[OpenSIPS-Users] OpenSIPS-mediaproxy, Asterisk (packet2packet bridge) - problem with RTP

[Dimitrios Giannakopoulos]

Hi,

I have implemented the following scenario:

[incoming pstn]--->[opensips]-->[asterisk] --->[sip phone]
                                                     |
[outgoing pstn]<---[opensips]<------|

Opensips acts as SBC with mediaproxy functionality. Moreover, I use
the LCR module to route calls.
The Asterisk is located at the public domain and we have activated the
packet2packet bridge. A soft phone is registered to asterisk and we
have created a ring group that sends an incoming call to soft phone
and external line (outbound pstn) that rings simultaneous both
devices. Opesips version 1.4.5 or 1.5 Asterisk version 1.6

Single calls without ring gourp:

Incoming calls from PSTN to asterisk through Opensips with mediaproxy
enabled. It works properly.
Outgoing calls from Asterisk to PSTN through Opensips with mediaproxy
enabled. It works properly.


Calls with ring group enabled:
Incoming call from PSTN to asterisk through opensips with mediaproxy
enabled. The incoming call activate the asterisk's  ring group and
sends the call to sip phone and external line – outgoing pstn call.
Both devices ring simultaneous. When hang-up:

A) soft phone, the signaling and media work properly.
B) External line, the signaling works properly but the media is not
open. The system (opensips/mediaproxy) generates two media
sessions(incoming and outgoing) but the ip of asterisk at both
sessions has value Unknown. The mediaproxy/opensips tries to connect
the two legs through asterisk. But this does not work because the
asterisk acts as packet2packet bridge.


Please, can you provide any help/sugestion about this problem?


Regards,

Dimitris