Thanks for the help everyone. You were right Anthony. I put the relevant section of the trace below. I also got ahold of a VoicePulse line and tried it, and it worked fine. I really appreciate it! Trace from my Gizmo line: 2008-05-09 11:30:32 [DEBUG] sofia.c:222 sofia_event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/outbound/17818640108 2008-05-09 11:30:32 [DEBUG] sofia.c:1714 sofia_handle_sip_i_state() Channel sofia/outbound/17818640108 entering state [calling] 2008-05-09 11:30:34 [DEBUG] sofia.c:1714 sofia_handle_sip_i_state() Channel sofia/outbound/17818640108 entering state [ready] 2008-05-09 11:30:34 [DEBUG] sofia.c:1718 sofia_handle_sip_i_state() Remote SDP: v=0 o=root 3158 3158 IN IP4 130.94.88.90 s=session c=IN IP4 130.94.88.90 t=0 0 m=audio 17186 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2008-05-09 11:30:34 [DEBUG] sofia_glue.c:2024 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2008-05-09 11:30:34 [DEBUG] sofia_glue.c:1333 sofia_glue_tech_set_codec() Set Codec sofia/outbound/17818640108 PCMU/8000 20 ms 160 samples 2008-05-09 11:30:34 [DEBUG] sofia_glue.c:1988 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2008-05-09 11:30:34 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() AUDIO RTP [sofia/outbound/17818640108] 69.64.92.160:26720->130.94.88.90:17186 codec: 0 ms: 20 2008-05-09 11:30:34 [DEBUG] switch_rtp.c:716 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-05-09 11:30:34 [NOTICE] sofia.c:2064 sofia_handle_sip_i_state() Channel [sofia/outbound/17818640108] has been answered 2008-05-09 11:30:34 [DEBUG] switch_ivr_originate.c:1213 switch_ivr_originate() Originate Resulted in Success: [sofia/outbound/17818640108] ----- Original Message ---- From: Anthony Minessale <anthony.minessale@gmail.com> To: freeswitch-users@lists.freeswitch.org Sent: Friday, May 9, 2008 8:49:58 AM Subject: Re: [Freeswitch-users] waitForAnswer returns too soon This block : new_session = new Session(); new_session.originate(session, "sofia/gateway/gizmo/1xxxxxxxxxx"); console_log("info","Waiting for answer!\n"); new_session.waitForAnswer(10000); console_log("info","***ANSWERED\n"); Can be refactored to: new_session = new Session(); console_log("info","Waiting for answer!\n"); new_session.originate(session, "{ignore_early_media=true}sofia/gateway/gizmo/1xxxxxxxxxx"); console_log("info","***ANSWERED\n"); Can be refactored to: console_log("info","Waiting for answer!\n"); new_session = new Session("{ignore_early_media=true}sofia/gateway/gizmo/1xxxxxxxxxx"); console_log("info","***ANSWERED\n"); You may want to compare the console log with debug level (press f8 on cli) and start FS with TPORT_LOG=1 env var set. Many voip providers mistakenly answer calls on the inbound SIP leg before bridging it to the PSTN. You should be able to see the SIP dialog and other console lines telling you if the call was actually answered or not. On Fri, May 9, 2008 at 12:56 AM, Jonathan Palley <jpalley@idapted.com> wrote: I think you want: ignore_early_media=true JP On May 9, 2008, at 1:29 PM, Jonas Gauffin wrote:
Doesn't that just allow media to be sent before a 200 ok? In that case, it doesn't solve the waitForAnswer problem?
On Fri, May 9, 2008 at 7:05 AM, Leonardo Alves <leoalves@teltec.com.br> wrote:
Try using the parameter early_media=true From: Marc Orenberg Sent: Thursday, May 08, 2008 9:39 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] waitForAnswer returns too soon Hello everyone. I'm a new FreeSwitch user, and I'm having a problem with waitForAnswer() with SpiderMonkey. I'm using it to make an outbound call as follows:
new_session = new Session(); new_session.originate(session, "sofia/gateway/gizmo/ 1xxxxxxxxxx"); console_log("info","Waiting for answer!\n"); new_session.waitForAnswer(10000); console_log("info","***ANSWERED\n"); do { console_log("info","***Playing file . . . \n"); playFile("we_are_trying_to_reach.wav"); } while(1);
When I run this code, the call is answered and it starts playing the file before my phone even starts ringing. I'm not sure if it's a bug with waitForAnswer, or a configuration issue on my side.
For the SIP provider, I'm using Gizmo Phone with the following settings in sip_profiles/outbound/gizmo.xml:
<include> <gateway name="gizmo"> <param name="username" value="1747xxxxxxx"/> <param name="password" value="xxxxxxx"/> <param name="realm" value="proxy01.sipphone.com"/> <param name="proxy" value="proxy01.sipphone.com"/> <param name="register" value="true"/> </gateway> </include>
I'd really appreciate any help.
Thanks, Marc
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_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400