I think I’ve found a bug in how the Manager API handles "Redirect"s for SIP phones.
If a SIP phone is connected to some internal service, say Voice mail or a meetme you can redirect the call to any other internal service or to an external device (another SIP phone).
However, once you redirect the call to another external device (SIP phone) and Asterisk sends the appropriate SIP invites to get the RTP streams connected together between the two external devices, Asterisk doesn’t seem to want to re-invite the call back to it self should the call need to return to some internal service.
Of course this can be avoided with “canreinvite=no” in the sip.conf on all of your SIP phone, but then all of the RTP streams have to be relayed by the Asterisk server, which would be sub-optimal in an environment where some of the phones have to travel over a wan to get to the Asterisk PBX.
I’m attempting to fix this my self but I only have a few days experience with the Asterisk code.
Has anyone else bumped into this, and maybe fixed it?
Or are we all happy with the “canreinvite=no” fix?