on registration attempt to 'sip:<myuser>@<myprovider>', stopping outbound registration On the other one: [2020-01-19 15:23:46] WARNING[17469]: res_pjsip_outbound...

I am trying to log my SIP registration attempts. PJSIP is in logger mode, and I can see INVITES comingh, my SIP Register does not show, especially the packet I send. The only thing shown is: res_pjsip_outbound_registration.c: No response received from 'sxxxxnet' on registration attempt to 'sip:7866314772-xxxxxnet', retrying in '60' How do I see my own packet? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/att...

...ten_state.so res_pjsip_header_funcs.so res_pjsip_logger.so res_pjsip_messaging.so res_pjsip_multihomed.so res_pjsip_mwi_body_generator.so res_pjsip_mwi.so res_pjsip_nat.so res_pjsip_notify.so res_pjsip_one_touch_record_info.so res_pjsip_outbound_authenticator_digest.so res_pjsip_outbound_publish.so res_pjsip_outbound_registration.so res_pjsip_path.so res_pjsip_pidf_body_generator.so res_pjsip_pidf_digium_body_supplement.so res_pjsip_pidf_eyebeam_body_supplement.so res_pjsip_publish_asterisk.so res_pjsip_pubsub.so res_pjsip_refer.so res_pjsip_registrar_expire.so res_pjsip_registrar.so res_pjsip_rfc3326.so res_pjsip_sdp_rtp.s...

...sting >> documentation! >> If I switch to "registrations=realtime,px1_ps_registrations" the >> error stops, but I get now calls from the px1_ps_registrations table >> from the database. >> What could be missing? > > Outbound registrations are done in res_pjsip_outbound_registration, as > a result the registration= needs to be in a section for that module instead. > > -- > Joshua > That seems to have been the issue. Is there a documentation page out > there that highlights which options goes under which modules. > I have not run across this yet and am w...

...pjsip' because it has no existing > documentation! > If I switch to "registrations=realtime,px1_ps_registrations" the error > stops, but I get now calls from the px1_ps_registrations table from the > database. > What could be missing? Outbound registrations are done in res_pjsip_outbound_registration, as a result the registration= needs to be in a section for that module instead. -- Joshua That seems to have been the issue. Is there a documentation page out there that highlights which options goes under which modules. I have not run across this yet and am wondering if I am going to...

On Sun, Jan 19, 2020 at 10:45 AM Administrator <admin at tootai.net> wrote: <snip> > It become stranger and stranger: on one of the register peer we receive in > asterisk: > > *CLI> [2020-01-19 15:23:18] WARNING[17469]: > res_pjsip_outbound_registration.c:1021 handle_registration_response: Fatal > response '401' received from 'sip:<myprovider>' on registration attempt to > 'sip:<myuser>@<myprovider>', stopping outbound registration > What is the actual full configuration for this? > On the oth...

...even successfully shown in asterisk (it is a freepbx installation). But when doing a second call the number is busy ("provider" busy, I don't see anything in asterisk verbose mode). Sending a pjsip unregister results in the following messages: [2016-10-15 10:03:22] WARNING[10162]: res_pjsip_outbound_registration.c:761 schedule_retry: No response received from 'sip:sipgate.de:5060' on registration attempt to 'sip:2636146e0 at sipgate.de:5060', retrying in '60' -- Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now Reachable. RTT: 434.393 msec == Endpoint pjsip_s...

...ulling from ODBC On 15-10-05 09:16 AM, Ryan, Travis wrote: [snip] > > > So should anyone using realtime PJSIP be using the registrations line? Even if it's not used for any trunking? A registrations line in sorcery.conf for res_pjsip would do absolutely nothing. If you put it under res_pjsip_outbound_registration and have no outbound registrations it will execute some queries against your database but otherwise do nothing. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- ____________________...

On 15-10-04 09:54 AM, Bryant Zimmerman wrote: > I have a pjsip install that is not pulling it's realtime registrations > from an ODBC database. > When I have them directly pull from a MySQL database everything seems to > work. > I am having trouble finding a requirements document for the pjsip > realtime registrations. > Is there some kind of special config that has to be

...r Addr of a peer (Reported by Andrey V. T.) * ASTERISK-28580 - Bypass SYSTEM write permission in manager action allows system commands execution (Reported by Eliel Sarda��ons) Improvements made in this release: ----------------------------------- * ASTERISK-28602 - res_pjsip_outbound_registration: Maximum retries reached (Reported by Daniel) * ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam Banks) * ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column (Reported by cm...

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple

http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ -- Dave Topping e: info at dntopping.uk t: 03445 888 888 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170901/ae060564/attachment.html>

...eško marek.gresko at protonmail.com wrote: > > > Hello, > > > > sure I have local DNS server and public resolving should not be needed for phone registrations. Running pjsip show endpojnt show the endpoints as not in use. > > > > When looking into logs I see only res_pjsip_outbound_registration.c: No response > > received from sip provider. Nothing else. > > > > In phone log I see: > > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > > lid=0, par=0, par2=(nil)) > > > > The phone is Cisco SPA525G2. > > > > Thanks. > > >...

...k Greško < > marek.gresko at protonmail.com> wrote: > > Hello, > > sure I have local DNS server and public resolving should not be needed for > phone registrations. Running pjsip show endpojnt show the endpoints as not > in use. > > When looking into logs I see only res_pjsip_outbound_registration.c: No > response > received from sip provider. Nothing else. > > In phone log I see: > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > lid=0, par=0, par2=(nil)) > > The phone is Cisco SPA525G2. > > Thanks. > > Marek > > > > ------- Original Messa...

Le 17/01/2020 à 11:54, Administrator a écrit : > > Le 15/01/2020 à 19:24, Administrator a écrit : >> Hi all, >> >> we face a strange behavior while connecting an Asterisk16 instance >> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of >> them having Kamailio as front-end. With other providers -we don't >> know if they run

...stem commands execution (Reported by Eliel Sarda��ons) * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash (Reported by Alexei Gradinari) Improvements made in this release: ----------------------------------- * ASTERISK-28602 - res_pjsip_outbound_registration: Maximum retries reached (Reported by Daniel) * ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam Banks) * ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column (Reported by cm...

doesnt work for me Dne 29.5.2016 v 17:48 Niklas Larsson napsal(a): > Hi, > > > On 2016-05-27 18:28, Marek ?ervenka wrote: >> after downgrade to 13.8.2 >> May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip >> b49162cd sp bfac0940 error 4 in >> libmysqlclient.so.16.0.0[b48f1000+12e000] >> >> after downgrade to 13.7.2 >>

...(Reported by Matt Jordan) * ASTERISK-25040 - pbx: Improve performance of reloads by making hint destruction more performant (Reported by Matt Jordan) * ASTERISK-25114 - res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes (Reported by George Joseph) * ASTERISK-25072 - res_pjsip_outbound_registration: line functionality. Additional check for using the request URI (Reported by Dmitriy Serov) Bugs fixed in this release: ----------------------------------- * ASTERISK-25250 - chan_sip - Despite the channel being answered, caller on a call established via Local channel continues...

...rote: > > > > > Hello, > > > > > > sure I have local DNS server and public resolving should not be needed > for phone registrations. Running pjsip show endpojnt show the endpoints as > not in use. > > > > > > When looking into logs I see only res_pjsip_outbound_registration.c: > No response > > > received from sip provider. Nothing else. > > > > > > In phone log I see: > > > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > > > lid=0, par=0, par2=(nil)) > > > > > > The phone is Cisco SPA525G2. > &g...