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<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=676143013-16112006>e</SPAN>n you'll need a H324M gateway in the
middle<SPAN class=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=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=676143013-16112006></SPAN></FONT></FONT></FONT><FONT face=Arial><FONT
color=#0000ff><FONT size=2><SPAN
class=676143013-16112006></SPAN></FONT></FONT></FONT> </DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><FONT size=2><SPAN
class=676143013-16112006>Greetings</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><FONT size=2><SPAN
class=676143013-16112006> Sergio</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><FONT size=2><SPAN
class=676143013-16112006> </SPAN></FONT></FONT></FONT></DIV>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
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<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=gmail_quote>On 11/16/06, <B class=gmail_sendername>Sergio
García Murillo</B> <<A
href="mailto:Sergio.Garcia@ydilo.com">Sergio.Garcia@ydilo.com</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; 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">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>>> >
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><BR>>> ><BR>>> ><BR>>>
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UNSUBSCRIBE or update options visit:<BR>>> <A
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----------------------------------------------------------------------
<BR>> --<BR>><BR>>
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--<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>_______________________________________________
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