[asterisk-dev] Bridging two H324M calls

zhuoqun Li zhuoqunli at gmail.com
Wed May 3 06:08:36 MST 2006


Hi Sergio,
Could you leave your email address here so I can email my trace files  to
you?

regards,
Zhuoqun


> Message: 4
> Date: Wed, 3 May 2006 08:45:04 +0200
> From: Sergio Garc?a Murillo <Sergio.Garcia at ydilo.com>
> Subject: RE: [asterisk-dev] Bridging two H324M calls
> To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
> Message-ID:
>         <F0B2E3841C70D644ADEB6E7578EA2D670232BC at lisa.people-com.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Could you share the dumped files at least?
> They would be very usefull..
>
> ________________________________
>
> From: asterisk-dev-bounces at lists.digium.com [mailto:
> asterisk-dev-bounces at lists.digium.com] On Behalf Of zhuoqun Li
> Sent: martes, 02 de mayo de 2006 18:19
> To: asterisk-dev at lists.digium.com
> Subject: Re: [asterisk-dev] Bridging two H324M calls
>
>
>
>
> Hi Klaus,
> 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:I inserted several lines (e.g. tmp = write(ftrace, f->data,
> f->datalen); ) in line 3464 ( zt_bridge(), chan_zap.c).
> BTW, I did the  H324M call briding in a v-1.2.4 Asterisk in the UK.
>
> regards,
> Zhuoqun Li
>
>
>
>
>
>
>                 Date: Tue, 02 May 2006 11:17:14 +0200
>                 From: Klaus Darilion < klaus.mailinglists at pernau.at>
>                 Subject: Re: [asterisk-dev] Bridging two H324M calls
>                 To: Asterisk Developers Mailing List <
> asterisk-dev at lists.digium.com>
>                 Message-ID: <4457239A.3020808 at pernau.at>
>                 Content-Type: text/plain; charset=ISO-8859-1;
> format=flowed
>
>                 zhuoqun Li wrote:
>                 >  Hi,
>                 >  I have successfully bridged H324m calls through
> Asterisk (configured
>                 > with a ISDN BRI interface).
>                 >  I have aslo dumped the live video conversation into a
> binary file.
>                 >  What I did is a "native channel bridge" and the dump
> functions are
>                 > inserted in the zt_bridge() in chan_zap.c.
>                 >  Hope this helps...
>
>                 Can you share your code? E.g. post it on bugs.digium.com
>
>                 regards
>                 klaus
>
>                 >
>                 >
>                 >  regards,
>                 >  Zhuoqun Li
>                 >
>                 >
>                 >
>                 >     ------------------------------
>                 >
>                 >     Message: 4
>                 >     Date: Fri, 28 Apr 2006 08:41:24 +0200
>                 >     From: Sergio Garc?a Murillo <
> Sergio.Garcia at ydilo.com
>                 >     <mailto: Sergio.Garcia at ydilo.com <mailto:
> Sergio.Garcia at ydilo.com> >>
>                 >     Subject: RE: [asterisk-dev] Bridging two H324M calls
>                 >     To: "Asterisk Developers Mailing List" <
>                 >     asterisk-dev at lists.digium.com <mailto:
> asterisk-dev at lists.digium.com >>
>                 >     Message-ID:
>                 >            <
> F0B2E3841C70D644ADEB6E7578EA2D670232A5 at lisa.people-com.com
>                 >     <mailto:
> F0B2E3841C70D644ADEB6E7578EA2D670232A5 at lisa.people-com.com>>
>                 >     Content-Type: text/plain;       charset="iso-8859-1"
>                 >
>                 >     Klaus Darilion wrote:
>                 >      > Hi Sergio!
>                 >      >
>                 >      > I've done this once and it worked (relaying). But
> I was not able to
>                 >      > record the sessions. When I tried the various
> "recording"
>                 >      > applications the video call setup did not worked
> anymore. Relaying
>                 >      > was only successful when the bridging was done
> directly on the ISDN
>                 >      > card.
>                 >      >
>                 >      > I did this once with an old Asterisk version.
> With newer Asterisk
>                 >      > version relaying is not possible anymore, as the
> zaptel code changes
>                 >      > some call parameters (from data calls to anything
> else ...).
>                 >      >
>                 >      > I tried to debug this once (message 0025307)
>                 >      > http://bugs.digium.com/view.php?id=3891
>                 >     < http://bugs.digium.com/view.php?id=3891 <
> http://bugs.digium.com/view.php?id=3891>  >
>                 >      >
>                 >      > but did not received any help and could not
> solved it myself.
>                 >
>                 >     Could it be possible to modify the zapdump app in
> order to make to
>                 >     bridge two incoming calls through a pipe or socket?
>                 >     It's probably easier than bridging two channels
> through asterisk.
>                 >     And it would not affect the H324M as the
> master-slave determination
>                 >     is done in H245.
>                 >
>                 >     Best regards
>                 >     Sergio
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20060503/64d0989e/attachment.htm


More information about the asterisk-dev mailing list