Bridging an incoming call to outbound gateway
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
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 On 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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hi Michael, sip traces : http://pastebin.freeswitch.org/20209 gsm is ringing and no audio in both ways.. Best regards 2012/11/12 Michael Collins <msc@freeswitch.org>
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
On 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
_________________________________________________________________________ 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 FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ 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 FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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Meditel
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Michael Collins