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Sip destination signature RRS feed

  • Question

  •  

    Is anyone aware of correct SIP destination format for VXML call transfer?

     

    I am trying to transfer current call to an extension, but this is failing.

     

    <transfer name="test_sub" destexpr="'sip:153@gateway_ip:gateway_port'" type="Consultation"/>

     

    Thanks,

    Ashish.

    Thursday, August 2, 2007 1:51 PM

Answers

  • I believe you are correct -- you are meant to supply just the phone number, and the platform constructs the SIP URI. For outbound calls, we will round-robin over all available outbound SIP peers.

     

    Dan

     

    Thursday, August 2, 2007 6:05 PM

All replies

  • In addition to the above post: Please note that specifying "153" in the destination address works just fine. OCS is internally translating this as the exact same address noted above. But me explicitly specifying that address doesn't work.??

     

    Also, does OCS support the post dial feature as discussed in the RFC2086 http://www.ietf.org/rfc/rfc2806.txt?

     

    Thanks.

    Thursday, August 2, 2007 3:48 PM
  • I believe you are correct -- you are meant to supply just the phone number, and the platform constructs the SIP URI. For outbound calls, we will round-robin over all available outbound SIP peers.

     

    Dan

     

    Thursday, August 2, 2007 6:05 PM
  • are you implying there is no way for us to specify a valid sip destination, and not leave it for the platform to construct the SIP URI. That seems like a big limitation.

     

    Please clarify.

     

    Wednesday, August 15, 2007 6:11 PM