asterisk users - Aug 2010 - No audio on call forward after upgrade from Asterisk 1.4 to 1.6

Hi everyone,

This is my first post to the list, although I am a long term user of Asterisk. I
have recently found a problem that I just can't seem to solve.

I have a client that has an Ubuntu x64 based Asterisk server with and ISDN Dahdi
interface and about 25 SIP handsets. Everything was working fine in Asterisk 1.4
and now after migrating the config to Asterisk 1.6.2.5 I have one single issue
that I can't explain.

I have an extension that if you call it, it will play a sound file and hangup.
Pretty simple stuff. Below is the extensions.conf entry for this extension.

exten => 849,1,Playback(custom/ceh-meetingmsg)
exten => 849,n,Hangup

The following happens if I dial it from a SIP handset

  == Using SIP RTP CoS mark 5
    -- Executing [849 at smallanimals:1] Playback("SIP/812-00000074",
"custom/ceh-meetingmsg") in new stack
    -- <SIP/812-00000074> Playing 'custom/ceh-meetingmsg.gsm'
(language 'en')
    -- Executing [849 at smallanimals:2] Hangup("SIP/812-00000074",
"") in new stack
  == Spawn extension (smallanimals, 849, 2) exited non-zero on
'SIP/812-00000074'

The scenario is during the day, if my client has a staff meeting, they simply
turn on call forwarding on the reception phone to this extension. In the past,
the audio would start as soon as the caller dials in.

After upgrading to Asterisk 1.6, we simply get no audio until the dialplan
finishes. On the Asterisk console, I can see that the sound file is indeed
playing, but we can't hear it. This happens if I am dialing the from a SIP
extension on the phone system, or if I dial in from the public phone system.

 == Using SIP RTP CoS mark 5
    -- Executing [812 at smallanimals:1] Dial("SIP/811-00000046",
"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 SIP/811-00000046 to 'Local/849 at smallanimals'
(thanks to SIP/812-00000047)
    -- Executing [849 at smallanimals:1] Playback("Local/849 at
smallanimals-b5dd;2", "custom/ceh-meetingmsg") in new stack
    -- <Local/849 at smallanimals-b5dd;2> Playing
'custom/ceh-meetingmsg.gsm' (language 'en')

I have tried so many things that I have lost count, and I humbly ask the
collective intelligence of the Asterisk community for assistance.

Many thanks

aF

Hi!
> After upgrading to Asterisk 1.6, we simply get no audio until the dialplan
> finishes. On the Asterisk console, I can see that the sound file is indeed
> playing, but we can't hear it. [...]
> 
> I have tried so many things that I have lost count, and I humbly ask the
> collective intelligence of the Asterisk community for assistance.
For a start: 

* upgarde to the current release of 1.6.2.x
* does that message play when you call it without a forward (302) on your 
admin phone?
* convert the .gsm prompt to a .wav or .alaw or .ulaw prompt and see if 
that improves matters
* do a "RTP debug" to see if there is any RTP being sent at all
* consider ChanSpy for listening in (although I doubt that'll help you)

Philipp

Hi Alex,


I'm new to this list, but I had this problem too, and I solved it looking at
the codecs the sip handsets use, and then I converted the voice prompts to
that codec just like Philipp said..

Ondrej

On Tue, Aug 31, 2010 at 10:04 AM, Alex Ferrara <alex at
receptiveit.com.au>wrote:
> Hi everyone,
>
> This is my first post to the list, although I am a long term user of
> Asterisk. I have recently found a problem that I just can't seem to
solve.
>
> I have a client that has an Ubuntu x64 based Asterisk server with and ISDN
> Dahdi interface and about 25 SIP handsets. Everything was working fine in
> Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have
> one single issue that I can't explain.
>
> I have an extension that if you call it, it will play a sound file and
> hangup. Pretty simple stuff. Below is the extensions.conf entry for this
> extension.
>
> exten => 849,1,Playback(custom/ceh-meetingmsg)
> exten => 849,n,Hangup
>
> The following happens if I dial it from a SIP handset
>
>  == Using SIP RTP CoS mark 5
>    -- Executing [849 at smallanimals:1]
Playback("SIP/812-00000074",
> "custom/ceh-meetingmsg") in new stack
>    -- <SIP/812-00000074> Playing 'custom/ceh-meetingmsg.gsm'
(language
> 'en')
>    -- Executing [849 at smallanimals:2]
Hangup("SIP/812-00000074", "") in new
> stack
>  == Spawn extension (smallanimals, 849, 2) exited non-zero on
> 'SIP/812-00000074'
>
> The scenario is during the day, if my client has a staff meeting, they
> simply turn on call forwarding on the reception phone to this extension. In
> the past, the audio would start as soon as the caller dials in.
>
> After upgrading to Asterisk 1.6, we simply get no audio until the dialplan
> finishes. On the Asterisk console, I can see that the sound file is indeed
> playing, but we can't hear it. This happens if I am dialing the from a
SIP
> extension on the phone system, or if I dial in from the public phone
system.
>
>  == Using SIP RTP CoS mark 5
>    -- Executing [812 at smallanimals:1] Dial("SIP/811-00000046",
> "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 SIP/811-00000046 to 'Local/849 at
smallanimals' (thanks
> to SIP/812-00000047)
>    -- Executing [849 at smallanimals:1] Playback("Local/849 at
smallanimals-b5dd;2",
> "custom/ceh-meetingmsg") in new stack
>    -- <Local/849 at smallanimals-b5dd;2> Playing
'custom/ceh-meetingmsg.gsm'
> (language 'en')
>
> I have tried so many things that I have lost count, and I humbly ask the
> collective intelligence of the Asterisk community for assistance.
>
> Many thanks
>
> aF
> --
> _____________________________________________________________________
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-- 
-- Ondrej ?kopek
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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

Hi Paul,

No cigar unfortunately. I also tried encoding the message as gsm, ulaw and alaw
with no success.

The ISDN interface is alaw and the SIP phones I was testing with are definately
alaw.

Not sure what to do from here. I might just need to bypass the issue using some
alternate way to put the message in front of the inbound dialplan logic on some
condition.

aF

On 01/09/2010, at 8:06 AM, Paul Belanger wrote:
> On Tue, Aug 31, 2010 at 5:50 PM, Alex Ferrara <alex at
receptiveit.com.au> wrote:
>> Hi Paul,
>> 
>> I tried adding Progress() to no avail. I still get no audio and below
is what comes up in the console.
>> 
> Try moving Progress() before the Dial().  If you Answer() the channel,
> do you have the same problem?
> 
> -- 
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
> 
> -- 
> _____________________________________________________________________
> -- 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