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<DIV dir=ltr align=left><SPAN class=986171914-16112006>Could you tell us which
are your plans about it? Schedule, licensing, pricing.. </SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<B>From:</B> asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] <B>On Behalf Of </B>Ramtin
Amin<BR><B>Sent:</B> jueves, 16 de noviembre de 2006 14:56<BR><B>To:</B>
Development discussion of video media support in Asterisk<BR><B>Subject:</B> RE:
[Asterisk-video] 3G to SIP problem<BR><BR></DIV>
<DIV></DIV>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></FONT></SPAN> </DIV>
<DIV><FONT face=Arial><FONT color=#0000ff>Uff!! </FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff>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></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN
class=EC_676143013-16112006>until someone finally develope it for asterisk at
last.. :)</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN
class=EC_676143013-16112006></SPAN></FONT></FONT><FONT face=Arial><FONT
color=#0000ff><SPAN
class=EC_676143013-16112006></SPAN></FONT></FONT> </DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN
class=EC_676143013-16112006>Greetings</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN
class=EC_676143013-16112006> Sergio</SPAN></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><SPAN
class=EC_676143013-16112006></SPAN></FONT></FONT> </DIV>
<DIV class=EC_OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR>
<FONT face=Tahoma><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>_______________________________________________
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target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-video</A><BR>--------------------------------------------------------------------------------------<BR>This
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