Discussion:
[Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP
Jerry Richards
2009-11-05 23:49:30 UTC
Permalink
I am trying to make a call through a Gateway that sends the INVITE with no
SDP and ONLY wants the 200 OK w/SDP when the callee answers.

For some reason, Freeswitch answers the call with 200 OK w/SDP even before
the callee answers the phone. Is this to provide ringback? Can I disable
that action?

Best Regards,
Jerry
Brian West
2009-11-05 23:58:30 UTC
Permalink
This all depends on what you're doing in your dialplan if you do stuff
like record it requires media and will trigger it.

A sip trace or some such debug would be more helpful then a terse
description of a problem.

/b
Post by Jerry Richards
I am trying to make a call through a Gateway that sends the INVITE with no
SDP and ONLY wants the 200 OK w/SDP when the callee answers.
For some reason, Freeswitch answers the call with 200 OK w/SDP even before
the callee answers the phone. Is this to provide ringback? Can I disable
that action?
Best Regards,
Jerry
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
users
http://www.freeswitch.org
Claudiu Filip
2009-11-06 10:27:00 UTC
Permalink
Hi Jerry,


Have a look at 3pcc-enable option in your sip profile. You may want
to set it "proxy", even if it's not that RFC compliant and has some
issues with codec negotiation (FS advertise global_codecs to both
parties and it may result in having different codecs on each leg =>
transcoding or call drop if transcoding not possible).


Best regards,

Claudiu Filip
claudiu at departamentul.it


Friday, November 6, 2009, 1:49:30 AM, you wrote:
Jerry> I am trying to make a call through a Gateway that sends the INVITE with no
Jerry> SDP and ONLY wants the 200 OK w/SDP when the callee answers.

Jerry> For some reason, Freeswitch answers the call with 200 OK w/SDP even before
Jerry> the callee answers the phone. Is this to provide ringback? Can I disable
Jerry> that action?

Jerry> Best Regards,
Jerry> Jerry


Jerry> _______________________________________________
Jerry> FreeSWITCH-users mailing list
Jerry> FreeSWITCH-users at lists.freeswitch.org
Jerry> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Jerry> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Jerry> http://www.freeswitch.org
Mathieu Rene
2009-11-06 18:33:17 UTC
Permalink
Are you recording? I recall a recent change to force answer whenever
record_session is called.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca
Post by Claudiu Filip
Hi Jerry,
Have a look at 3pcc-enable option in your sip profile. You may want
to set it "proxy", even if it's not that RFC compliant and has some
issues with codec negotiation (FS advertise global_codecs to both
parties and it may result in having different codecs on each leg =>
transcoding or call drop if transcoding not possible).
Best regards,
Claudiu Filip
claudiu at departamentul.it
Jerry> I am trying to make a call through a Gateway that sends the INVITE with no
Jerry> SDP and ONLY wants the 200 OK w/SDP when the callee answers.
Jerry> For some reason, Freeswitch answers the call with 200 OK w/
SDP even before
Jerry> the callee answers the phone. Is this to provide ringback?
Can I disable
Jerry> that action?
Jerry> Best Regards,
Jerry> Jerry
Jerry> _______________________________________________
Jerry> FreeSWITCH-users mailing list
Jerry> FreeSWITCH-users at lists.freeswitch.org
Jerry> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Jerry> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Jerry> http://www.freeswitch.org
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Mathieu Rene
2009-11-06 18:33:17 UTC
Permalink
Are you recording? I recall a recent change to force answer whenever
record_session is called.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca
Post by Claudiu Filip
Hi Jerry,
Have a look at 3pcc-enable option in your sip profile. You may want
to set it "proxy", even if it's not that RFC compliant and has some
issues with codec negotiation (FS advertise global_codecs to both
parties and it may result in having different codecs on each leg =>
transcoding or call drop if transcoding not possible).
Best regards,
Claudiu Filip
claudiu at departamentul.it
Jerry> I am trying to make a call through a Gateway that sends the INVITE with no
Jerry> SDP and ONLY wants the 200 OK w/SDP when the callee answers.
Jerry> For some reason, Freeswitch answers the call with 200 OK w/
SDP even before
Jerry> the callee answers the phone. Is this to provide ringback?
Can I disable
Jerry> that action?
Jerry> Best Regards,
Jerry> Jerry
Jerry> _______________________________________________
Jerry> FreeSWITCH-users mailing list
Jerry> FreeSWITCH-users at lists.freeswitch.org
Jerry> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Jerry> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Jerry> http://www.freeswitch.org
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Brian West
2009-11-05 23:58:30 UTC
Permalink
This all depends on what you're doing in your dialplan if you do stuff
like record it requires media and will trigger it.

A sip trace or some such debug would be more helpful then a terse
description of a problem.

/b
Post by Jerry Richards
I am trying to make a call through a Gateway that sends the INVITE with no
SDP and ONLY wants the 200 OK w/SDP when the callee answers.
For some reason, Freeswitch answers the call with 200 OK w/SDP even before
the callee answers the phone. Is this to provide ringback? Can I disable
that action?
Best Regards,
Jerry
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
users
http://www.freeswitch.org
Claudiu Filip
2009-11-06 10:27:00 UTC
Permalink
Hi Jerry,


Have a look at 3pcc-enable option in your sip profile. You may want
to set it "proxy", even if it's not that RFC compliant and has some
issues with codec negotiation (FS advertise global_codecs to both
parties and it may result in having different codecs on each leg =>
transcoding or call drop if transcoding not possible).


Best regards,

Claudiu Filip
claudiu at departamentul.it


Friday, November 6, 2009, 1:49:30 AM, you wrote:
Jerry> I am trying to make a call through a Gateway that sends the INVITE with no
Jerry> SDP and ONLY wants the 200 OK w/SDP when the callee answers.

Jerry> For some reason, Freeswitch answers the call with 200 OK w/SDP even before
Jerry> the callee answers the phone. Is this to provide ringback? Can I disable
Jerry> that action?

Jerry> Best Regards,
Jerry> Jerry


Jerry> _______________________________________________
Jerry> FreeSWITCH-users mailing list
Jerry> FreeSWITCH-users at lists.freeswitch.org
Jerry> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Jerry> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Jerry> http://www.freeswitch.org
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