Thanks Andrey, <br>
<br>
the call is actually a video call and the video is comming from the softswitch as unrestricted digital <span>G.nX64/8000). I guess asterisk in general is not supporting unrestricted digital.<br><br>Could somebody explain to me what is that means:
<br></span><br>a=X-cpar: a=rtpmap:100 X-NSE/8000<br>a=X-cpar: a=fmtp:100 192-194,200-202
<br><br><br><span>BR, <br>Niko<br></span><br><div><span class="gmail_quote">
On 11/16/06, <b class="gmail_sendername">Andrey Kuprianov</b> <<a href="mailto:andrey.kouprianov@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
andrey.kouprianov@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Yup,<br><br>It looks like Asterisk does not support your codec. That's what your SDP says:
<br><br>a=rtpmap:125 G.nX64/8000<br>a=rtpmap:101 /8000<br>a=rtpmap:100 /8000<br><br>And that's what you have in config file:<br><br>allow=alaw<br>allow=speex<br>allow=gsm<br><br>Try switching codec to one of these listed in your
sip.conf.<br><br><br>On 11/16/06, Nikolay Milovanov <<a href="mailto:n_milovanov@mail.bg" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">n_milovanov@mail.bg</a>> wrote:<br>> Hi Guys,<br>>
<br>> My Scenario is<br>><br>> 3Gphone -> (3G network provider)->(Softswitch Cisco
<br>> PGW)->SIP<-Asterisk<-SIP->SIP phone<br>><br>> I am using Asterisk 1.4 beta 3. I am calling from 3G to SIP. As I see from<br>> the trace Asterisk is not supporting the clear chanel codec (a=rtpmap:125
<br>> G.nX64/8000) used by the PGW.<br>><br>> Am I right or the problem is somewhere else? Please take a look of my config<br>> and the trace of the asterisk cli.<br>><br>><br>> sip.conf<br>><br>> [general]
<br>><br>> videosupport=yes<br>>
disallow=all ;
First disallow all codecs<br>>
allow=alaw ;
Allow codecs in order of preference<br>> allow=h263<br>> allow=h263p<br>> allow=h261<br>><br>> [32515901]<br>> type=friend<br>> secret=phone1<br>> host=dynamic<br>> ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
<br>> mailbox=1000 ; Mailbox for message waiting indicator<br>> context=sip<br>> videosupport=yes<br>> maxcallbitrate=128<br>> callerid= "test" <32515901><br>> allow=alaw<br>> allow=speex
<br>> allow=gsm<br>> allow=h261<br>> allow=h263<br>> allow=h263p<br>><br>><br>> Debug<br>><br>> <--- SIP read from <a href="http://my.domain.com:5060" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
my.domain.com:5060</a> ---><br>> INVITE
<a href="mailto:sip:32515901@172.18.10.100" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip:32515901@172.18.10.100</a>;user=phone SIP/2.0<br>> Via: SIP/2.0/UDP <a href="http://my.domain.com:5060" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
my.domain.com:5060</a><br>> ;branch=z9hG4bKterm-30-myphone-32515901-17145
<br>> From: myphone <<a href="mailto:sip:myphone@my.domain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip:myphone@my.domain.com</a>;user=phone>;tag=1763495500<br>> To: 32515901 <
<a href="mailto:sip:32515901@172.18.10.100" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip:32515901@172.18.10.100
</a>;user=phone><br>> Call-ID: <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">3f7a5361-3057ba40-5f7fc171-25@my.domain.com</a><br>
> CSeq: 1 INVITE<br>> Supported: timer<br>> Session-Expires: 1800
<br>> Min-SE: 1800<br>> Contact: <<a href="http://sip:myphone@my.domain.com:5060" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip:myphone@my.domain.com:5060</a>><br>> Allow:<br>
> INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE<br>> Max-Forwards: 70
<br>> Content-Type: application/sdp<br>> Content-Length: 317<br>><br>> v=0<br>> c=IN IP4 <a href="http://85.118.195.7" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">85.118.195.7</a>
<br>> m=audio 18010 RTP/AVP 125<br>> a=rtpmap:125 G.nX64/8000<br>
> a=X-pc-codec: 125 101 100<br>> a=rtpmap:125 G.nX64/8000<br>> a=rtpmap:101 /8000<br>> a=rtpmap:100 /8000<br>> a=X-sqn:0<br>> a=X-cap: 1 audio RTP/AVP 100<br>> a=X-cpar: a=rtpmap:100 X-NSE/8000<br>> a=X-cpar: a=fmtp:100 192-194,200-202
<br>> a=X-cap: 2 image udptl t38<br>><br>> <-------------><br>> --- (14 headers 13 lines) ---<br>> Using INVITE request as basis request -<br>> <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
3f7a5361-3057ba40-5f7fc171-25@my.domain.com</a><br>> Found peer 'test'<br>> Found RTP audio format 125<br>> Peer audio RTP is at port <a href="http://85.118.195.7:18010" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
85.118.195.7:18010</a><br>> Found description format
G.nX64 for ID 125<br>> Found description format G.nX64 for ID 125<br>> Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer - audio=0x0<br>> (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)<br>> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
<br>> (nothing), combined - 0x0 (nothing)<br>> [Nov 15 23:15:53] NOTICE[22446]: chan_sip.c:4996 process_sdp: No compatible<br>> codecs, not accepting this offer!<br>><br>> <--- Reliably Transmitting (no NAT) to
<a href="http://my.domain.com:5060" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">my.domain.com:5060</a> ---><br>> SIP/2.0 488 Not acceptable here<br>> Via: SIP/2.0/UDP <a href="http://my.domain.com:5060" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
my.domain.com:5060</a><br>> ;branch=z9hG4bKterm-30-myphone-32515901-17145;received=
<a href="http://my.domain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">my.domain.com</a><br>> From: myphone <<a href="mailto:sip:myphone@my.domain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
sip:myphone@my.domain.com</a>;user=phone>;tag=1763495500<br>> To: 32515901 <<a href="mailto:sip:32515901@172.18.10.100" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
sip:32515901@172.18.10.100</a>;user=phone>;tag=as11b984c5<br>> Call-ID: <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">3f7a5361-3057ba40-5f7fc171-25@my.domain.com
</a><br>> CSeq: 1 INVITE<br>> User-Agent: Asterisk PBX
<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>> Supported: replaces<br>> Content-Length: 0<br>><br>> Appreciate any help,<br>><br>> Niko<br>><br>><br>> _______________________________________________
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