Archives : August-2016
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I installed PJSIP from the project git clone https://github.com/asterisk/pjproject pjproject cd pjproject make uninstall & make distclean./configure –libdir=/usr/lib64 –prefix=/ –enable-shared –disable-sound–disable-resample –disable-video –disable-opencore-amr–with-external-s..
Is there any configuration change in asterisk 13.9.1 to show original callerid on a transfer In asterisk 11.21 it works as expe..
I keep getting messages like these in the cli: [Aug 10 12:20:17] WARNING[23411]: 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[23411]: 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..
HelloIm trying for several days now to get ICE support for my Asterisk 11.23 on CentOS 6.My call setup : sipml5_webRTC (nat) –> public Asterisk on 178.18.90.230 –> softphone Zoiper(problem : no audio)Reverse does not work either.(problem : failed ..
We have been migrating our PBX system from Asterisk 1.8 and chan_sip to Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have stumbled on a behaviour difference I dont like. With chan_pjsip when a phone went unexpectedly offline (Ether..
I am writing a dialplan context under asterisk 11.21.0 to handle SIP message routing between registered SIP peers using chan_sip. I am having trouble with double-quotes when the source peer uses a display name, which appears in quotes before the ..
There is a separate app for recording voice (app_record) or dtmf input (app_read). But there is no way to allow the user to choose to enter by voice or by keypad in same time.app_record analyzes the dtmf input, but only the # and * (to quit). Noth..
All,We have asterisk 11.23 running sip to vitelity and from there IAX trunks split off to where they need to go.We are having a problem getting chan_sip to quit ignoring re-invites from Vitelity.Our side ends up sending a reinvite which their sid..
I have this in my config:exten => _800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})same => n,Dial(SIP/tollfree/1${EXTEN})exten => _1800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})same => n,Dial(SIP/tollfree/${EXTEN}) ex..