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Hi,<br>
This is the scenario <br>
I have the SJlabs phone with g711ulaw active and the rest disabled.<br>
I have * with chan_h323<br>
I have a Quintum DX that supports, g723.1 , g729AB, ulaw and alaw.<br>
<br>
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.<br>
I tried all possible codecs ulaw, alaw, g723 and g729 always the same
result.<br>
The phone rings but as soon as answered it dissconnects.<br>
<br>
<p class="MsoNormal"><font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;">The debug shows<o:p></o:p></span></font></p>
<p class="MsoNormal"><font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;"><span style=""> </span>--
Executing <span class="GramE">Dial(</span>"SIP/sj1-4ff7",
"H323/0797617729") in new stack<o:p></o:p></span></font></p>
<p class="MsoNormal"><font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;"><span style=""> </span>--
Called 0797617729<o:p></o:p></span></font></p>
<p class="MsoNormal"><font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;"><span style=""> </span>--
H323/0797617729 is ringing<o:p></o:p></span></font></p>
<p class="MsoNormal"><font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;"><span style=""> </span>--
H323/0797617729 answered SIP/sj1-4ff7<o:p></o:p></span></font></p>
<p class="MsoNormal"><font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;"><span style=""> </span>==
Spawn extension (default, 0797617729, 1) exited non-zero on
'SIP/sj1-4ff7'<o:p></o:p></span></font></p>
<p class="MsoNormal"><font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;"><span style=""> </span><span
style=""> </span>-- Executing <span class="GramE">Dial(</span>"SIP/sj1-4ff7",
"H323/h") in new stack<o:p></o:p></span></font></p>
<p class="MsoNormal"><font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;"><span style=""> </span>--
Called h<o:p></o:p></span></font></p>
<p class="MsoNormal"><font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;"><span style=""> </span>==
Spawn extension (default, h, 1) exited non-zero on 'SIP/sj1-4ff7'<o:p></o:p></span></font></p>
<p class="MsoNormal"><font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;"><o:p> </o:p>The first
Dial is normal but the 2<sup>nd</sup> <span class="GramE">Dial <span
style=""> </span>“</span>Executing Dial("SIP/sj1-4ff7", "H323/h") in
new stack”</span></font></p>
<font size="2" face="Arial"><span
style="font-size: 10pt; font-family: Arial;">Where <span class="GramE">do
that</span> come from?</span></font><br>
<br>
PLEASE someone HELP!<br>
<br>
The * have the config below<br>
<br>
In extensions.conf I use <br>
[globals]<br>
[default]<br>
exten => _.,1,Dial(H323/${EXTEN})<br>
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;<br>
in H323.conf I use<br>
[general]<br>
port = 1720<br>
bindaddr = 195.216.65.212<br>
tos=lowdelay<br>
allow=all<br>
gatekeeper = 195.216.65.215<br>
AllowGKRouted = yes<br>
context=default<br>
[AST37]<br>
type=h323<br>
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;<br>
<br>
In SIP.conf I have<br>
[general]<br>
port=5060 <br>
bindaddr=xxx.xxx.xxx.xxx<br>
<br>
[sj1]<br>
type=friend <br>
context=default<br>
host=dynamic <br>
disallow=all <br>
allow=all <br>
username=sj1 <br>
secret=sj1<br>
<br>
[sj2]<br>
type=friend <br>
context=default<br>
host=xxx.xxx.xxx.xxx <br>
allow=ulaw <br>
username=sj1 <br>
secret=sj1<br>
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;<br>
<br>
administrator tootai wrote:<br>
<blockquote type="cite" cite="mid411BD5A7.9020408@tootai.net">Krystian
Filiks a écrit : <br>
<br>
<blockquote type="cite">Like you suggested I tried the g.711 now and
got the same, The called number rings but when answered it dropped. <br>
I connect to a Quintum Tenor DX. <br>
<br>
The part I'm curious about is 6:53.985
Transactor:8140ee8 h323trans.cxx(678) Trans admissionRequest
rejected: requestDenied <br>
6:53.988 H225 Caller:8159198 h323.cxx(2660) H225
Gatekeeper refused admission: requestDenied <br>
6:53.959 H225 Caller:813c890 h323pdu.cxx(1159) H225
Read error (0): <br>
<br>
Does anyone have a clue where to look for the problem? <br>
<br>
here is a trace, <br>
-- Executing Dial("SIP/sj1-a7e9", <a class="moz-txt-link-rfc2396E" href="mailto:H323/h@195.216.65.215">"H323/h@195.216.65.215"</a>) in new stack <br>
Allowed Codecs: <br>
Table: <br>
G.711-uLaw-64k{sw} <1> <br>
Set: <br>
0: <br>
0: <br>
G.711-uLaw-64k{sw} <1> <br>
<br>
-- Making call to <a class="moz-txt-link-abbreviated" href="mailto:h@195.216.65.215">h@195.216.65.215</a> using gatekeeper. <br>
channelsOpen = 1 <br>
channelsOpen = 0 <br>
6:53.959 H225 Caller:813c890 h323pdu.cxx(1159) H225
Read error (0): <br>
== New H.323 Connection created. <br>
-- sj1 is calling host <a class="moz-txt-link-abbreviated" href="mailto:h@195.216.65.215">h@195.216.65.215</a> <br>
-- Call token is ip$localhost/31767 <br>
-- Call reference is 31767 <br>
-- Called <a class="moz-txt-link-abbreviated" href="mailto:h@195.216.65.215">h@195.216.65.215</a> <br>
-- ClearCall: Request to clear call with token
ip$localhost/31767 <br>
-- Sending RELEASE COMPLETE <br>
== Spawn extension (default, h, 1) exited non-zero on 'SIP/sj1-a7e9' <br>
6:53.985 Transactor:8140ee8 h323trans.cxx(678) Trans
admissionRequest rejected: requestDenied <br>
6:53.988 H225 Caller:8159198 h323.cxx(2660) H225
Gatekeeper refused admission: requestDenied <br>
6:54.004 H323 Cleaner h323.cxx(1542) H323
Connection ip$localhost/31766 terminated. <br>
-- Call with Tenor Gateway [195.216.65.215] completed
(EndedByLocalUser) <br>
== H.323 Connection deleted. <br>
<br>
<br>
</blockquote>
What's <a class="moz-txt-link-abbreviated" href="mailto:h@195.216.65.215">h@195.216.65.215</a>? If you are register to GK
H323/<EndPoint> is enough. I don't understand your h EP. And
also request denied seems that you need to register. But I don't know
how work Quintum, maybe I'm wrong. <br>
<br>
<blockquote type="cite">-----Original Message----- <br>
From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-admin@lists.digium.com">asterisk-users-admin@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-admin@lists.digium.com">mailto:asterisk-users-admin@lists.digium.com</a>] On Behalf Of
administrator tootai <br>
Sent: Thursday, August 12, 2004 6:02 PM <br>
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a> <br>
Subject: Re: [Asterisk-Users] H323 call dropped when answered <br>
<br>
Krystian.Filiks a écrit : <br>
<br>
<br>
<br>
<blockquote type="cite">Hello anyone that can help me here?? please
read below. <br>
[...] <br>
<br>
<br>
<blockquote type="cite">Allowed Codecs: <br>
<br>
Table: <br>
<br>
G.723.1{sw} <1> <br>
<br>
Set: <br>
<br>
0: <br>
<br>
0: <br>
<br>
G.723.1{sw} <1> <br>
<br>
<br>
</blockquote>
</blockquote>
G.723.1 is not a codec in * Use g711 instead. If your GK is GnuGK, see
his debug logs. Also, run asteriks in debug mode and check logs in full
file. <br>
<br>
<br>
<br>
</blockquote>
<br>
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</blockquote>
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