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Could you please help me to allow transfer option for SIP and PRI/PSTN inc Calls?

Hi Pals! I have a problem.

Let me try to explain.

We have a 'machine' SIP server as a Call Center, conecter to our Call Manager with a Trunk SIP, that Call Center recieves Calls from our PRI PBX, and the PBX is conected to our E1 voice link.

When we receive a call from pstn, I goes to our E1 link, then it goes to our PRI Loop, then it goes to our Call manager, Then it goes to the SIP call center, When the SIP call Center, wants to tranfer an incoming call, It holds the first call and try to tranfer to a second one, when the second call answers the tranfers button goes "grey", and the user cant tranfers the first call, BUT that is not all...

After that, when the user tries to get again the first call (from hold state), there is no answer, the call is gone (it seems to be there, but not) and both call are lost, nothing can to do. it's not transferring. both lines disconnecting.

¿Could someone tell me how to allow those call transfer?



Can you turn up troubleshooting traces on CUCM, reproduce the behavior, and attach the CallManager logs (downloadable through RTMT) to this thread? We need to see what is happening in the SIP dialogs to know how to proceed.

The SDI log can get pretty busy. Any details you have about the call (calling/called number, time, what number the agent dialed when starting the transfer, etc) will help us find it. If you have multiple nodes in the cluster please be sure you include the node that the call arrived on from the gateway and the node(s) in the CallManager Group assigned to the SIP trunk of the call center server.



have you resolved this issue? Because I have exactly the same problem from CUCM to a third party SIP Application over a SIP Trunk.

Best regards



I´ve not, the SIP Serve rprovider just learned to live with this, but I did a research on this issue and I found that It could work by setting the 'offnet' Call clasification parameter in the route pattern configuration of you SIP Trunk. I Think this is an implementation of a kind of Toll Fraud Prevention.

Here is a Link

Please, if this information did work to you, reply with what you found.



Might also want to look into MTP just as a suggestion. is MTP checked on the SIP trunk on the CUCM?


Did,t worked yet, =( The "button" to tranfer it remains as gray (inactive), in MTP I have the g711 codec and can not be changed, any idea to resolve this?

I have on every route patter the option for "off-net call" with "provide outside dial tone.

regards. =)