[Asterisk-Dev] SIP REFER transferring

[Florian Overkamp]

Hi,

So I've been delving into my issues with Cisco gear again and I think the
ATA is doing REFER differently from other gear. I have made a transfer dump
for evaluation, and would like to compare it with a REFER transfer
(supervised) that _DID_ work properly.

What goes wrong with Asterisk and ATA186:

After the proper changes to CallCmd have been made, you can do supervised
transfer almost as documented:

Speak to party A
Hit hookflash, dial party B
Speak to party B
Hangup
Party A is REFER'ed to party B

However, during this REFER procedure, all calls are hung up and it seems as
though party A is actually _calling_ party B immediately after. Since party
B may still be busy (listening to the hangup signals or silence) this only
works if party B has callwaiting active.

Apparently, CCM _can_ do this properly so I'm wondering if this is a nasty
implementation of REFER in CCM or a bug in Asterisk ? Or even something
else...

I would be glad to look at an ethereal dump of a REFER transfer that _DID_
work the way people expect it to, and I would be happy to provide the
'faulty' REFER dump to anyone who has an eye to evaluate it.

Thanks,
Florian