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.
-MC
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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