hi Michael,
sip traces : http://pastebin.freeswitch.org/20209
gsm is ringing and no audio in both ways..
Best regards
You'll want to get the SIP trace and console debug log and put on pastebin.freeswitch.org with "FreeSWITCH Log" as the syntax highlighting.
-MCOn Sun, Nov 11, 2012 at 4:18 PM, Meditel <meditel@gmail.com> wrote:_________________________________________________________________________Hi,
I have actually 2 providers : p1 & p2
p1 is used as a DID only for incoming calls
p2 is used for making outgoing calls
I have 1 registered user : ext 1000
I have 2 GSM phones : GSM1 & GSM2
My dialplan is as :
<include>
<extension name="public_did">
<condition field="destination_number" expression="^(MY_P1_DID)$">
<action application="transfer" data="1000 XML default"/>
</condition>
</extension>
</include>
For incoming calls "1000" is ringing and i can make a call without any problem (GSM1 => P1_DID => ext 1000)
But whene trying to bridge the incoming call to p2 gateway (GSM1 => P1_DID => P2_GW => GSM2), i can see that GSM2 is ringing but whene i answer the call i can't hear any audio ...
This is my dialplan for the second case:
<include>
<extension name="public_did">
<condition field="destination_number" expression="^(MY_P1_DID)$">
<action application="bridge" data="sofia/gateway/p2_gw/GSM2_NUMBER"/>
</condition>
</extension>
</include>
Any help are welcome
best regards
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--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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