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<TITLE>Re: [Freeswitch-users] Freetdm error</TITLE>
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<FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>Here you go<BR>
<BR>
<a href="http://pastebin.freeswitch.org/13057">http://pastebin.freeswitch.org/13057</a><BR>
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Thanks<BR>
<BR>
<BR>
On 5/26/10 2:39 PM, "Moises Silva" <<a href="moises.silva@gmail.com">moises.silva@gmail.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>Pastebin the debug output of the FreeSWITCH log.<BR>
<BR>
Moises Silva<BR>
Senior Software Engineer<BR>
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada<BR>
t. 1 905 474 1990 x 128 | e. <a href="moy@sangoma.com">moy@sangoma.com</a><BR>
<BR>
<BR>
On Wed, May 26, 2010 at 5:12 PM, Sean Holt <<a href="sean@obscuradigital.com">sean@obscuradigital.com</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>Hello all,<BR>
<BR>
I’ve been hacking at this issue for the last 3 days and I can’t get beyond this error when attempting to call out through my sangoma card.<BR>
<BR>
SIP/2.0 503 Service Unavailable<BR>
Via: SIP/2.0/UDP 192.168.100.6:12026;branch=z9hG4bK-d8754z-77564c04563bac12-1---d8754z-;rport=12026<BR>
From: "Sean Holt"<sip:<a href="2028@172.16.15.10">2028@172.16.15.10</a> <<a href="http://2028@172.16.15.10">http://2028@172.16.15.10</a>> >;tag=25dacb3d<BR>
To: <sip:<a href="5102079553@172.16.15.10">5102079553@172.16.15.10</a> <<a href="http://5102079553@172.16.15.10">http://5102079553@172.16.15.10</a>> >;tag=4St9j3N9yXr8g<BR>
Call-ID: NWViZTI0ODljMmY1YTM5ZDRiZmQzNTUwYzgwNWVkMDQ.<BR>
CSeq: 1 INVITE<BR>
User-Agent: Configured by TCAPI<BR>
Accept: application/sdp<BR>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<BR>
Supported: precondition, path, replaces<BR>
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer<BR>
Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION"<BR>
Content-Length: 0<BR>
Remote-Party-ID: "5102079553" <sip:<a href="5102079553@172.16.15.10">5102079553@172.16.15.10</a> <<a href="http://5102079553@172.16.15.10">http://5102079553@172.16.15.10</a>> >;party=calling;privacy=off;screen=no<BR>
<BR>
Here’s my configuration:<BR>
I’m trying to call out from my internal voip phone through my sangoma card b600. My phone line is plugged into the FXO port. Module (mod_freetdm) is successfully loaded<BR>
Dialplan:<BR>
<extension name="main_number_53"><BR>
<condition field="destination_number" expression="^(\d{11})$|^(\d{10})$$"><BR>
<action application="bridge" data="freetdm/2/2/$1"/><BR>
</condition><BR>
</extension> <BR>
<BR>
Freetdm.conf file<BR>
[span wanpipe wp1]<BR>
name => FreeTDM<BR>
number => 1<BR>
trunk_type => FXS<BR>
fxs-channel => 1:5<BR>
<BR>
[span wanpipe wp2]<BR>
name => FreeTDM<BR>
number => 2<BR>
trunk_type => FXO<BR>
fxo-channel => 1:1-4<BR>
<BR>
Freetdm.conf.xml file<BR>
<configuration name="freetdm.conf" description="FreeTDM Configuration"><BR>
<settings><BR>
<param name="debug" value="9"/><BR>
<!--<param name="hold-music" value="$${moh_uri}"/>--><BR>
</settings><BR>
<analog_spans><BR>
<span name="wp1"><BR>
<param name="context" value="context_1"/><BR>
<param name="dialplan" value="XML"/><BR>
</span><BR>
<span name="wp2"><BR>
<param name="context" value="context_1"/><BR>
<param name="dialplan" value="XML"/><BR>
</span><BR>
</analog_spans><BR>
</configuration><BR>
<BR>
Let me know if I need to provide additional info<BR>
<BR>
Thanks<BR>
<FONT COLOR="#888888">Sean</FONT> <BR>
<BR>
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