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Can one Cisco gw do media directly to another gw on the SIP interface?
I.e. will you see any difference if you proxy the call through your
rtpproxy?<br>
You don't say anything about how the transfer fails. As you have an
existing media session going, it needs to be replaced and you get
hairpinning of the media.<br>
g-)<br>
<br>
Steven C. Blair wrote:
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<div>Hello:</div>
<div> </div>
<div> We have been having persistent problems with consultative
(attended) call transfer and I’d like to ask the list for some input.</div>
<div> </div>
<div> We have three Cisco 2821 gateways each with two PRIs connecting
to Verizon Centrex service. The route advance sequence defined by
Verizon results in all calls entering our VoIP environment entering via
gateway #1. If no channels exist calls fail over
to gateway #2 and finally #3. </div>
<div> </div>
<div> For load balancing purposes our SER 0.9.7-pre3 proxy will send
outbound calls to gateway #3, then #2 and finally #1.</div>
<div> </div>
<div> We are using Cisco 79x0 phones running SIP load 7.3. All other
types of calls including blind transfers work. </div>
<div> </div>
<div> Call transfers in which both call legs traverse the same
gateway work. Only transfers where each call leg traverses different
gateways fail. Given the inbound and outbound routing described above
it is easy to see how most transfers fail.</div>
<div> </div>
<div> From a protocol perspective the transfer follows RFCs. I have
ethereal traces if anyone needs them.</div>
<div> </div>
<div> Cisco has worked on this problem and is now suspecting that a
transfer will only work if both call legs traverse the same gateway. It
is unclear if this is technically correct, a configuration issue in our
gateway or a mistake.</div>
<div> </div>
<div> To further complicate the issue calls from the PSTN, through a
gateway to an IP phone which is transfers to another IP phone on the
same proxy work. Failures only happen when the second call leg is to
the PSTN and traverse a different gateway.</div>
<div> </div>
<div> Has anyone on this list seen this issue or have any input
whatsoever? </div>
<div> </div>
<div>Thanks,Steve</div>
<div> </div>
<div> </div>
<div> </div>
<div> </div>
<div>Senior Network Engineer, <br>
Information Systems and Computing <br>
Networking and Telecommunications , Suite 221A /6228<br>
University of Pennsylvania <br>
Voice:215-573-8396</div>
<div>FAX:215-898-9348 </div>
<div> </div>
<div> </div>
<div> </div>
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