Hi Lonnie and all

Thanks for the help. "directmedia=no" seems to have done the trick (awaiting
firmer confirmation), even though we had not set canreinvite before.

Thanks again

Tom


-----Original Message-----
From: Lonnie Abelbeck [mailto:li...@lonnie.abelbeck.com] 
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:li...@mksolutions.info] 
> Sent: 22 July 2013 15:57
> To: nnpait.servi...@googlemail.com; AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] Delay before audio is audible
> 
> 
> Am 22.07.2013 um 16:45 schrieb "Tom Chadwin"
> <nnpait.servi...@googlemail.com>:
> 
>> 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
> 
> 
> 
> 
> 
> 
>
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