[asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6

[Alex Ferrara]

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 [xxxx0812 at 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/849 at smallanimals' (thanks to SIP/812-00000016)
    -- Executing [849 at smallanimals:1] Progress("Local/849 at smallanimals-21bd;2", "") in new stack
    -- Executing [849 at smallanimals:2] Playback("Local/849 at smallanimals-21bd;2", "custom/ceh-meetingmsg") in new stack
    -- <Local/849 at 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/849 at 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 <alex at receptiveit.com.au> 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: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
> 
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