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[Freeswitch-users] Freeswitch not processing calls


 
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brian at freeswitch.org
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PostPosted: Mon Feb 09, 2009 5:17 am    Post subject: [Freeswitch-users] Freeswitch not processing calls Reply with quote

I notice it offers 18 which is G729 but these listed below are 100%
invalid. There is no such thing as G.729a, G.729b or G.729ab that are
valid in the SDP. I suspect if you start FreeSWITCH and crank it up
to debug level ("console loglevel debug") you'll clearly see why this
is taking place. I think we talked to you on IRC about this and told
this. If your termination gateway requires any of the above listed on
96,97 or 98 the call will fail. We can only do passthru on G729 which
is the official name which is listed on payload 18. Your termination
provider needs to be informed of this fact. The fmtp line is what
controls annexb operation!

/b

On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote:

Quote:
a=rtpmap:97 G.729b/8000.
a=rtpmap:98 G.729a/8000.
a=rtpmap:96 G.729ab/8000.


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ahgindia308 at gmail.com
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PostPosted: Mon Feb 09, 2009 6:50 am    Post subject: [Freeswitch-users] Freeswitch not processing calls Reply with quote

Hi Brian,
But issue here is that, FS is not processing any such calls and not sending
488 to the caller.
Also the sip trace I had provided was from the caller to fs. FS does not
even bridge the call to terminator in between the INVITE and CANCEL from the
caller.
It just gives so many errors in log like following :

2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is
only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is
only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is
only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is
only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is
only usable in passthrough mode!

It seems odd that due to codec mismatch, fs does not process the call at all
and just times out and the caller sends CANCEL to us.
Waiting for your reply.


Brian West-3 wrote:
Quote:

I notice it offers 18 which is G729 but these listed below are 100%
invalid. There is no such thing as G.729a, G.729b or G.729ab that are
valid in the SDP. I suspect if you start FreeSWITCH and crank it up
to debug level ("console loglevel debug") you'll clearly see why this
is taking place. I think we talked to you on IRC about this and told
this. If your termination gateway requires any of the above listed on
96,97 or 98 the call will fail. We can only do passthru on G729 which
is the official name which is listed on payload 18. Your termination
provider needs to be informed of this fact. The fmtp line is what
controls annexb operation!

/b

On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote:

Quote:
a=rtpmap:97 G.729b/8000.
a=rtpmap:98 G.729a/8000.
a=rtpmap:96 G.729ab/8000.


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ahgindia308 at gmail.com
Guest





PostPosted: Mon Feb 09, 2009 7:09 am    Post subject: [Freeswitch-users] Freeswitch not processing calls Reply with quote

Here is the correct codec sent to fs, but it times out again.
http://www.nabble.com/file/p21911561/correct_codec_with_cancel.txt
correct_codec_with_cancel.txt


Ankit Gandhi wrote:
Quote:

Hi Brian,
But issue here is that, FS is not processing any such calls and not
sending 488 to the caller.
Also the sip trace I had provided was from the caller to fs. FS does not
even bridge the call to terminator in between the INVITE and CANCEL from
the caller.
It just gives so many errors in log like following :

2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!

It seems odd that due to codec mismatch, fs does not process the call at
all and just times out and the caller sends CANCEL to us.
Waiting for your reply.


Brian West-3 wrote:
Quote:

I notice it offers 18 which is G729 but these listed below are 100%
invalid. There is no such thing as G.729a, G.729b or G.729ab that are
valid in the SDP. I suspect if you start FreeSWITCH and crank it up
to debug level ("console loglevel debug") you'll clearly see why this
is taking place. I think we talked to you on IRC about this and told
this. If your termination gateway requires any of the above listed on
96,97 or 98 the call will fail. We can only do passthru on G729 which
is the official name which is listed on payload 18. Your termination
provider needs to be informed of this fact. The fmtp line is what
controls annexb operation!

/b

On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote:

Quote:
a=rtpmap:97 G.729b/8000.
a=rtpmap:98 G.729a/8000.
a=rtpmap:96 G.729ab/8000.


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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





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View this message in context: http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21911561.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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anthony.minessale at g...
Guest





PostPosted: Mon Feb 09, 2009 8:53 am    Post subject: [Freeswitch-users] Freeswitch not processing calls Reply with quote

1) please do not report bugs on the mailing list.
2) please report the bug on jira http://jira.freeswitch.org according to the rules: http://wiki.freeswitch.org/wiki/Reporting_Bugs

If you have an issue that you want us to correct you will have to try the latest SVN trunk (not a snapshot) issue "make current" from your RC1 directory.

Attach the entire console log output from start of call to finish with console loglevel debug.



On Mon, Feb 9, 2009 at 6:06 AM, Ankit Gandhi <ahgindia308@gmail.com (ahgindia308@gmail.com)> wrote:
Quote:

Here is the correct codec sent to fs, but it times out again.
http://www.nabble.com/file/p21911561/correct_codec_with_cancel.txt
correct_codec_with_cancel.txt



Ankit Gandhi wrote:
Quote:

Hi Brian,
But issue here is that, FS is not processing any such calls and not
sending 488 to the caller.
Also the sip trace I had provided was from the caller to fs. FS does not
even bridge the call to terminator in between the INVITE and CANCEL from
the caller.
It just gives so many errors in log like following :

2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!

It seems odd that due to codec mismatch, fs does not process the call at
all and just times out and the caller sends CANCEL to us.
Waiting for your reply.


Brian West-3 wrote:
Quote:

I notice it offers 18 which is G729 but these listed below are 100%
invalid. There is no such thing as G.729a, G.729b or G.729ab that are
valid in the SDP. I suspect if you start FreeSWITCH and crank it up
to debug level ("console loglevel debug") you'll clearly see why this
is taking place. I think we talked to you on IRC about this and told
this. If your termination gateway requires any of the above listed on
96,97 or 98 the call will fail. We can only do passthru on G729 which
is the official name which is listed on payload 18. Your termination
provider needs to be informed of this fact. The fmtp line is what
controls annexb operation!

/b

On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote:

Quote:
a=rtpmap:97 G.729b/8000.
a=rtpmap:98 G.729a/8000.
a=rtpmap:96 G.729ab/8000.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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





--


View this message in context: http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21911561.html

Sent from the Freeswitch-users mailing list archive at Nabble.com.


_______________________________________________
Freeswitch-users mailing list
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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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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