<html>
<head>
<style>
P
{
margin:0px;
padding:0px
}
body
{
FONT-SIZE: 10pt;
FONT-FAMILY:Tahoma
}
</style>
</head>
<body>Hopefully, I'll release soon :)<BR>
<BR><BR><BR><BR><BR> <BR>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #008080 2px solid; MARGIN-RIGHT: 0px">
<HR id=EC_stopSpelling>
Subject: RE: [Asterisk-video] 3G to SIP problem<BR>Date: Thu, 16 Nov 2006 14:34:26 +0100<BR>From: Sergio.Garcia@ydilo.com<BR>To: asterisk-video@lists.digium.com<BR><BR>
<META content="Microsoft SafeHTML" name=Generator>
<DIV dir=ltr align=left><SPAN class=EC_676143013-16112006><FONT face=Arial color=#0000ff 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 class=EC_676143013-16112006>e</SPAN>n you'll need a H324M gateway in the middle<SPAN class=EC_676143013-16112006> 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 class=EC_676143013-16112006>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 class=EC_676143013-16112006></SPAN></FONT></FONT></FONT><FONT face=Arial><FONT color=#0000ff><FONT size=2><SPAN class=EC_676143013-16112006></SPAN></FONT></FONT></FONT> </DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><FONT size=2><SPAN class=EC_676143013-16112006>Greetings</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><FONT size=2><SPAN class=EC_676143013-16112006> Sergio</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><FONT size=2><SPAN class=EC_676143013-16112006> </SPAN></FONT></FONT></FONT></DIV>
<DIV class=EC_OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR>
<FONT face=Tahoma size=2><B>From:</B> asterisk-video-bounces@lists.digium.com [mailto:asterisk-video-bounces@lists.digium.com] <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 class=EC_gmail_quote>On 11/16/06, <B class=EC_gmail_sendername>Sergio García Murillo</B> <<A href="mailto:Sergio.Garcia@ydilo.com">Sergio.Garcia@ydilo.com</A>> wrote:</SPAN>
<BLOCKQUOTE class=EC_gmail_quote style="PADDING-LEFT: 1ex; BORDER-LEFT: rgb(204,204,204) 1px solid">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:<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 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 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>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/" target=_blank>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://my.domain.com:5060/" target=_blank>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>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/" target=_blank>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>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>my.domain.com:5060</A><BR>>> > ;branch=z9hG4bKterm-30-myphone-32515901-17145;received= <BR>>> > <A href="http://my.domain.com/" target=_blank>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/" target=_blank>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>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>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" target=_blank>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/" target=_blank>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" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-video</A><BR><BR><BR>--<BR>Klaus Darilion<BR><A href="http://nic.at/" target=_blank>nic.at</A><BR><BR>_______________________________________________ <BR>--Bandwidth and Colocation provided by <A href="http://easynews.com/" target=_blank>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" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-video</A><BR>--------------------------------------------------------------------------------------<BR>This message and any files transmitted with it are confidential and intended solely <BR>for the use of the individual or entity to whom they are addressed. No confidentiality<BR>or privilege is waived or lost by any wrong transmission.<BR>If you have received this message in error, please immediately destroy it and kindly <BR>notify the sender by reply email.<BR>You must not, directly or indirectly, use, disclose, distribute, print, or copy any<BR>part of this message if you are not the intended recipient. Opinions, conclusions and<BR>other information in this message that do not relate to the official business of <BR>Ydilo Advanced Voice Solutions, S.A. shall be understood as neither given nor endorsed by it.<BR>--------------------------------------------------------------------------------------<BR>_______________________________________________ <BR>--Bandwidth and Colocation provided by <A href="http://easynews.com/" target=_blank>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" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-video</A><BR></BLOCKQUOTE></DIV><BR>
<DIV> </DIV></BLOCKQUOTE><br /><hr />Exprimez-vous en direct avec Windows Live Messenger ! <a href='http://imagine-msn.com/messenger/launch80/default.aspx?locale=fr-fr&source=joinmsncom/messenger' target='_new'>Windows Live Messenger !</a></body>
</html>