Is there some documentation for all the available sorcery.conf mappings for realtime?Asterisk already includes some tables in the database that are not enabled by default on the sorcery.conf like transports and outbound registrations.There are no examp..
I am unable to force a hangup on a channel that has been stuck for over two days: IAX2/from-CD-11006 oficina27701 Up Dial IAX2/to-CD/2883 346713000746:24:59 Sotelo SoteloIAX2/to-CD-20713 I have tried hangup request IAX2/from-CD-11006 several times ..
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: res_config_mysql.c:1162 require_mysql: Realtime table general@ps_contac..
Anyone know an efficient way to get a list of the DAHDI channels?
Is there an AMI or ARI variable to get a list of all the channels?
Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez
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: [Sep2 15:38:46] WARNING: res_config_mysql.c:1162 require_mysql: Realtime table general@ps_contacts: col..
I find that using hints with PJSIP on Asterisk 13 is very unreliable compared to regular SIP.I see many phones as unavailable when they are in fact available.Usually hints will work fine for a while after a phone registers but after a while it will rem..
I am having a problem with Fanvil phones (X3) when they make a call through DAHDI.Pure SIP calls flow normally but when a call goes through a DANDinterface to the PSTN we only get one way audio.This is Asterisk 13.10.0 (bundled pjsip) and Dahdi 2.1..
I keep getting messages like these in the cli: [Aug 10 12:20:17] WARNING: res_config_mysql.c:1162 require_mysql: Realtime table general@ps_contacts: column qualify_timeout cannot be type int(10) (need char) [Aug 10 12:20:17] WARNING: res_config_mysql.c:1..
Anyone know a good replacement for phpagi?Unfortunately development stalled long ago and it has not been updated.What is the best solution for AMI and AGI on PHP?Thanks. — Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91..
Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL.What would be the proper way to do this for Asterisk 13 and PJSIP? — Telecomunicaciones Abier..