[Asterisk-Users] play_and_record: No audio available

[Eric Wieling]

Don't allow=all.  Don't ever allow=all.  In fact don't even think about
allow=all.  Personally I would like the allow=all option REMOVED.

disallow=all
allow=ulaw

If that works then allow whatever codec you want instead of ulaw.

On Mon, 2004-02-09 at 14:01, Ryan Courtnage wrote:
> 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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