Hi Paul,
I tried adding Progress() to no avail. I still get no audio and below is what
comes up in the console.
-- Accepting call from '403xxxxxx' to 'xxxx0812' on channel 0/10, span 1
-- Executing [xxxx0...@isdn-incoming:1] Dial("DAHDI/10-1", "SIP/812,60") in
new stack
== Using SIP RTP CoS mark 5
-- Called 812
-- Got SIP response 302 "Moved Temporarily" back from 192.168.1.148
-- Now forwarding DAHDI/10-1 to 'Local/8...@smallanimals' (thanks to
SIP/812-00000016)
-- Executing [...@smallanimals:1]
Progress("Local/8...@smallanimals-21bd;2", "") in new stack
-- Executing [...@smallanimals:2]
Playback("Local/8...@smallanimals-21bd;2", "custom/ceh-meetingmsg") in new stack
-- <Local/8...@smallanimals-21bd;2> Playing 'custom/ceh-meetingmsg.gsm'
(language 'en')
-- Channel 0/10, span 1 got hangup request, cause 16
== Spawn extension (isdn-incoming, xxxx0812, 1) exited non-zero on
'DAHDI/10-1'
-- Hungup 'DAHDI/10-1'
== Spawn extension (smallanimals, 849, 2) exited non-zero on
'Local/8...@smallanimals-21bd;2'
The notion brought up earlier of a codec mismatch and Asterisk not transcoding
feels like the right answer, but I won't know until I get on site.
Thanks for the reply.
aF
On 31/08/2010, at 10:47 PM, Paul Belanger wrote:
> On Tue, Aug 31, 2010 at 4:04 AM, Alex Ferrara <[email protected]> wrote:
>> exten => 849,1,Playback(custom/ceh-meetingmsg)
>> exten => 849,n,Hangup
>>
> exten => 849,1,Progress()
> exten => 849,n,Playback(custom/ceh-meetingmsg)
> exten => 849,n,Hangup
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: [email protected] | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
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