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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I’ve found that the Asterisk SIP implementation does not
work well with Polycom video. You need to use the video branch, and add code to
gain support for fmtp SIP messages in order to get the Asterisk SIP channel to
recognize video properly. There was a hack that was posted some time ago that I’m
using to get this to work in my own development code. An associated comment
indicated that some thought needed to be applied to make the hack something that
could be checked into the repository. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>So, my specific request is that FMTP messages get added to
chan_sip.c as well as add in some of the changes from the video branch to
support Polycom video. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Jonathan<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal align=left style='text-align:left'><b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>
asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] <b>On Behalf Of </b>Tim King<br>
<b>Sent:</b> Thursday, September 17, 2009 2:23 AM<br>
<b>To:</b> Development discussion of video media support in Asterisk<br>
<b>Subject:</b> Re: [Asterisk-video] Video with Polycom VVX1500<o:p></o:p></span></p>
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</div>
<p class=MsoNormal align=left style='text-align:left'><o:p> </o:p></p>
<p class=MsoNormal>First off, try Asterisk 1.6.0 and only allow H.264 codec and
whatever audio codec you want. Ensure videosupport=yes. Then see what works and
doesn't. Video voicemail should be OK but may not be sync'ed very well. PolyCom
to PolyCom should work OK. PolyCom to Grandstream may work, depending on who
initiates the call. IVR usage should be irrelevant, but Queuing might be an
issue. Also it is vital that all phones are upgraded to latest firmware.<br>
<br>
Then come back and ask for something more specific to be fixed.<br>
<br>
johnny_xing wrote: <o:p></o:p></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Arial","sans-serif"'>Hi,
I understand there are some problems while integrating Polycom video phone with
asterisk and passing through is fine for Grandstream.</span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Arial","sans-serif"'> </span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Arial","sans-serif"'>I
would like to pay for any patch / customization service to asterisk so that I
could make video calls between Polycom VVX 1500, video recordings, voicemail,
and IVR.</span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Arial","sans-serif"'> </span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Arial","sans-serif"'>Please
contact me if anybody has a suitable solution.</span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Arial","sans-serif"'> </span><o:p></o:p></p>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks
& Best Regards,</span><o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Johnny
Xing</span><o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal align=left style='text-align:left'><span style='font-size:
12.0pt'><br>
<br>
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<br>
<o:p></o:p></span></p>
<pre><o:p> </o:p></pre><pre style='text-align:center'>
<hr size=4 width="90%" align=center>
</pre><pre><o:p> </o:p></pre><pre>_______________________________________________<o:p></o:p></pre><pre>--Bandwidth and Colocation Provided by <a
href="http://www.api-digital.com">http://www.api-digital.com</a>--<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>asterisk-video mailing list<o:p></o:p></pre><pre>To UNSUBSCRIBE or update options visit:<o:p></o:p></pre><pre> <a
href="http://lists.digium.com/mailman/listinfo/asterisk-video">http://lists.digium.com/mailman/listinfo/asterisk-video</a><o:p></o:p></pre></div>
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