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New Member

SIP Call Transfer Using the Refer Method

Hi,

I'm having trouble with call transfers. I'm trying to do an attended transfer using the Refer method, as explained in http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipc1_c/chapter2.htm (Table 2 and Figure 17).

I have a SIP UA (Originator), a Cisco 3620 with a T1 to a PBX (Recipient), and another SIP UA (Final-Recipient).

The Originator correctly sends a Refer to the Recipient, and the Recipient replies with 202 Accepted. However, the Recipient does not send an Invite to the Final-Recipient...

Any hint/suggestion is appreciated.

Thank you,

Tatsuo

210.81.25.248 Originator

210.81.25.149 SIP Proxy

210.81.25.140 Recipient

210.81.25.216 Final-Recipient

SIP MESSAGE 45 210.81.25.248:5060() -> 210.81.25.149:5060() UDP Frame 45 5/Dec/03 15:16:25.1372 TimeFromPreviousSipFrame=4.2130 TimeFromStart=19.2390 REFER sip:snom.sipcenter.jp:5060;maddr=210.81.25.149;transport=udp SIP/2.0 Via: SIP/2.0/UDP 210.81.25.248:5060;branch=z9hG4bK-mfwcoehm5v5e Route: <sip:1503@210.81.25.140:5060> From: <sip:1001@snom.sipcenter.jp>;tag=mevq6ztza7 To: <sip:1503@snom.sipcenter.jp;user=phone>;tag=860C70-98A Call-ID: 3c279e68ee0c-iw483wrl88j7@210-81-25-248 CSeq: 4 REFER Max-Forwards: 70 Contact: <sip:1001@210.81.25.248:5060;transport=udp;line=1> Refer-To: sip:1003@snom.sipcenter.jp;user=phone?Replaces=3c279e759568-ls8rwho3969y%40210-81-25-248%3Bto-tag%3D8gbd7zcbwl%3Bfrom-tag%3D1qjl7pgamw Referred-By: sip:1001@snom.sipcenter.jp Content-Language: en Content-Length: 0

--------------------------------------------------------------------------------

SIP MESSAGE 46 210.81.25.149:5060() -> 210.81.25.248:5060() UDP Frame 46 5/Dec/03 15:16:25.1383 TimeFromPreviousSipFrame=0.0011 TimeFromStart=19.2402 SIP/2.0 100 Trying Via: SIP/2.0/UDP 210.81.25.248:5060;branch=z9hG4bK-mfwcoehm5v5e From: <sip:1001@snom.sipcenter.jp>;tag=mevq6ztza7 To: <sip:1503@snom.sipcenter.jp;user=phone>;tag=860C70-98A Call-ID: 3c279e68ee0c-iw483wrl88j7@210-81-25-248 CSeq: 4 REFER Content-Length: 0

--------------------------------------------------------------------------------

SIP MESSAGE 47 210.81.25.149:5060() -> 210.81.25.140:5060() UDP Frame 47 5/Dec/03 15:16:25.1401 TimeFromPreviousSipFrame=0.0018 TimeFromStart=19.2420 REFER sip:1503@210.81.25.140:5060 SIP/2.0 Via: SIP/2.0/UDP 210.81.25.149:5060;branch=z9hG4bK-a049450e3389abf5029c95f21b959907.1 Via: SIP/2.0/UDP 210.81.25.248:5060;branch=z9hG4bK-mfwcoehm5v5e Record-Route: <sip:snom.sipcenter.jp:5060;maddr=210.81.25.149;transport=udp> From: <sip:1001@snom.sipcenter.jp>;tag=mevq6ztza7 To: <sip:1503@snom.sipcenter.jp;user=phone>;tag=860C70-98A Call-ID: 3c279e68ee0c-iw483wrl88j7@210-81-25-248 CSeq: 4 REFER Max-Forwards: 69 Contact: <sip:1001@210.81.25.248:5060;transport=udp;line=1> Content-Language: en Referred-By: sip:1001@snom.sipcenter.jp Refer-To: sip:1003@snom.sipcenter.jp;user=phone?Replaces=3c279e759568-ls8rwho3969y%40210-81-25-248%3Bto-tag%3D8gbd7zcbwl%3Bfrom-tag%3D1qjl7pgamw Content-Length: 0

--------------------------------------------------------------------------------

SIP MESSAGE 48 210.81.25.140:5060() -> 210.81.25.149:5060() UDP Frame 48 5/Dec/03 15:16:25.1578 TimeFromPreviousSipFrame=0.0177 TimeFromStart=19.2596 SIP/2.0 202 Accepted Via: SIP/2.0/UDP 210.81.25.149:5060;branch=z9hG4bK-a049450e3389abf5029c95f21b959907.1,SIP/2.0/UDP 210.81.25.248:5060;branch=z9hG4bK-mfwcoehm5v5e From: <sip:1001@snom.sipcenter.jp>;tag=mevq6ztza7 To: <sip:1503@snom.sipcenter.jp;user=phone>;tag=860C70-98A Date: Mon, 01 Mar 1993 02:26:44 GMT Call-ID: 3c279e68ee0c-iw483wrl88j7@210-81-25-248 Server: Cisco-SIPGateway/IOS-12.x CSeq: 4 REFER Content-Length: 0

--------------------------------------------------------------------------------

SIP MESSAGE 49 210.81.25.149:5060() -> 210.81.25.248:5060() UDP Frame 49 5/Dec/03 15:16:25.1588 TimeFromPreviousSipFrame=0.0010 TimeFromStart=19.2606 SIP/2.0 202 Accepted Via: SIP/2.0/UDP 210.81.25.248:5060;branch=z9hG4bK-mfwcoehm5v5e From: <sip:1001@snom.sipcenter.jp>;tag=mevq6ztza7 To: <sip:1503@snom.sipcenter.jp;user=phone>;tag=860C70-98A Call-ID: 3c279e68ee0c-iw483wrl88j7@210-81-25-248 CSeq: 4 REFER Server: Cisco-SIPGateway/IOS-12.x Date: Mon, 01 Mar 1993 02:26:44 GMT Content-Length: 0

Expecting here to see an Invite from the Recipient to the Final-Recipient. But nothing...

END

1 ACCEPTED SOLUTION

Accepted Solutions
Silver

Re: SIP Call Transfer Using the Refer Method

Since INVITE is not sent by the receipent, please configure "application session" in the pots and voip dial-peers of the receipent. If this did not work please send your configs and the image that is being used.

2 REPLIES
Silver

Re: SIP Call Transfer Using the Refer Method

Since INVITE is not sent by the receipent, please configure "application session" in the pots and voip dial-peers of the receipent. If this did not work please send your configs and the image that is being used.

New Member

Re: SIP Call Transfer Using the Refer Method

amritpatek, that was it! Thank you very much, Tatsuo

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