[Freeswitch-users] FS - MjSip no voice [SOLVED] SIP 200 / 183 problem
can_man at gmx.de
can_man at gmx.de
Tue Mar 31 07:06:48 PDT 2009
Hello,
I have found the problem. FS on my local network sends "SIP/2.0 200 OK"
after an invite and FS on the net through the external profil sends
SIP/2.0 183 Session Progress. But MjSip doesn't know how to deal with
183, so it just ignores the message. For testing I have changed
the 183 header to the 200 one and now it works.
Thank you for your help and the quick response time.
Best wishes,
Phil
>From FS on the net through the external profil:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 90.181.59.141:5090;rport=60315;branch=z9hG4bK256321;received=78.105.17.88
From: <sip:puli at 90.181.59.141:5090>;tag=z9hG4bK40977269
To: <sip:2345 at 90.181.59.141:5090>;tag=vgg3Zja8pNQcg
Call-ID: 507347917247 at 90.181.59.141
CSeq: 1 INVITE
Contact: <sip:mod_sofia at 90.181.59.141:5090;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12839M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 267
v=0
o=FreeSWITCH 1072777625698755085 8893522831081357051 IN IP4 90.181.59.141
s=FreeSWITCH
c=IN IP4 91.121.59.148
t=0 0
m=audio 26722 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
>From FS in my local network:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.143:5060;rport=5060;branch=z9hG4bK423233;received=192.168.1.102
From: <sip:brian at 192.168.1.143>;tag=z9hG4bK42598163
To: <sip:1000 at 192.168.1.143>;tag=Q0X494ZUNaKHH
Call-ID: 961142687222 at 192.168.1.143
CSeq: 2 INVITE
Contact: <sip:mod_sofia at 192.168.1.143:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12712M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Session-Expires: 120;refresher=uas
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 267
v=0
o=FreeSWITCH 5195745633884389954 8941954824002056485 IN IP4 192.168.1.143
s=FreeSWITCH
c=IN IP4 192.168.1.143
t=0 0
m=audio 22680 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
> maybe that phone does not support early media
>
> try adding the answer application to your dialplan
>
>
> On Mon, Mar 30, 2009 at 3:33 PM, <can_man at gmx.de> wrote:
>
> > Hallo,
> >
> > thank you for your answer Anthony.
> >
> > >
> > > starting at line 192 you seem to be sending yourself a notify, not
> sure
> > > how you did that.
> >
> > That is indeed strange, I have looked at the MjSip code but haven't
> found
> > the cause yet.
> >
> > > you are not by any chance trying to call a registered endpoint using
> the
> > > FS
> > > ip together with @ are you?
> > > say you fs box is 1.2.3.4 and the phone is registered as 1000
> > >
> > > If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you
> > > would
> > > use sofia/internal/1000%1.2.3.4
> > > The % tells it to resolve the domain as a locally hosted domain and
> > > translate it to the registered contact instead of using dns.
> > >
> >
> > For testing I at the moment send the incoming call to the voicemail of
> user
> > 1000 with this code:
> >
> > return '''<?xml version="1.0" encoding="UTF-8" standalone="no"?>\n'''\
> > '''<document type="freeswitch/xml">\n'''\
> > '''<section name="dialplan" description="RE Dial Plan For
> > FreeSwitch">\n'''\
> > '''<context name="public">\n'''\
> > '''<extension name="voicemail%s">\n'''\
> > '''<condition field="destination_number"
> expression="^(%s)$">\n'''\
> > '''<action application="voicemail" data="default $${domain}
> > %s"/>\n'''\
> > '''</condition>\n'''\
> > '''</extension>\n'''\
> > '''</context>\n'''\
> > '''</section>\n'''\
> > '''</document>''' % (didNumber, didNumber, id)
> >
> >
> > Works fine with a normal SIP client.
> > I have captured more output with debug enabled and have also captured
> the
> > SIP messages originating from MjSip.
> >
> > FS: http://pastebin.freeswitch.org/8045
> > MjSip: http://pastebin.freeswitch.org/8046
> >
> > Thank you very much for your help.
> > Best wishes,
> > Phil
> >
> > >
> > >
> > > On Sun, Mar 29, 2009 at 5:09 PM, <can_man at gmx.de> wrote:
> > >
> > > > Hello everyone,
> > > >
> > > > I am trying to get FS working with the MjSip Java Sip-stack, the
> > > SipToSis
> > > > source and the normal one. Everything works well within my own
> network
> > > and
> > > > when using x-lite, but when it comes to making calls from MjSip to
> an
> > > > outside FS server I don't hear any voice - seems to be a NAT problem
> or
> > > some
> > > > kind of other MjSip problem. Registration works fine though and SIP
> > > messages
> > > > get through ok, but non of the UDP RTP ones. Would be great if
> someone
> > > could
> > > > advice me on how to do the setup correctly.
> > > >
> > > > The whole FS trace can be found here:
> > > http://pastebin.freeswitch.org/8029
> > > >
> > > > The settings for MjSip are:
> > > >
> > > > "via_addr=91.101.58.142 (changed in the whole
> trace)","host_port=5090",
> > > > "transport_protocols=udp tcp","from_url=<sip:puli at 91.101.58.142:5090
> > >",
> > > >
> > > >
> > >
> >
> "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes",
> > > >
> > > >
> > >
> >
> "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068",
> > > >
> > > >
> > >
> >
> "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500",
> > > > "bin_rat=rat","bin_vic=vic"
> > > >
> > > >
> > > > Thank you very much.
> > > > Best wishes,
> > > > Phil
> > > >
> > > > --
> > > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate +
> > > > Telefonanschluss für nur 17,95 Euro/mtl.!*
> > > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a
> > > >
> > > > _______________________________________________
> > > > 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
> > > >
> > >
> > >
> > >
> > > --
> > > Anthony Minessale II
> > >
> > > FreeSWITCH http://www.freeswitch.org/
> > > ClueCon http://www.cluecon.com/
> > >
> > > AIM: anthm
> > > MSN:anthony_minessale at hotmail.com
> <MSN%3Aanthony_minessale at hotmail.com><
> >
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> > >
> > >
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> > >
> > > IRC: irc.freenode.net #freeswitch
> > >
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org
> <sip%3A888 at conference.freeswitch.org><
> >
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> > >
> > > iax:guest at conference.freeswitch.org/888
> > >
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> > >
> > > pstn:213-799-1400
> >
> > --
> > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate +
> > Telefonanschluss für nur 17,95 Euro/mtl.!*
> > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a
> >
> > _______________________________________________
> > 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
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
--
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