Re: [asterisk-users] Calls are dropped after 15 minutes

2016-08-09 Thread Keith Heppner
The solution that fixed our problem was to Edit the sip_general_additional.conf file by adding the line "session-timers=refuse" Thank you to each one who gave suggestions. Keith Keith Heppner Rio Grande Bible Institute 4300 S Business Highway 281 Edinburg, TX 78539-9650 Office 956-380-8171 Cell

[asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-09 Thread Jonas Kellens
Hello I'm 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 get local SDP) I followed this

Re: [asterisk-users] EAGI script with missing audio on /dev/fd/3

2016-08-09 Thread Matt Fredrickson
On Tue, Aug 2, 2016 at 11:42 AM, nik600 wrote: > Dear all > > i'm trying to access to the input audio raw stream with a very basic EAGI > script: > > > #!/bin/sh > echo "EXEC Queue 2001" > cat /dev/fd/3 > /tmp/pippo > > This is my dialplan: > > exten => 001,NoOp(test) > exten

Re: [asterisk-users] Asterisk & Vitelity Invite issues

2016-08-09 Thread Matt Fredrickson
On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly wrote: > Hi 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

Re: [asterisk-users] Trouble applying regex to dialplan variable that contains double-quotes

2016-08-09 Thread Alex Villací­s Lasso
El 08/08/16 a las 21:34, Eric Wieling escribió: How Set handles quotes can be changed with the 'app_set' setting in the [compat] section of /etc/asterisk/asterisk.conf. See also: https://wiki.asterisk.org/wiki/display/AST/Application_Set Perhaps you have the value left over from an old

Re: [asterisk-users] Asterisk 13 High CPU usage

2016-08-09 Thread Matthew Jordan
On Sat, Aug 6, 2016 at 11:13 AM, Chirag Desai wrote: > All, > > I upgraded to asterisk 13.10. I have minimal load on the box. 20-30 calls a > day. > > Right now, there are no calls on the box at all. > > top shows me this: > > PR 20 > > NI 0 > > VIRT 1570540 > > RES 84620 >

Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

2016-08-09 Thread Faheem Muhammad
Jacek, This might be a bug or configuration issue, but you need to understand the SIP Session Timers. With Session Timers you can control the round trip time and Call Setup time of SIP Requests. In case of GSM Network with high delay you need to set the T1 timer a higher value like 1000ms (500 ms

[asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

2016-08-09 Thread Jacek Konieczny
Hi, 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 don't like. With chan_pjsip when a phone went unexpectedly offline (Ethernet cable disconnected) Asterisk would