On 01/18/2011 05:27 PM, Shaun Ruffell wrote:
On 01/18/2011 04:06 PM, Danny Nicholas wrote:
-----Original Message-----
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Tuesday, January 18, 2011 3:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] 1.8.2: dahdi-2.4: calls dropping
Here's a call coming in over PSTN to dahdi/4, connected to a local
extension dahdi/1:
-- Executing [s@incoming-pstn-line:1] Answer("DAHDI/4-1", "") in
new stack
..........
-- Executing [s@incoming-pstn-line:6] Dial("DAHDI/4-1",
"DAHDI/g0,36") in new stack
-- Called g0
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 answered DAHDI/4-1
-- Native bridging DAHDI/4-1 and DAHDI/1-1
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
== Spawn extension (incoming-pstn-line, s, 6) exited non-zero on
'DAHDI/4-1'
-- Hanging up on 'DAHDI/4-1'
-- Hungup 'DAHDI/4-1'
I'm on dadhi/1. After 10-15 minutes, the call just drops. What gives?
sean
Just a WAG - the bridge isn't really happening and you're getting a dial
timeout.
If you were running trunk...this is a very good guess. The following
commit resolved an issue with bridging that's been in trunk for the past
few weeks.
http://svn.asterisk.org/view/dahdi?view=revision&revision=9642
Cheers,
Shaun
Wasn't running trunk. It was the 1.8.2 release. Not sure I understand:
the dial timeout is 36 seconds. Yet the call doesn't drop for at least
5, probably 10, maybe more minutes.
And no audio was muted while the call was up. It was all just fine.
sean
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
