Yes. Klaus. In this case the Asterisk or the IP PBX should do that. Unfortunately it is still not so advanced :( Hopefully in the future. <br><br><br><br><br><div><span class="gmail_quote">On 11/20/06, <b class="gmail_sendername">
Klaus Darilion</b> <<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</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;">
Yes. looks like the gateway does not demultiplex the video and audio but<br>just forwards the raw ISDN B channel.<br><br>regards<br>klaus<br><br>Nikolay Milovanov wrote:<br>> Hi Sergio,<br>><br>> I guess that's because of the clear channel. For me that means that both
<br>> are<br>> encoded in it.<br>><br>> Thanks for the help.<br>><br>> Niko<br>><br>><br>> On 11/16/06, Sergio García Murillo <<a href="mailto:Sergio.Garcia@ydilo.com">Sergio.Garcia@ydilo.com
</a>> wrote:<br>>><br>>> Appart of that, no video media is specified in the sdp.<br>>><br>>> Greetings<br>>> Sergio<br>>><br>>> -----Original Message-----<br>>> From:
<a href="mailto:asterisk-video-bounces@lists.digium.com">asterisk-video-bounces@lists.digium.com</a> [mailto:<br>>> <a href="mailto:asterisk-video-bounces@lists.digium.com">asterisk-video-bounces@lists.digium.com</a>
] On Behalf Of Klaus Darilion<br>>> Sent: jueves, 16 de noviembre de 2006 12:02<br>>> To: Development discussion of video media support in Asterisk<br>>> Subject: Re: [Asterisk-video] 3G to SIP problem<br>
>><br>>> Hi!<br>>><br>>> I guess these lines are irrelevant, because the m line offers only codec<br>>> 125.<br>>><br>>> regards<br>>> klaus<br>>><br>>> Nikolay Milovanov wrote:
<br>>> > Thanks Andrey,<br>>> ><br>>> > the call is actually a video call and the video is comming from the<br>>> > softswitch as unrestricted digital G.nX64/8000). I guess asterisk in<br>
>> > general is not supporting unrestricted digital.<br>>> ><br>>> > Could somebody explain to me what is that means:<br>>> ><br>>> > a=X-cpar: a=rtpmap:100 X-NSE/8000<br>>> > a=X-cpar: a=fmtp:100 192-194,200-202
<br>>> ><br>>> ><br>>> > BR,<br>>> > Niko<br>>> ><br>>> > On 11/16/06, Andrey Kuprianov < <a href="mailto:andrey.kouprianov@gmail.com">andrey.kouprianov@gmail.com</a>
> wrote:<br>>> >><br>>> >> Yup,<br>>> >><br>>> >> It looks like Asterisk does not support your codec. That's what your<br>>> >> SDP<br>>> >> 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">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<br>>> >> > see<br>>> >> from<br>>> >> > the trace Asterisk is not supporting the clear chanel codec
<br>>> >> (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<br>>> >> > my
<br>>> >> 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
<br>>> 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 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">my.domain.com:5060</a> ---> INVITE<br>>> >> > <a href="mailto:sip:32515901@172.18.10.100">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">my.domain.com:5060</a><br>>> >> > ;branch=z9hG4bKterm-30-myphone-32515901-17145<br>>> >> > From: myphone <
<a href="mailto:sip:myphone@my.domain.com">sip:myphone@my.domain.com</a>;user=phone>;tag=1763495500<br>>> >> > To: 32515901 < <a href="mailto:sip:32515901@172.18.10.100">sip:32515901@172.18.10.100</a>
;user=phone><br>>> >> > Call-ID: <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com">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">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">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">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">85.118.195.7:18010</a> Found description<br>>> >> > format G.nX64 for ID 125 Found description format G.nX64 for ID 125<br>>> >> > Capabilities: us - 0x1c0008 (alaw|h261|h263|h263p), peer -
<br>>> >> > audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)<br>>> >> > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -<br>>> >> > 0x0 (nothing), combined - 0x0 (nothing) [Nov 15 23:15:53]
<br>>> >> > NOTICE[22446]: chan_sip.c:4996 process_sdp: No<br>>> >> compatible<br>>> >> > codecs, not accepting this offer!<br>>> >> ><br>>> >> > <--- Reliably Transmitting (no NAT) to
<a href="http://my.domain.com:5060">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">my.domain.com:5060
</a><br>>> >> > ;branch=z9hG4bKterm-30-myphone-32515901-17145;received=<br>>> >> > <a href="http://my.domain.com">my.domain.com</a><br>>> >> > From: myphone <<br>>> >> >
<a href="mailto:sip:myphone@my.domain.com">sip:myphone@my.domain.com</a>;user=phone>;tag=1763495500<br>>> >> > To: 32515901 <<br>>> >> > <a href="mailto:sip:32515901@172.18.10.100">sip:32515901@172.18.10.100
</a>;user=phone>;tag=as11b984c5<br>>> >> > Call-ID: <a href="mailto:3f7a5361-3057ba40-5f7fc171-25@my.domain.com">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>>> >> > _______________________________________________
<br>>> >> > --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>>> >> ><br>>> >> > asterisk-video mailing list<br>>> >> > To UNSUBSCRIBE or update options visit:
<br>>> >> ><br>>> >> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-video">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br>>> >> ><br>>> >> >
<br>>> >> ><br>>> >> _______________________________________________<br>>> >> --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>>> >>
<br>>> >> asterisk-video mailing list<br>>> >> To UNSUBSCRIBE or update options visit:<br>>> >> <a href="http://lists.digium.com/mailman/listinfo/asterisk-video">http://lists.digium.com/mailman/listinfo/asterisk-video
</a><br>>> >><br>>> ><br>>> ><br>>> > ----------------------------------------------------------------------<br>>> > --<br>>> ><br>>> > _______________________________________________
<br>>> > --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>>> ><br>>> > asterisk-video mailing list<br>>> > To UNSUBSCRIBE or update options visit:
<br>>> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-video">http://lists.digium.com/mailman/listinfo/asterisk-video</a><br>>><br>>><br>>> --<br>>> Klaus Darilion<br>>>
<a href="http://nic.at">nic.at</a><br>>><br>>> _______________________________________________<br>>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>>><br>
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