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did you ever get the chan_h323 working?<br>
<br>
Asterisk . wrote:<br>
<blockquote type="cite"
cite="mid20040813093005.70096.qmail@web90010.mail.scd.yahoo.com">
<pre wrap="">Hello,
I posted mails regarding the same issue just 1 week back (Sub: H323 Call Dropping, and then OH323
Call Dropping). I was trying to connect from my CISCO ATA to Nextone using Asterisk and the calls
were dropping immediately after the calls were answered. I used chan_h323 for about 4 days, but
could not make it. Then i changed to chan_oh323 and finally got it working after trying that for
another 3 days using g729 codec. I also had issues with g711, and g723. I think your problem is
codec. Try SIP debug also and see the packets. If nothing works, try using chan_oh323.
</pre>
<blockquote type="cite">
<pre wrap="">"Executing Dial("SIP/sj1-4ff7", "H323/h") in new stack"
</pre>
</blockquote>
<pre wrap=""><!---->
This one is really frustrating. I had no clue when it happened to me, and i had no hangup command
in my dialplan.
Good Luck!
Girish
--- "Krystian.Filiks" <a class="moz-txt-link-rfc2396E" href="mailto:Krystian.Filiks@kfiliks.com"><Krystian.Filiks@kfiliks.com></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hi,
This is the scenario
I have the SJlabs phone with g711ulaw active and the rest disabled.
I have * with chan_h323
I have a Quintum DX that supports, g723.1 , g729AB, ulaw and alaw.
The problem is that, it does not mather what I put in the
extensions.conf I have tried all possible ways that I so far could find
using the net.
I tried all possible codecs ulaw, alaw, g723 and g729 always the same
result.
The phone rings but as soon as answered it dissconnects.
</pre>
</blockquote>
<pre wrap=""><!---->
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</pre>
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