Attended transfers with original CID information - Polycom


we use Polycom SP IP 501 phones. We use the standard key/soft key configuration to do attended transfers. The only thing we miss is the CID info of the original caller after the call is transfered. This behaviour is different than the blind/direct transfer.
We already opened a call (in 2006) with Polycom JIRA. This is what they told us.
I have had some discussions with the Asterisk developers. This feature is not currently supported on the Asterisk platform. We do support it with some of our other soft-switch partners. This feature is being considered for an Asterisk release towards the end of 2007.

We use Asterisk 1.2.28. The firmware of the phone 2.12.

What do you think about this? I know that we can reach the wanted with the Asterisk “attended transfer”, but the Polycom integrated is much more comfortable.

Your comments on this are much appreciated.

Thanks, Bernd

Yes. Asterisk does not support it. The logic is as follows.

Blind transfer: Call comes in -> Callee A picks up. -> Callee A blind transfers to callee B. Since Asterisk knows the cid of the call that you are transfering it can pass it to callee B.
Attended Transfer: Call comes in -> Callee A Picks up. With an attended transfer you are actually creating a new call (so your CID shows up) then the call is sent over. Since you initially made a “new” call they see your CID.

If you want asterisk to support “proper” CID on attended transfers why not put a bounty for some one to code it in to asterisk ?

Don’t think it’s fair to suggest paying someone a bounty to fix what is at best a ‘feaure’ and at worst a bug. The ‘o’ flag in the dial command should work as advertised. I understand why it doesn’t, but never the less it should. Failing that the docs should spell it out.

The ‘o’ flag is useless on attended transfers!

This week we got the Wireshark log for working attended transfer of the Polycom phones with another softswitch from Polycom. Now we are planning to let this develop and we then, if it will work, let the code go to the Asterisk trunks, hopefully. :wink: