Hi Klaus,<br>
to record a live video conversation, you just need to insert some
pieces of code into chan_zap.c, i.e. in the part where chan_zap do
native bridging:<br>
I inserted several lines (e.g. tmp = write(ftrace, f->data, f->datalen); ) in line 3464 ( zt_bridge(), chan_zap.c).<br>
BTW, I did the H324M call briding in a v-1.2.4 Asterisk in the UK. <br>
<br>
regards,<br>
Zhuoqun Li <br>
<div><span class="gmail_quote"></span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>Date: Tue, 02 May 2006 11:17:14 +0200<br>From: Klaus Darilion <
<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>><br>Subject: Re: [asterisk-dev] Bridging two H324M calls<br>To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">
asterisk-dev@lists.digium.com</a>><br>Message-ID: <<a href="mailto:4457239A.3020808@pernau.at">4457239A.3020808@pernau.at</a>><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>zhuoqun Li wrote:
<br>> Hi,<br>> I have successfully bridged H324m calls through Asterisk (configured<br>> with a ISDN BRI interface).<br>> I have aslo dumped the live video conversation into a binary file.<br>> What I did is a "native channel bridge" and the dump functions are
<br>> inserted in the zt_bridge() in chan_zap.c.<br>> Hope this helps...<br><br>Can you share your code? E.g. post it on <a href="http://bugs.digium.com">bugs.digium.com</a><br><br>regards<br>klaus<br><br>><br>>
<br>> regards,<br>> Zhuoqun Li<br>><br>><br>><br>> ------------------------------<br>><br>> Message: 4<br>> Date: Fri, 28 Apr 2006 08:41:24 +0200<br>> From: Sergio Garc?a Murillo <
<a href="mailto:Sergio.Garcia@ydilo.com">Sergio.Garcia@ydilo.com</a><br>> <mailto:<a href="mailto:Sergio.Garcia@ydilo.com">Sergio.Garcia@ydilo.com</a>>><br>> Subject: RE: [asterisk-dev] Bridging two H324M calls
<br>> To: "Asterisk Developers Mailing List" <<br>> <a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a> <mailto:<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com
</a>>><br>> Message-ID:<br>> <<a href="mailto:F0B2E3841C70D644ADEB6E7578EA2D670232A5@lisa.people-com.com">F0B2E3841C70D644ADEB6E7578EA2D670232A5@lisa.people-com.com</a><br>> <mailto:
<a href="mailto:F0B2E3841C70D644ADEB6E7578EA2D670232A5@lisa.people-com.com">F0B2E3841C70D644ADEB6E7578EA2D670232A5@lisa.people-com.com</a>>><br>> Content-Type: text/plain; charset="iso-8859-1"
<br>><br>> Klaus Darilion wrote:<br>> > Hi Sergio!<br>> ><br>> > I've done this once and it worked (relaying). But I was not able to<br>> > record the sessions. When I tried the various "recording"
<br>> > applications the video call setup did not worked anymore. Relaying<br>> > was only successful when the bridging was done directly on the ISDN<br>> > card.<br>> ><br>> > I did this once with an old Asterisk version. With newer Asterisk
<br>> > version relaying is not possible anymore, as the zaptel code changes<br>> > some call parameters (from data calls to anything else ...).<br>> ><br>> > I tried to debug this once (message 0025307)
<br>> > <a href="http://bugs.digium.com/view.php?id=3891">http://bugs.digium.com/view.php?id=3891</a><br>> <<a href="http://bugs.digium.com/view.php?id=3891">http://bugs.digium.com/view.php?id=3891</a>
><br>> ><br>> > but did not received any help and could not solved it myself.<br>><br>> Could it be possible to modify the zapdump app in order to make to<br>> bridge two incoming calls through a pipe or socket?
<br>> It's probably easier than bridging two channels through asterisk.<br>> And it would not affect the H324M as the master-slave determination<br>> is done in H245.<br>><br>> Best regards<br>
> Sergio<br>><br>><br>><br>> ------------------------------------------------------------------------<br>><br>> _______________________________________________<br>> --Bandwidth and Colocation provided by
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