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:[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
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Go to http://www.digium.com/index.php?menu=documentation and look at
the "Unofficial Links" section. This section has links to a wide
variety of 3rd party Asterisk related pages. My page is the
"Asterisk Resource Pages".
BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
