Trying to work out some kinks in a SIP trunk I'm working on between a CUCM 6.1.4 cluster and a modified Asterisk 1.1 server. The problems I have stem from the fact that the Asterisk server will not transfer (i.e. use the SIP REFER method). When I try to transfer a call between a CUCM device and a Asterisk device from the Asterisk to a CUCM device, the Asterisk server INVITEs the other device, and when the transfer is complete, it will drop the call from the Asterisk phone, but the Asterisk server still sits in the middle of the RTP stream, conferencing the two device together. This causes problems for us.
So, I'm wondering if using RFC 2833 DTMF signaling would force Asterisk to use the CUCM device to transfer, rather than a pseudo-transfer conference hosted on the Asterisk. But I'm trying to figure out how that would work. Here is what I see so far...
Cisco IP Phone calls the Asterisk server over the SIP Trunk. I see the SIP INVITE from CUCM to Asterisk. CUCM and Asterisk negotiate and the RTP stream is built between Asterisk and CUCM Phone A. The user on the Aterisk system tries to use hookflash to transfer to a different number. However, the RTP stream is going to Phone A, which cannot interpret the DTMF messages. So how can the DTMF messages get the CUCM server? Or is DTMF signaling not possible over a SIP Trunk?
Cisco CM does not handle transfers on SIP trunks by hookflash or DTMF.
SIP Calls are instead transferred by variety of other methods more or less confusing. It is also quite normal that over a trunk no actual transfer occours and both systems will act ad media termination point to reduce interworking issues.
So, I am not sure what is the problem with the open source system in transferring calls, but if it fails to interpret hookflash and DTMF, there is nothing CM can do about.