[Asterisk-Users] play_and_record: No audio available

[Ryan Courtnage]

There appears to be a problem with Asterisk negotiating a codec with my 
x-lite clients (both mac and windows).

All codecs were enable in the clients, and sip.conf contained:

      allow=all             ; Allow all codecs

Using 'show sip debug' on the * console would print the following when 
a call from the client was answered (by * voicemail):

------SNIP-----
     v=0
     o=2000 55640315 55640315 IN IP4 192.168.1.222
     s=X-Lite
     c=IN IP4 192.168.1.222
     t=0 0
     m=audio 8000 RTP/AVP 3 0 8 98 97 101
     a=rtpmap:0 pcmu/8000
     a=rtpmap:8 pcma/8000
     a=rtpmap:3 gsm/8000
     a=rtpmap:98 iLBC/8000
     a=rtpmap:97 speex/8000
     a=rtpmap:101 telephone-event/8000
     a=fmtp:101 0-15

     12 headers, 13 lines
     Using latest request as basis request
     Sending to 192.168.1.222 : 5060 (non-NAT)
     Found audio format UNKN
     Found audio format UNKN
     Found audio format ALAW
     Found audio format UNKN
     Found audio format UNKN
     Found audio format UNKN
     Found description format pcmu
     Found description format pcma
     Found description format gsm
     Found description format iLBC
     Found description format speex
     Found description format telephone-event
     Capabilities: us - 2147483647, them - 1550/0, combined - 1550
     Non-codec capabilities: us - 1, them - 1, combined - 1
     Looking for 2000 in from-sip
     list_route: hop: <sip:2000 at 192.168.1.222:5060>
     Transmitting (no NAT):
     SIP/2.0 100 Trying
     Via: SIP/2.0/UDP 
192.168.1.222:5060;rport;branch=z9hG4bK4C44954A5B3711D8A19E000393AFBA66
     From: XLite <sip:2000 at courtnage.ca>;tag=1188382427
     To: <sip:2000 at courtnage.ca>;tag=as31ceb95d
     Call-ID: 4A019EBE-5B37-11D8-A19E-000393AFBA66 at 192.168.1.222
     CSeq: 30827 INVITE
     User-Agent: Asterisk PBX
     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
     Contact: <sip:2000 at 192.168.1.101>
     Content-Length: 0


      to 192.168.1.222:5060
         -- Executing Dial("SIP/2000-0255", "SIP/2000|20") in new stack
     We're at 192.168.1.101 port 18164
     Answering with preferred capability 2147483647
     Answering with non-codec capability 1
------SNIP-----

The end result being the "No audio available on SIP" problem I reported 
below.

Modifying the allowed codec in sip.conf has solved the problem:

      allow=gsm             ; Allow all codecs


Any idea why the codec negotiation fails when all codecs are allowed?

Cheers,
Ryan

On 7-Feb-04, at 9:05 AM, Ryan Courtnage wrote:

> Hello all,
>
> I have just installed the latest Asterisk csv on Slackware 9.1.
>
> I've configured the system using the example config in OnLamp's 
> "Asterisk: A Bare-Bones VoIP Example" tutorial.  
> [http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1]
>
> Using X-Lite SIP client on Mac OS X
>
> When trying to leave a voicemail on one of the extensions, the 
> Asterisk console reports:
>
> Feb  7 23:26:17 WARNING[409618]: app_voicemail.c:1200 play_and_record: 
> No audio available on SIP/2001-dc12??
>
> and the resulting .wav file in /var/spool/asterisk is empty/invalid.
>
> I did find a thread (via google) that described someone with the same 
> problem, who rectified it by re-installing.  I've reinstalled, but to 
> no avail.
>
> Can someone please point me in the right direction to troubleshoot 
> this?
> Thanks
> Ryan
>
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