Tom,
I don't know if my recent experience has anything to do with yours.
However, I saw a coincidental change in Asterisk behavior between 1.8.21
and 1.8.22 regarding session timers. Perhaps this is related.
Over the weekend, I upgraded from Asterisk 1.8.21 to Asterisk 1.8.22.
Following the upgrade, incoming DID calls would ring, have no audio for
a few seconds, and then fail. After subsequently downgrading back to
1.8.21, the symptom cleared.
In reviewing SIP traces from a successful 1.8.21 call and an
unsuccessful 1.8.22 call, I found the following differences. In the
"Not Working" example, it appears that Asterisk 1.8.22 is repeatedly
sending a 200 OK but never receives an ACK response. Significantly, the
200 OK from 1.8.22 (which fails) includes these two lines that don't
appear in 1.8.21 (which works):
* Session-Expires: 1800;refresher=uas
* Require: timer
While doing some research this morning, I learned that the DID provider
now recommends session-timers=refuse to be set in sip.conf. This makes
me wonder if they routinely ignore any 200 OK with a required timer.
Next weekend, I'll upgrade to 1.8.22 and try this configuration change.
If desired, I can post the results.
I hope this is useful.
Dan
On 7/22/2013 11:42 AM, Michael Keuter wrote:
Am 22.07.2013 um 17:26 schrieb Tom Chadwin <[email protected]>:
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
In AstLinux 1.1.2 Asterisk version 1.8.22.0 is included, in AstLinux 1.1.1 it
was version 1.8.21.0, in 1.1.0 it was 1.8.20.1.
Maybe you can find something related in the Changelogs, or you can see
something related on the console.
-----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
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Donations to support AstLinux are graciously accepted via PayPal to
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