[asterisk-users] Codes negotiation problems between Asterisk1.4beta2 and Aastra 480i

[Marco Mouta]

Have you ever tried allow=alaw&ulaw in the same line? just a tip...


On 10/6/06, Morten Isaksen <misak at misak.dk> wrote:
>
>
>
> On 10/6/06, Gareth Owen <gowen at aastra.com> wrote:
> >
> > Morten,
> >
> > Hmm, I haven't tried Asterisk 1.4 - I guess I should upgrade my system
> > to see what is going on.  Can you post the INVITE message that is being
> > rejected?
>
>
>
> This INVITE results in a 488 from the phone:
>
>
> INVITE sip:1014 at 192.168.10.100 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK42f78e77;rport
> From: "1011" < sip:1011 at 192.168.10.2>;tag=as3a35aa3a
> To: <sip:1014 at 192.168.10.100>
> Contact: < sip:1011 at 192.168.10.2>
> Call-ID: 15467e4462b5620e1e7155e96a5dc0ba at 192.168.10.2
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Fri, 06 Oct 2006 14:22:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 309
>
> v=0
> o=root 4746 4746 IN IP4 192.168.10.2
> s=session
> c=IN IP4 192.168.10.2
> t=0 0
> m=audio 10066 RTP/AVP 8 0 3 101
> a=rtpmap:8 PCMA/8000
> a=ptime:20
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:3 GSM/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=sendrecv
>
> And this INVITE works (only alaw is enabled):
>
> INVITE sip:1014 at 192.168.10.100 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK3c04692a;rport
> From: "1011" < sip:1011 at 192.168.10.2>;tag=as39cd0724
> To: <sip:1014 at 192.168.10.100>
> Contact: < sip:1011 at 192.168.10.2>
> Call-ID: 32a8f09a785b36cf5e8b6ba02b5afb00 at 192.168.10.2
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Fri, 06 Oct 2006 14:23:51 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 238
>
> v=0
> o=root 4762 4762 IN IP4 192.168.10.2
> s=session
> c=IN IP4 192.168.10.2
> t=0 0
> m=audio 10042 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=sendrecv
>
> Also, I know we've fixed a number of SDP related issues in 1.4.1, so if
> > you haven't already you might want to try the 1.4.1 beta.  Info on how
> > to get the beta is available here:
> >
> > http://groups.google.com/group/Aastra-480i-Users/browse_frm/thread/8f6f0f3419ef396d
> >
>
>
>
> I will try that and report back here.
>
>
> --
> Morten Isaksen
> http://www.misak.dk/blog/
>
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-- 
Com os melhores cumprimentos,

Marco Mouta
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