Hey all, I've been playing around with trying to bridge dingaling to sofia for some time now. I've had some success. 1) I can call to googletalk using a local sip registration 2) I can call to googletalk using a remote sip registration 3) I can call from googletalk to a local sip registration 4) I can call from a local sip registration to a remote sip registration In case my terminology is bad when I say local registration I mean that the soft-phone is registered with freeswitch. When I say remote sip registration I mean that the soft-phone is registered with my standard SIP provider. The problem is if I call from googletalk and attempt to bridge the call to my remote sip provider the call rings, but there is no audio. I've tracked through the console and here is the piece that's different. This piece never shows up if I attempt to bridge the call to a remote sip registration. I copied this from the console when bridging to a local sip registration. 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:4982 event_callback() event [nua_r_invite] status [180][Ringing] session: sofia/XX.XX.XX.XX/kieran 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:4982 event_callback() event [nua_i_state] status [180][Ringing] session: sofia/XX.XX.XX.XX/kieran 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/XX.XX.XX.XX/kieran entering state [proceeding] 2007-03-21 12:56:13 [NOTICE] mod_sofia.c:3018 sip_i_state() Ring-Ready sofia/XX.XX.XX.XX/kieran! 2007-03-21 12:56:13 [INFO] switch_core.c:1872 2007-03-21 12:56:13 [INFO] switch_core.c:1872 switch_core_session_receive_message() Kill DingaLing/1004j [BREAK] 2007-03-21 12:56:13 [DEBUG] mod_dingaling.c:1182 channel_kill_channel() DingaLing/1004j CHANNEL KILL 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/XX.XX.XX.XX/kieran entering state [ready] 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2996 sip_i_state() Remote SDP: v=0 o=- 7 2 IN IP4 192.168.2.53 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.2.53 t=0 0 m=audio 37468 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2595 negotiate_sdp() Codec Compare [PCMU:0]/[PCMU:0] 2007-03-21 12:56:15 [INFO] mod_sofia.c:1578 tech_set_codec() Set Codec sofia/XX.XX.XX.XX/kieran PCMU/8000 20 ms 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2571 negotiate_sdp() Set 2833 dtmf payload to 101 2007-03-21 12:56:15 [INFO] mod_sofia.c:1635 activate_rtp() RTP [sofia/XX.XX.XX.XX/kieran] XX.XX.XX.XX:16386->192.168.2.53:37468 codec: 0 ms: 20 2007-03-21 12:56:15 [DEBUG] switch_rtp.c:487 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2007-03-21 12:56:15 [NOTICE] mod_sofia.c:3247 sip_i_state() Channel [sofia/XX.XX.XX.XX/kieran] has been answered 2007-03-21 12:56:15 [DEBUG] switch_ivr.c:3074 switch_ivr_originate() Originate Resulted in Success: [sofia/XX.XX.XX.XX/kieran] The weird thing is that if I call the same extension in default_context.xml using a sip phone registered locally it bridges without any problem to the remote sip registration. Can anyone please help with this? I've been battling it for a long time now. I've learned a lot which is good though :-). By the way I just downloaded the last svn version today, so I couldn't be on a more recent version :-) Also if this isn't the best place to address this type of question if anyone could point me in the right direction that would be greatly appreciated. Thanks for any help. Kieran.