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 <[EMAIL PROTECTED]>
To: [email protected]
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 <[EMAIL PROTECTED]> 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
> <[EMAIL PROTECTED]> wrote:
>> Try using the parameter early_media=true
>> From: Marc Orenberg
>> Sent: Thursday, May 08, 2008 9:39 PM
>> To: [email protected]
>> 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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>>
>
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Anthony Minessale II
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