You'll want to get the SIP trace and console debug log and put on <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a> with "FreeSWITCH Log" as the syntax highlighting.<br><br>-MC<br><br><div class="gmail_quote">
On Sun, Nov 11, 2012 at 4:18 PM, Meditel <span dir="ltr"><<a href="mailto:meditel@gmail.com" target="_blank">meditel@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,<br><br>I have actually 2 providers : p1 & p2<br>p1 is used as a DID only for incoming calls<br>p2 is used for making outgoing calls<br><br>I have 1 registered user : ext 1000<br><br>I have 2 GSM phones : GSM1 & GSM2<br>
<br>My dialplan is as :<br><include><br> <extension name="public_did"><br> <condition field="destination_number" expression="^(MY_P1_DID)$"><br> <action application="transfer" data="1000 XML default"/><br>
</condition><br> </extension><br></include><br><br>For incoming calls "1000" is ringing and i can make a call without any problem (GSM1 => P1_DID => ext 1000)<br>But whene trying to bridge the incoming call to p2 gateway (GSM1 => P1_DID => P2_GW => GSM2), i can see that GSM2 is ringing but whene i answer the call i can't hear any audio ...<br>
This is my dialplan for the second case:<br><include><br> <extension name="public_did"><br> <condition field="destination_number" expression="^(MY_P1_DID)$"><br> <action application="bridge" data="sofia/gateway/p2_gw/GSM2_NUMBER"/><br>
</condition><br> </extension><br></include><br><br>Any help are welcome<br>best regards<br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
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