Hello again I set directmedia=no, and this seemed to fix the issue. However, it has now reoccurred. What else could cause this? Silence (no ringtone for caller) for some seconds, then audio establishes. Could it be something to do with our phones (Snom 300)? I had to disable SRTP (I think it was)to get them to work at all when I did the first upgrade to 1.8.
Thanks Tom > -----Original Message----- > From: Lonnie Abelbeck [mailto:[email protected]] > Sent: 22 July 2013 17:23 > To: AstLinux Users Mailing List > Cc: Tom Chadwin > Subject: Re: [Astlinux-users] Delay before audio is audible > > Tom, > > Are you setting "directmedia=no" in your sip.conf for the local extensions ? > This supersedes the old "canreinvite=no" in Asterisk 1.4 . > > Lonnie > > > > On Jul 22, 2013, at 10:26 AM, Tom Chadwin wrote: > > > Hi Michael > > > > I'm not using the r option in the dial command (we use TtHh). Also, calls > > via our berofix work correctly - it is only internal SIP calls (between > > Snom300/320s) which exhibit the behaviour. > > > > We were using Asterisk 1.8 before, whichever version was packaged in > with > > the immediately previous version of Astlinux. > > > > Any other ideas? > > > > Thanks > > > > Tom > > > > > > -----Original Message----- > > From: Michael Keuter [mailto:[email protected]] > > Sent: 22 July 2013 15:57 > > To: [email protected]; AstLinux Users Mailing List > > Subject: Re: [Astlinux-users] Delay before audio is audible > > > > > > Am 22.07.2013 um 16:45 schrieb "Tom Chadwin" > > <[email protected]>: > > > >> Hello all > >> > >> We upgraded recently to the most recent 1.8 Astlinux, and we have a > > problem > >> we've not encountered before. Intermittently on internal SIP-SIP calls > > only, > >> there is no audio for a varying number of seconds (between 2 and 10) - > >> caller hears no ringing tone, and neither party can hear each other until > >> audio establishes after this delay. The problem has appeared since the > >> latest Astlinux upgrade. I don't see anything untoward on the console, but > >> I'm no expert. External (DAHDI ISDN) calls are fine. > >> > >> Does anyone have any ideas, or can point me toward how to debug? > >> > >> Thanks > >> > >> Tom > > > > > > I had a similar issue in combination with a Berofix and the dial command > > using the "r" option. > > The solution was not using the "r" option, and instead activating the "Early > > Audio" option in the Berofix ("ea=1"). > > Check if you are using a Dial with "r" (ringing indication) option > > somewhere. > > Which Asterisk version did you use before the upgrade? > > > > Michael > > > > http://www.mksolutions.info > > > > > > > > > > > > > > ---------------------------------------------------------------------------- -- > > See everything from the browser to the database with AppDynamics > > Get end-to-end visibility with application monitoring from AppDynamics > > Isolate bottlenecks and diagnose root cause in seconds. > > Start your free trial of AppDynamics Pro today! > > > http://pubads.g.doubleclick.net/gampad/clk?id=48808831&iu=/4140/ostg.clk > trk > > _______________________________________________ > > Astlinux-users mailing list > > [email protected] > > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > > > Donations to support AstLinux are graciously accepted via PayPal to > [email protected]. > > > > ------------------------------------------------------------------------------ Get 100% visibility into Java/.NET code with AppDynamics Lite! It's a free troubleshooting tool designed for production. Get down to code-level detail for bottlenecks, with <2% overhead. Download for free and get started troubleshooting in minutes. http://pubads.g.doubleclick.net/gampad/clk?id=48897031&iu=/4140/ostg.clktrk _______________________________________________ Astlinux-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [email protected].
