[Freeswitch-users] Call setup failes on outbound calls
Babak Yakhchali
babak.freeswitch at gmail.com
Sat Apr 8 15:47:12 MSD 2017
tried it.... but it seems the problem is not related to silence suppression
I donno why this is hapening ...
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Reason: Q.850;cause=98;text="Message not compatible with call state or
message type non-existent or not implemented"
On Sat, Apr 8, 2017 at 3:34 PM, Vallimamod Abdullah <vma at vallimamod.org>
wrote:
> Hi,
>
> After a look in the source code, it seems that it is either CN or
> silenceSupp:off, no other choice as the latter is pretty standard.
> If you are compiling from sources, you can make a quick patch by
> commenting out the 2 lines with "a=silenceSupp:off" in switch_core_media.c
> (not tested.)
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sipsolutions.fr
> .
>
> > On 8 Apr 2017, at 12:26, Babak Yakhchali <babak.freeswitch at gmail.com>
> wrote:
> >
> > thank
> > doing so rsults in :
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > it seems SoftX3000 has problems with any silence related config.. I need
> a way to remove any silence related parameter in sdp
> >
> >
> > On Sat, Apr 8, 2017 at 2:48 PM, Vallimamod Abdullah <vma at vallimamod.org>
> wrote:
> > Hi,
> >
> > It is for comfort noise generation. You can set suppress_cng channel var
> or suppress-cng profile param to true to disable it.
> >
> > Best Regards,
> > --
> > Vallimamod Abdullah
> > SIP Solutions
> > vma at sipsolutions.fr
> > .
> >
> > > On 8 Apr 2017, at 12:08, Babak Yakhchali <babak.freeswitch at gmail.com>
> wrote:
> > >
> > > As I searched more it seems the problem is with this line:
> > > a=rtpmap:13 CN/8000
> > > how can I remove this?
> > >
> > > On Sat, Apr 8, 2017 at 1:48 PM, Babak Yakhchali <
> babak.freeswitch at gmail.com> wrote:
> > > Hi
> > > I'm trying to send calls through a provider with ip 10.10.10.3 but
> call fails with result "Message not compatible with call state or message
> type non-existent or not implemented". how can I debug this problem?
> > > this is siptrace from the failed call:
> > >
> > > send 1061 bytes to udp/[10.10.10.3]:5060 at 13:35:27.382378:
> > > ------------------------------------------------------------
> ------------
> > > INVITE sip:12345678 at 10.10.10.3 SIP/2.0
> > > Via: SIP/2.0/UDP 10.10.10.4;rport;branch=z9hG4bKeN5gp5S5KrZce
> > > Max-Forwards: 69
> > > From: "100" <sip:74413 at 10.10.10.3>;tag=N69FS63aHU7ta
> > > To: <sip:12345678 at 10.10.10.3>
> > > Call-ID: 596dba35-96dd-1235-59bc-00505698455d
> > > CSeq: 105485099 INVITE
> > > Contact: <sip:gw+trunk_1 at 10.10.10.4:5060;transport=udp;gw=trunk_1>
> > > User-Agent: FreeSWITCH-mod_sofia/1.6.15+git~20170303T233525Z~
> 02c0860de6~64bit
> > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY
> > > Supported: timer, path, replaces
> > > Allow-Events: talk, hold, conference, refer
> > > Content-Type: application/sdp
> > > Content-Disposition: session
> > > Content-Length: 242
> > > X-FS-Support: update_display,send_info
> > > Remote-Party-ID: "100" <sip:100 at 10.10.10.3>;party=
> calling;screen=yes;privacy=off
> > >
> > > v=0
> > > o=FreeSWITCH 1491655976 1491655977 IN IP4 10.10.10.4
> > > s=FreeSWITCH
> > > c=IN IP4 10.10.10.4
> > > t=0 0
> > > m=audio 27714 RTP/AVP 0 101 13
> > > a=rtpmap:0 PCMU/8000
> > > a=rtpmap:101 telephone-event/8000
> > > a=fmtp:101 0-16
> > > a=rtpmap:13 CN/8000
> > > a=ptime:20
> > > ------------------------------------------------------------
> ------------
> > > recv 274 bytes from udp/[10.10.10.3]:5060 at 13:35:27.398407:
> > > ------------------------------------------------------------
> ------------
> > > SIP/2.0 100 Trying
> > > Via: SIP/2.0/UDP 10.10.10.4;branch=z9hG4bKeN5gp5S5KrZce;rport=5060
> > > Call-ID: 596dba35-96dd-1235-59bc-00505698455d
> > > From: "100"<sip:74413 at 10.10.10.3>;tag=N69FS63aHU7ta
> > > To: <sip:12345678 at 10.10.10.3>
> > > CSeq: 105485099 INVITE
> > > Content-Length: 0
> > >
> > > ------------------------------------------------------------
> ------------
> > > recv 488 bytes from udp/[10.10.10.3]:5060 at 13:35:27.448335:
> > > ------------------------------------------------------------
> ------------
> > > SIP/2.0 500 Server Internal Error
> > > Via: SIP/2.0/UDP 10.10.10.4;branch=z9hG4bKeN5gp5S5KrZce;rport=5060
> > > Call-ID: 596dba35-96dd-1235-59bc-00505698455d
> > > From: "100"<sip:74413 at 10.10.10.3>;tag=N69FS63aHU7ta
> > > To: <sip:12345678 at 10.10.10.3>;tag=juiky7nh
> > > CSeq: 105485099 INVITE
> > > Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
> > > Reason: Q.850;cause=98;text="Message not compatible with call state
> or message type non-existent or not implemented"
> > > Content-Length: 0
> > >
> > > ------------------------------------------------------------
> ------------
> > > send 321 bytes to udp/[10.10.10.3]:5060 at 13:35:27.448500:
> > > ------------------------------------------------------------
> ------------
> > > ACK sip:12345678 at 10.10.10.3 SIP/2.0
> > > Via: SIP/2.0/UDP 10.10.10.4;rport;branch=z9hG4bKeN5gp5S5KrZce
> > > Max-Forwards: 69
> > > From: "100" <sip:74413 at 10.10.10.3>;tag=N69FS63aHU7ta
> > > To: <sip:12345678 at 10.10.10.3>;tag=juiky7nh
> > > Call-ID: 596dba35-96dd-1235-59bc-00505698455d
> > > CSeq: 105485099 ACK
> > > Content-Length: 0
> > >
> > >
> > > ____________________________________________________________
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> _____________
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> >
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> _____________
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> >
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>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
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>
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