We have a customer using Avaya. Currently, they are using chan_sip. We are working to migrate them to PJSIP. I have not been filled in on the exact scenario. I suspect they have some auto forward feature on the number. Rather than their Avaya transferring internally, they tell Asterisk to transfer to a number (with the Asterisk IP). Doesn't seem correct to me, but it's pretty common for the major switch vendors to do things incorrectly.
We originate from Asterisk to the Avaya endpoint. Send INVITE (with Allow REFER) Receive 100 Trying Receive 180 Ringing Receive 200 O Transmit ACK About a half second later, we receive a REFER from Avaya. I'm not sure if this is normal, but they send us a number to Refer-To @ asterisk IP address. From what I'm being told, they want a one-legged transfer where Asterisk would perform a transfer of this call to the number given @ the Avaya IP address. [11/12 15:09:00.680] VERBOSE[1338] chan_sip.c: SIP transfer to extension 12345@ABC by num...@xyz.org:5060 Transmit 202 Accepted Transmit NOTIFY "SIP/2.0 503 Service Unavailable (can't handle one-legged xfers)" 1) Is there a way to disable sending the REFER in the INVITE's Allow? (chan_sip currently). If not on chan_sip, how about PJSIP? Our theory is if Avaya doesn't receive the Allow: REFER they would do the transfer themselves. 2) Would PJSIP help in any way? Any other thoughts on how to solve this? My next inclination is something like Kamailio between Avaya and Asterisk. This type of transfer seems very insecure. It's basically, Avaya able to tell us to transfer to any number they want (without any restriction). Have a great day! Dan
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