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SIP call transfer / sending SDP in invite

Hi,

I am using a 2811 running 12.4(9)T as a Voice Gateway.

I have incoming calls from h.323, and want to redirect them to a sip endpoint.

This works fine, when I redirect a call to say SJPhone on a PC.

However, when I transfer to a 3rd party SIP enabled VoiceMail system, it does not like the fact that SDP information is not sent in the initial INVITE, on a transfer.

The far end (VoiceMail system) rejects the call with "488 Not Acceptable Here".

Is there any way, in which I can force the gateway to send SDP information in the invite for call transfers ?

I believe its called "delayed media", which is causing me this problem.

For regular direct calls to the 3rd party VoiceMail system, it works fine. (i,e calling from an IP Phone to VoiceMail using sipv2).

Thanks

Shahed.

1 REPLY
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Re: SIP call transfer / sending SDP in invite

Call transfer allows a wide variety of decentralized multiparty call operations. These decentralized call operations form the basis for third-party call control, and thus are important features for VoIP and SIP. Call transfer is also critical for conference calling, where calls can transition smoothly between multiple point-to-point links and IP level multicasting.

Refer Method

The SIP Refer method provides call-transfer capabilities to supplement the Bye and Also methods already implemented on Cisco IOS SIP gateways

Try:

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_chapter09186a00807561f3.html#wp1007537

http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_example09186a0080672b8b.shtml

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