[Freeswitch-users] Incoming call from registered gateway with DID in To: not URI

[Steven Ayre]

I'm currently handling it with:

  <param name="context" value="numbergroup"/>

and:

  <context name="numbergroup">
    <extension name="numbergroup">
      <condition field="destination_number" expression="^USERNAME$">
        <action application="transfer" data="${sip_to_user} XML default"/>
      </condition>
    </extension>
  </context>

Is there a better way of handling it? Such as a param that tells Sofia
to use the To not the URI to populate destination_number?

-Steve


On 7 January 2011 21:08, Steven Ayre <steveayre at gmail.com> wrote:
> Hi everyone,
>
> I have a gateway registering to numbergroup.com, this is the configuration:
>
> <gateway name="numbergroup">
>    <param name="realm" value="sip.numbergroup-services.com"/>
>    <param name="username" value="XXXXXX"/>
>    <param name="password" value="XXXXXX"/>
>    <param name="register" value="true"/>
>    <param name="extension-in-contact" value="true"/>
>    <param name="caller-id-in-from" value="true"/>
>    <param name="ping" value="60"/>
>    <param name="ping-max" value="10"/>
>    <param name="ping-min" value="1"/>
>    <param name="retry-seconds" value="5"/>
>    <param name="expire-seconds" value="60"/>
> </gateway>
>
> Incoming calls from my DID to a SIP URI on the server work fine.
> However I'm having problems sending it to the SIP Trunk (the
> registration above).
>
> The call arrives on the server fine, but the URI is the
> username at sip.numbergroup-services.com. The DID is in the To: header
>
> destination_number in the dialplan is the username not the DID as a
> result. Does anyone know how to configure the gateway so that the
> destination_number would contain the DID from the To header instead?
>
> Here's the INVITE:
>
>   INVITE sip:username at 81.27.101.246:5060;transport=udp;gw=numbergroup SIP/2.0
>   Via: SIP/2.0/UDP 80.93.165.111;rport;branch=z9hG4bKD4m88SNemDSBg
>   Max-Forwards: 65
>   From: "+myphonenumber" <sip:+myphonenumber at 80.93.165.111>;tag=y9DBpB33yBHNN
>   To: <sip:+mydidnumber at sip.numbergroup-services.com>
>   Call-ID: 9ee8ce2d-9544-122e-c2a7-002655d1d302
>   CSeq: 6881084 INVITE
>   Contact: <sip:numbergroup at 80.93.165.111:5060>
>   User-Agent: numbergroup.com
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, hold, refer
>   Session-Expires: 900
>   Min-SE: 90
>   Privacy: none
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 593
>   P-Charging-Vector:
> icid-value=f4571a9c-1aa1-11e0-8310-5798bfd3ad89;icid-generated-at=80.93.165.110;orig-ioi=numbergroup.com
>   P-Asserted-Identity: "+myphonenumber" <sip:+myphonenumber at 80.93.165.111>
>
>   v=0
>   o=numbergroup 1294416066 1294416067 IN IP4 80.93.165.111
>   s=numbergroup
>   c=IN IP4 80.93.165.111
>   t=0 0
>   m=audio 18230 RTP/AVP 8 9 98 3 18 99 100 101 102 103 104 0 105 101 13
>   a=rtpmap:98 G7221/32000
>   a=fmtp:98 bitrate=48000
>   a=rtpmap:99 SPEEX/8000
>   a=rtpmap:100 iLBC/8000
>   a=fmtp:100 mode=20
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=rtpmap:102 G726-24/8000
>   a=rtpmap:103 G726-32/8000
>   a=rtpmap:104 G726-40/8000
>   a=rtpmap:105 CELT/48000
>   m=video 18540 RTP/AVP 106 107 34 31
>   a=rtpmap:106 THEORA/90000
>   a=rtpmap:107 H264/90000
>   a=rtpmap:34 H263/90000
>   a=rtpmap:31 H261/90000
>
> -Steve
>