[Freeswitch-users] Bridging an incoming call to outbound gateway
Michael Collins
msc at freeswitch.org
Mon Nov 12 21:58:53 MSK 2012
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 at 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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> http://www.freeswitchsolutions.com
>
>
>
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--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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