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 0-1512 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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=1188382427
To: <sip:[EMAIL PROTECTED]>;tag=as31ceb95d
Call-ID: [EMAIL PROTECTED]
CSeq: 30827 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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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