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[Freeswitch-users] problems with Faktortel (AU) and multiple DID's and extensions


 
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pawzlion at gmail.com
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PostPosted: Tue Apr 07, 2009 8:54 am    Post subject: [Freeswitch-users] problems with Faktortel (AU) and multiple Reply with quote

I have been trying to setup 2 DID's to route to 2 extensions but
whenever I try it, the second configured DID always routes to the first
extension.

In my public.xml I have the following:

<include>
<context name="public">

<extension name="DID 1">
<condition field="destination_number" expression="^(0746029000)$">
<action application="transfer" data="1000 XML default"/>
</condition>
</extension>

<extension name="DID 2">
<condition field="destination_number" expression="^(0746029002)$">
<action application="transfer" data="1001 XML default"/>
</condition>
</extension>

.... rest of file continues here ...


While in my default.xml I have this:

<include>
<context name="default">

<extension name="David">
<condition field="destination_number"
expression="^(0746029000)$" continue="on-true">
<action application="set" data="dialed_ext=$1" />
</condition>
<condition field="destination_number"
expression="^${caller_id_number}$">
<action application="set"
data="voicemail_authorized=${sip_authorized}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="check default
$${domain} ${dialed_ext}"/>
<anti-action application="ring_ready"/>
<anti-action application="set" data="call_timeout=10"/>
<anti-action application="set" data="hangup_after_bridge=true"/>
<anti-action application="set" data="continue_on_fail=true"/>
<anti-action application="bridge" data="USER/1000@$${domain}"/>
<anti-action application="answer"/>
<anti-action application="sleep" data="1000"/>
<anti-action application="voicemail" data="default
$${domain} ${dialed_ext}"/>
</condition>
</extension>

<extension name="Jake">
<condition field="destination_number"
expression="^(0746029001)$" continue="on-true">
<action application="set" data="dialed_ext=$2" />
</condition>
<condition field="destination_number"
expression="^${caller_id_number}$">
<action application="set"
data="voicemail_authorized=${sip_authorized}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="check default
$${domain} ${dialed_ext}"/>
<anti-action application="ring_ready"/>
<anti-action application="set" data="call_timeout=10"/>
<anti-action application="set" data="hangup_after_bridge=true"/>
<anti-action application="set" data="continue_on_fail=true"/>
<anti-action application="bridge" data="USER/1001@$${domain}"/>
<anti-action application="answer"/>
<anti-action application="sleep" data="1000"/>
<anti-action application="voicemail" data="default
$${domain} ${dialed_ext}"/>
</condition>
</extension>


<extension name="outgoing - voicepulse">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<action application="set"
data="effective_caller_id_number=0746029000"/>
<action application="bridge"
data="sofia/gateway/voicepulse/$1"/>
</condition>
</extension>

.. file continues here ...

I got my new friend swk to try and diagnose the problem and using ngrep
he found with ngrep that the incoming call to the second extension
looked like this:

U 203.161.130.133:5060 -> 10.0.0.12:5080
INVITE sip:gw+voicepulse@10.0.0.12:5080;transport=udp SIP/2.0..Via:
SIP/2.0/UDP 203.161.130.133:5060;branch=z9hG4bK75f53071;rport..From:
"0451282630" <sip:
0451282630@203.161.130.133>;tag=as555c5b50..To:
<sip:gw+voicepulse@124.254.94.41:5080;transport=udp>..Contact:
<sip:0451282630@203.161.130.133>..Call-ID: 7
47befb63a2def723e6796294853cc22@203.161.130.133..CSeq: 102
INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 07 Apr
2009 08:18:23 GMT..Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY..Supported: replaces..Content-Type:
application/sdp..Content-Length: 290....v=0..o=root 1244 12
44 IN IP4 203.161.130.133..s=session..c=IN IP4 203.161.130.133..t=0
0..m=audio 13806 RTP/AVP 18 3 101..a=rtpmap:18 G729/8000..a=fmtp:18
annexb=no..a=rtpmap
:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..

He says that the INVITE line should have a DNIS (not sure what that is)
in that field to indicate which number to route it to but that for some
reason, my provider (Faktortel in Australia) is not supplying that
information.

Does anyone know whether the problem is really at my provider's end or
at my end, and if it's at my end, where ?

thanks,
pawz

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