Hopefully :)) Anyway if somebody is doing such a thing I could help at least in testing and probably documentation. <br><br><br>BR, <br>Niko<br><br><div><span class="gmail_quote">On 11/16/06, <b class="gmail_sendername">Ramtin Amin
</b> <<a href="mailto:keytwho@hotmail.com">keytwho@hotmail.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;">
<div>Hopefully, I'll release soon :)<br>
<br><br><br><br><br> <br>
<blockquote style="border-left: 2px solid rgb(0, 128, 128); padding-left: 5px; margin-left: 5px; margin-right: 0px;">
<hr>
Subject: RE: [Asterisk-video] 3G to SIP problem<br>Date: Thu, 16 Nov 2006 14:34:26 +0100<br>From: <a href="mailto:Sergio.Garcia@ydilo.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Sergio.Garcia@ydilo.com
</a><br>To: <a href="mailto:asterisk-video@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-video@lists.digium.com</a><div><span class="e" id="q_10ef10da48c53caf_1"><br><br>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div><font face="Arial"><font color="#0000ff"><font size="2">Uff!! </font></font></font></div>
<div><font face="Arial"><font color="#0000ff"><font size="2">Th<span>e</span>n you'll need a H324M gateway in the middle<span> to handle the negotiation and decode both streams, at least</span></font></font></font></div>
<div><font face="Arial"><font color="#0000ff"><font size="2"><span>until someone finally develope it for asterisk at last.. :)</span></font></font></font></div>
<div><font face="Arial"><font color="#0000ff"><font size="2"><span></span></font></font></font><font face="Arial"><font color="#0000ff"><font size="2"><span></span></font></font></font> </div>
<div><font face="Arial"><font color="#0000ff"><font size="2"><span>Greetings</span></font></font></font></div>
<div><font face="Arial"><font color="#0000ff"><font size="2"><span> Sergio</span></font></font></font></div>
<div><font face="Arial"><font color="#0000ff"><font size="2"><span> </span></font></font></font></div>
<div dir="ltr" align="left" lang="en-us">
<hr>
<font face="Tahoma" size="2"><b>From:</b> <a href="mailto:asterisk-video-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-video-bounces@lists.digium.com</a> [mailto:
<a href="mailto:asterisk-video-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-video-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Nikolay Milovanov<br><b>Sent:
</b> jueves, 16 de noviembre de 2006 13:51<br><b>To:</b> Development discussion of video media support in Asterisk<br><b>Subject:</b> Re: [Asterisk-video] 3G to SIP problem<br></font><br></div>
<div></div>Hi Sergio, <br><br>I guess that's because of the clear channel. For me that means that both are encoded in it. <br><br>Thanks for the help. <br><br>Niko<br><br><br>
<div><span>On 11/16/06, <b>Sergio García Murillo</b> <<a href="mailto:Sergio.Garcia@ydilo.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Sergio.Garcia@ydilo.com</a>> wrote:</span>
<blockquote style="border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
asterisk-video-bounces@lists.digium.com </a>[mailto:<a href="mailto:asterisk-video-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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 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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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" 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<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 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/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
my.domain.com:5060 </a>---> INVITE<br>>> > <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://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> 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/" 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= <br>>> > <a href="http://my.domain.com/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">my.domain.com
</a><br>>> > From: myphone <<br>>> > <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 <<br>>> > <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>>> > _______________________________________________ <br>>> > --Bandwidth and Colocation provided by <a href="http://easynews.com/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
http://lists.digium.com/mailman/listinfo/asterisk-video</a><br>>> ><br>>> ><br>>> ><br>>> _______________________________________________ <br>>> --Bandwidth and Colocation provided by
<a href="http://easynews.com/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Easynews.com</a> --<br>>><br>>> asterisk-video mailing list<br>>> To UNSUBSCRIBE or update options visit:
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<br>><br>><br>> ---------------------------------------------------------------------- <br>> --<br>><br>> _______________________________________________<br>> --Bandwidth and Colocation provided by <a href="http://easynews.com/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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http://lists.digium.com/mailman/listinfo/asterisk-video</a><br><br><br>--<br>Klaus Darilion<br><a href="http://nic.at/" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">nic.at</a><br><br>_______________________________________________
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