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[jim at evolutiontel.net]

Hi David,
Have seen a similar issue reported on whirlpool recently with another provider, essentially if the ITSP does not forward the To: header with the correct terminating DID you will not be able to determine the extention to route the call to. Am I correct in saying you only have one Faktortel account with 2 DIDs attached?
Regards,
Jim
- original message -
Subject: [Freeswitch-users] problems with Faktortel (AU) and multiple DID's and extensions
From: David Robinson <pawzlion@gmail.com>
Date: 07/04/2009 13:50

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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