Hey all,<br><br>I've been playing around with trying to bridge dingaling to sofia for some time now. I've had some success.<br><br>1) I can call to googletalk using a local sip registration<br>2) I can call to googletalk using a remote sip registration
<br>3) I can call from googletalk to a local sip registration<br>4) I can call from a local sip registration to a remote sip registration<br><br>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.
<br><br>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.<br><br>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.
<br><br>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<br>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
<br>2007-03-21 12:56:13 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/XX.XX.XX.XX/kieran entering state [proceeding]<br>2007-03-21 12:56:13 [NOTICE] mod_sofia.c:3018 sip_i_state() Ring-Ready sofia/XX.XX.XX.XX/kieran!
<br>2007-03-21 12:56:13 [INFO] switch_core.c:1872 <br>2007-03-21 12:56:13 [INFO] switch_core.c:1872 switch_core_session_receive_message() Kill DingaLing/1004j [BREAK]<br>2007-03-21 12:56:13 [DEBUG] mod_dingaling.c:1182 channel_kill_channel() DingaLing/1004j CHANNEL KILL
<br>2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/XX.XX.XX.XX/kieran entering state [ready]<br>2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2996 sip_i_state() Remote SDP:<br>v=0<br>o=- 7 2 IN IP4 <a href="http://192.168.2.53">
192.168.2.53</a><br>s=CounterPath X-Lite 3.0<br>c=IN IP4 <a href="http://192.168.2.53">192.168.2.53</a><br>t=0 0<br>m=audio 37468 RTP/AVP 0 101<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br><br>2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2595 negotiate_sdp() Codec Compare [PCMU:0]/[PCMU:0]
<br>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<br>2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2571 negotiate_sdp() Set 2833 dtmf payload to 101<br>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-><a href="http://192.168.2.53:37468">192.168.2.53:37468</a> codec: 0 ms: 20<br>2007-03-21 12:56:15 [DEBUG] switch_rtp.c:487 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms<br>2007-03-21 12:56:15 [NOTICE] mod_sofia.c:3247 sip_i_state() Channel [sofia/XX.XX.XX.XX/kieran] has been answered
<br>2007-03-21 12:56:15 [DEBUG] switch_ivr.c:3074 switch_ivr_originate() Originate Resulted in Success: [sofia/XX.XX.XX.XX/kieran]<br><br>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.
<br><br>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 :-).<br><br>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.
<br><br>Thanks for any help.<br><br>Kieran.<br>