search for: ps_contacts

Has anyone successfully used Mysql realtime PJSIP with Asterisk 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the following error now: Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'expiration_time' cannot be type 'bigint(20)' (need char) [Sep 12 14:42:35] WARNIN...

...Has anyone successfully used Mysql realtime PJSIP with Asterisk >> 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the >> following error now: >> >> Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: >> Realtime table general at ps_contacts: column 'qualify_timeout' cannot be >> type 'int(10)' (need char) >> [Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: >> Realtime table general at ps_contacts: column 'expiration_time' cannot be >> type 'bigint(20)' (ne...

I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'expiration_time' cannot be type 'bigint(20)' (need char) [Sep 2 15:38:46] WARNING...

...sfully used Mysql realtime PJSIP with Asterisk >>> 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the >>> following error now: >>> >>> Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: >>> Realtime table general at ps_contacts: column 'qualify_timeout' cannot be >>> type 'int(10)' (need char) >>> [Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: >>> Realtime table general at ps_contacts: column 'expiration_time' cannot be >>> type 'big...

...DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8) called (obj->txf = (nil)) [Jun 15 16:20:03] ERROR[5116] res_pjsip_registrar.c: Unable to bind contact 'sip:812 at 10.1.80.112:5062' to AOR '812' [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: SELECT * FROM ps_contacts WHERE id LIKE ? ORDER BY id [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id LIKE') = '812;@%' [Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8) called (obj->txf = (nil)) [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: SEL...

Hi, Yes, we're implementing the dialplan in realtime too. Here the contents of sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints aor=realtime,ps_aors contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Annus Fictus Sent: 13 June 2016 14:11 To: Asterisk Users Mailing List - Non-Commercial Dis...

...Contact: pbx-node-1/sip:myurl.ch:5060 771bf6a7d4 Avail 8.833 The extconfig.conf file looks like this: [settings] ps_endpoints => odbc,asterisk ps_auths => odbc,asterisk ps_aors => odbc,asterisk ps_domain_aliases => odbc,asterisk ps_endpoint_id_ips => odbc,asterisk ps_contacts => odbc,asterisk extensions => odbc,asterisk Any idea why I need to reload PJSIP if I want the aors to be qualified? Cheers, Francisco. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/201606...

I have a project that is requiring the use of MS SQL from asterisk. I get an error when the pjsip contact tries to update the contact table. [Oct 15 21:34:55] WARNING[3033]: res_odbc.c:649 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 22018: [FreeTDS][SQL Server]Conversion failed when converting the varchar value '3.000000' to data type int. (101) The datatype

Hi all, Is there any native way to save endpoint statuses to database? I use asterisk 13 with pjsip and realtime, and didn't found proper way. I read that there is config parameter in sip.conf: rtupdate=yes. But how can I do that with pjsip? Or I should use sip.conf with pjsip simultaneously. Or is there any kind of hooks, which allows make custom action on endpoint status change. Thanks.

On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote: > Joshua > > That is the interesting part of it. We took our configs and database > tables from our working 13.12.2 deployments and tried to use them with > our > new 13.17.1 deployments and we are having issues where the tables are not > working. On the new server asterisk keeps saying it can't find the

...fixed in this release: ----------------------------------- * ASTERISK-29057 - pjsip: Crash on call rejection during high load (Reported by Sandro Gauci) Improvements made in this release: ----------------------------------- * ASTERISK-29056 - Increase reg_server column size for ps_contacts table realtime (Reported by sungtae kim) * ASTERISK-29055 - Create a Bridge with video_single mode (Reported by sungtae kim) Bugs fixed in this release: ----------------------------------- * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending invites (with auth)...

...ll rejection during high load (Reported by Sandro Gauci) Improvements made in this release: ----------------------------------- * ASTERISK-29055 - Create a Bridge with video_single mode (Reported by sungtae kim) * ASTERISK-29056 - Increase reg_server column size for ps_contacts table realtime (Reported by sungtae kim) Bugs fixed in this release: ----------------------------------- * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies (Reported by Sebastian Damm) * ASTERISK-29124 - res_pjsip: flow tran...

...wi_subscribe_replaces_unsolicited" field is integer not boolean (Reported by Joshua C. Colp) * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set 'received' for IPv6 (Reported by Sean Bright) * ASTERISK-28022 - res_pjsip realtime: uri column in ps_contacts table can be too short (Reported by Florian Floimair) Improvements made in this release: ----------------------------------- * ASTERISK-28046 - Remove stale nonoptreq references (Reported by Walter Doekes) For a full list of changes in this release, please see the ChangeL...

...elease: ----------------------------------- * ASTERISK-29054 - Logger: Add debug logging categories (Reported by Kevin Harwell) * ASTERISK-29055 - Create a Bridge with video_single mode (Reported by sungtae kim) * ASTERISK-29056 - Increase reg_server column size for ps_contacts table realtime (Reported by sungtae kim) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0 Thank you for your continued support of Asterisk! -------------- next part -------------- An HTML attachment...

...aren't handled correctly (Reported by George Joseph) Improvements made in this release: ----------------------------------- * ASTERISK-29054 - Logger: Add debug logging categories (Reported by Kevin Harwell) * ASTERISK-29056 - Increase reg_server column size for ps_contacts table realtime (Reported by sungtae kim) * ASTERISK-29055 - Create a Bridge with video_single mode (Reported by sungtae kim) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0 Thank you fo...

...The following issues are resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get authentification error (Reported by Ian Gilmour) * ASTERISK-28022 - res_pjsip realtime: uri column in ps_contacts table can be too short (Reported by Florian Floimair) * ASTERISK-28011 - chan_sip: get_refer_info() attempted unlock mutex 'peer' without owning it! (Reported by Alec Davis) * ASTERISK-28002 - When T.140 realtime text is negociated, a lot of debug traces are...

Hello, I am moving from realtime chan_sip to pjsip and one of the problem I am facing is the lack of "sipregs". With chan_sip, when an extension registers, the server where it has registered to is stored in sipregs. Is there something similar in pjsip? How can I find on which server the pjsip extension has registered to? Leandro -------------- next part -------------- An HTML

...6359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen) * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled (Reported by Anthony Messina) * ASTERISK-26263 - SQL error when using realtime and registering extension / inserting into ps_contacts (Reported by Jeppe Ryskov Larsen) * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten for connectionful protocols (Reported by Joshua Colp) * ASTERISK-26367 - rtp: Timestamps broken when video frame is across multiple RTP packets (Reported by Joshua Colp) * ASTER...

...n T.140 realtime text is negociated, a lot of debug traces are generated (Reported by Emmanuel BUU) * ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get authentification error (Reported by Ian Gilmour) * ASTERISK-28022 - res_pjsip realtime: uri column in ps_contacts table can be too short (Reported by Florian Floimair) * ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE (Reported by Joshua Elson) * ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of offer...

...#39;t compile when LOW_MEMORY is enabled (Reported by Anthony Messina) * ASTERISK-26391 - Consoles do not display verbose logger messages even when requested. (Reported by Marcelo Terres) * ASTERISK-26263 - SQL error when using realtime and registering extension / inserting into ps_contacts (Reported by Jeppe Ryskov Larsen) * ASTERISK-26365 - rtp: Offer with multiple payloads for same codec is incorrectly handled (Reported by Joshua Colp) * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra) * ASTERISK-26374 - res_pjsip_multihomed: Co...