like i said:
> maybe that phone does not support early media
>
> try adding the answer application to your dialplan
early media == 183
answer = 200
it depends on your dialplan in FS
On Tue, Mar 31, 2009 at 9:06 AM, <can_...@gmx.de> wrote:
> 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:p...@90.181.59.141:5090>;tag=z9hG4bK40977269
> To: <sip:2...@90.181.59.141:5090>;tag=vgg3Zja8pNQcg
> Call-ID: 507347917...@90.181.59.141
> CSeq: 1 INVITE
> Contact: <sip:mod_so...@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:br...@192.168.1.143 <sip%3abr...@192.168.1.143>
> >;tag=z9hG4bK42598163
> To: <sip:1...@192.168.1.143 <sip%3a1...@192.168.1.143>>;tag=Q0X494ZUNaKHH
> Call-ID: 961142687...@192.168.1.143
> CSeq: 2 INVITE
> Contact: <sip:mod_so...@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_...@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/1...@1.2.3.4you
> > > > 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_...@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:p...@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@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_miness...@hotmail.com<msn%3aanthony_miness...@hotmail.com>
> > <msn%3aanthony_miness...@hotmail.com<msn%253aanthony_miness...@hotmail.com>
> ><
> > >
> > msn%3aanthony_miness...@hotmail.com<msn%253aanthony_miness...@hotmail.com>
> <msn%253aanthony_miness...@hotmail.com<msn%25253aanthony_miness...@hotmail.com>
> >
> > > >
> > > >
> > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com>
> <paypal%3aanthony.miness...@gmail.com<paypal%253aanthony.miness...@gmail.com>
> >
> > >
> > <paypal%3aanthony.miness...@gmail.com<paypal%253aanthony.miness...@gmail.com>
> <paypal%253aanthony.miness...@gmail.com<paypal%25253aanthony.miness...@gmail.com>
> >
> > > >
> > > > IRC: irc.freenode.net #freeswitch
> > > >
> > > > FreeSWITCH Developer Conference
> > > > sip:8...@conference.freeswitch.org<sip%3a...@conference.freeswitch.org>
> > <sip%3a...@conference.freeswitch.org<sip%253a...@conference.freeswitch.org>
> ><
> > >
> > sip%3a...@conference.freeswitch.org<sip%253a...@conference.freeswitch.org>
> <sip%253a...@conference.freeswitch.org<sip%25253a...@conference.freeswitch.org>
> >
> > > >
> > > > iax:gu...@conference.freeswitch.org/888
> > > >
> > googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org>
> <googletalk%3aconf%2b...@conference.freeswitch.org<googletalk%253aconf%252b...@conference.freeswitch.org>
> >
> > >
> > <googletalk%3aconf%2b...@conference.freeswitch.org<googletalk%253aconf%252b...@conference.freeswitch.org>
> <googletalk%253aconf%252b...@conference.freeswitch.org<googletalk%25253aconf%25252b...@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@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_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com><
> msn%3aanthony_miness...@hotmail.com<msn%253aanthony_miness...@hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com>
> <paypal%3aanthony.miness...@gmail.com<paypal%253aanthony.miness...@gmail.com>
> >
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org><
> sip%3a...@conference.freeswitch.org<sip%253a...@conference.freeswitch.org>
> >
> > iax:gu...@conference.freeswitch.org/888
> > googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org>
> <googletalk%3aconf%2b...@conference.freeswitch.org<googletalk%253aconf%252b...@conference.freeswitch.org>
> >
> > pstn:213-799-1400
>
> --
> Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen:
> http://www.gmx.net/de/go/multimessenger01
>
> _______________________________________________
> Freeswitch-users mailing list
> 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
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org>
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org>
pstn:213-799-1400
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