like i said:<br>> maybe that phone does not support early media<br>
><br>
> try adding the answer application to your dialplan<br><br>early media == 183<br>answer = 200<br><br>it depends on your dialplan in FS<br><br><br><br><div class="gmail_quote">On Tue, Mar 31, 2009 at 9:06 AM, <span dir="ltr"><<a href="mailto:can_man@gmx.de">can_man@gmx.de</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hello,<br>
<br>
I have found the problem. FS on my local network sends "SIP/2.0 200 OK"<br>
after an invite and FS on the net through the external profil sends<br>
SIP/2.0 183 Session Progress. But MjSip doesn't know how to deal with<br>
183, so it just ignores the message. For testing I have changed<br>
the 183 header to the 200 one and now it works.<br>
<br>
Thank you for your help and the quick response time.<br>
Best wishes,<br>
Phil<br>
<br>
<br>
>From FS on the net through the external profil:<br>
<br>
SIP/2.0 183 Session Progress<br>
Via: SIP/2.0/UDP 90.181.59.141:5090;rport=60315;branch=z9hG4bK256321;received=78.105.17.88<br>
From: <<a href="http://sip:puli@90.181.59.141:5090" target="_blank">sip:puli@90.181.59.141:5090</a>>;tag=z9hG4bK40977269<br>
To: <<a href="http://sip:2345@90.181.59.141:5090" target="_blank">sip:2345@90.181.59.141:5090</a>>;tag=vgg3Zja8pNQcg<br>
Call-ID: <a href="mailto:507347917247@90.181.59.141">507347917247@90.181.59.141</a><br>
CSeq: 1 INVITE<br>
Contact: <sip:mod_sofia@90.181.59.141:5090;transport=udp><br>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12839M<br>
Accept: application/sdp<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, refer<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 267<br>
<br>
v=0<br>
o=FreeSWITCH 1072777625698755085 8893522831081357051 IN IP4 90.181.59.141<br>
s=FreeSWITCH<br>
c=IN IP4 91.121.59.148<br>
t=0 0<br>
m=audio 26722 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
<br>
<br>
<br>
>From FS in my local network:<br>
<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 192.168.1.143:5060;rport=5060;branch=z9hG4bK423233;received=192.168.1.102<br>
From: <<a href="mailto:sip%3Abrian@192.168.1.143">sip:brian@192.168.1.143</a>>;tag=z9hG4bK42598163<br>
To: <<a href="mailto:sip%3A1000@192.168.1.143">sip:1000@192.168.1.143</a>>;tag=Q0X494ZUNaKHH<br>
Call-ID: <a href="mailto:961142687222@192.168.1.143">961142687222@192.168.1.143</a><br>
CSeq: 2 INVITE<br>
Contact: <sip:mod_sofia@192.168.1.143:5060;transport=udp><br>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12712M<br>
Accept: application/sdp<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br>
Session-Expires: 120;refresher=uas<br>
Min-SE: 120<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 267<br>
<br>
v=0<br>
o=FreeSWITCH 5195745633884389954 8941954824002056485 IN IP4 192.168.1.143<br>
s=FreeSWITCH<br>
c=IN IP4 192.168.1.143<br>
t=0 0<br>
m=audio 22680 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
<br>
<br>
<br>
<br>
> maybe that phone does not support early media<br>
><br>
> try adding the answer application to your dialplan<br>
><br>
><br>
> On Mon, Mar 30, 2009 at 3:33 PM, <<a href="mailto:can_man@gmx.de">can_man@gmx.de</a>> wrote:<br>
><br>
> > Hallo,<br>
> ><br>
> > thank you for your answer Anthony.<br>
> ><br>
> > ><br>
> > > starting at line 192 you seem to be sending yourself a notify, not<br>
> sure<br>
> > > how you did that.<br>
> ><br>
> > That is indeed strange, I have looked at the MjSip code but haven't<br>
> found<br>
> > the cause yet.<br>
> ><br>
> > > you are not by any chance trying to call a registered endpoint using<br>
> the<br>
> > > FS<br>
> > > ip together with @ are you?<br>
> > > say you fs box is 1.2.3.4 and the phone is registered as 1000<br>
> > ><br>
> > > If you want to call 1000 you don't use sofia/internal/<a href="mailto:1000@1.2.3.4">1000@1.2.3.4</a> you<br>
> > > would<br>
> > > use sofia/internal/1000%1.2.3.4<br>
> > > The % tells it to resolve the domain as a locally hosted domain and<br>
> > > translate it to the registered contact instead of using dns.<br>
> > ><br>
> ><br>
> > For testing I at the moment send the incoming call to the voicemail of<br>
> user<br>
> > 1000 with this code:<br>
> ><br>
> > return '''<?xml version="1.0" encoding="UTF-8" standalone="no"?>\n'''\<br>
> > '''<document type="freeswitch/xml">\n'''\<br>
> > '''<section name="dialplan" description="RE Dial Plan For<br>
> > FreeSwitch">\n'''\<br>
> > '''<context name="public">\n'''\<br>
> > '''<extension name="voicemail%s">\n'''\<br>
> > '''<condition field="destination_number"<br>
> expression="^(%s)$">\n'''\<br>
> > '''<action application="voicemail" data="default $${domain}<br>
> > %s"/>\n'''\<br>
> > '''</condition>\n'''\<br>
> > '''</extension>\n'''\<br>
> > '''</context>\n'''\<br>
> > '''</section>\n'''\<br>
> > '''</document>''' % (didNumber, didNumber, id)<br>
> ><br>
> ><br>
> > Works fine with a normal SIP client.<br>
> > I have captured more output with debug enabled and have also captured<br>
> the<br>
> > SIP messages originating from MjSip.<br>
> ><br>
> > FS: <a href="http://pastebin.freeswitch.org/8045" target="_blank">http://pastebin.freeswitch.org/8045</a><br>
> > MjSip: <a href="http://pastebin.freeswitch.org/8046" target="_blank">http://pastebin.freeswitch.org/8046</a><br>
> ><br>
> > Thank you very much for your help.<br>
> > Best wishes,<br>
> > Phil<br>
> ><br>
> > ><br>
> > ><br>
> > > On Sun, Mar 29, 2009 at 5:09 PM, <<a href="mailto:can_man@gmx.de">can_man@gmx.de</a>> wrote:<br>
> > ><br>
> > > > Hello everyone,<br>
> > > ><br>
> > > > I am trying to get FS working with the MjSip Java Sip-stack, the<br>
> > > SipToSis<br>
> > > > source and the normal one. Everything works well within my own<br>
> network<br>
> > > and<br>
> > > > when using x-lite, but when it comes to making calls from MjSip to<br>
> an<br>
> > > > outside FS server I don't hear any voice - seems to be a NAT problem<br>
> or<br>
> > > some<br>
> > > > kind of other MjSip problem. Registration works fine though and SIP<br>
> > > messages<br>
> > > > get through ok, but non of the UDP RTP ones. Would be great if<br>
> someone<br>
> > > could<br>
> > > > advice me on how to do the setup correctly.<br>
> > > ><br>
> > > > The whole FS trace can be found here:<br>
> > > <a href="http://pastebin.freeswitch.org/8029" target="_blank">http://pastebin.freeswitch.org/8029</a><br>
> > > ><br>
> > > > The settings for MjSip are:<br>
> > > ><br>
> > > > "via_addr=91.101.58.142 (changed in the whole<br>
> trace)","host_port=5090",<br>
> > > > "transport_protocols=udp tcp","from_url=<<a href="http://sip:puli@91.101.58.142:5090" target="_blank">sip:puli@91.101.58.142:5090</a><br>
> > >",<br>
> > > ><br>
> > > ><br>
> > ><br>
> ><br>
> "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes",<br>
> > > ><br>
> > > ><br>
> > ><br>
> ><br>
> "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068",<br>
> > > ><br>
> > > ><br>
> > ><br>
> ><br>
> "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500",<br>
> > > > "bin_rat=rat","bin_vic=vic"<br>
> > > ><br>
> > > ><br>
> > > > Thank you very much.<br>
> > > > Best wishes,<br>
> > > > Phil<br>
> > > ><br>
> > > > --<br>
> > > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate +<br>
> > > > Telefonanschluss für nur 17,95 Euro/mtl.!*<br>
> > > > <a href="http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a" target="_blank">http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a</a><br>
> > > ><br>
> > > > _______________________________________________<br>
> > > > Freeswitch-users mailing list<br>
> > > > <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>
> > > > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
> > > > UNSUBSCRIBE:<br>
> > <a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
> > > > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
> > > ><br>
> > ><br>
> > ><br>
> > ><br>
> > > --<br>
> > > Anthony Minessale II<br>
> > ><br>
> > > FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
> > > ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
> > ><br>
> > > AIM: anthm<br>
> > > <a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>
> <<a href="mailto:MSN%253Aanthony_minessale@hotmail.com">MSN%3Aanthony_minessale@hotmail.com</a>><<br>
> ><br>
> <a href="mailto:MSN%253Aanthony_minessale@hotmail.com">MSN%3Aanthony_minessale@hotmail.com</a><<a href="mailto:MSN%25253Aanthony_minessale@hotmail.com">MSN%253Aanthony_minessale@hotmail.com</a>><br>
> > ><br>
> > ><br>
> GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><<a href="mailto:PAYPAL%253Aanthony.minessale@gmail.com">PAYPAL%3Aanthony.minessale@gmail.com</a>><br>
> ><br>
> <<a href="mailto:PAYPAL%253Aanthony.minessale@gmail.com">PAYPAL%3Aanthony.minessale@gmail.com</a><<a href="mailto:PAYPAL%25253Aanthony.minessale@gmail.com">PAYPAL%253Aanthony.minessale@gmail.com</a>><br>
> > ><br>
> > > IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
> > ><br>
> > > FreeSWITCH Developer Conference<br>
> > > <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br>
> <<a href="mailto:sip%253A888@conference.freeswitch.org">sip%3A888@conference.freeswitch.org</a>><<br>
> ><br>
> <a href="mailto:sip%253A888@conference.freeswitch.org">sip%3A888@conference.freeswitch.org</a><<a href="mailto:sip%25253A888@conference.freeswitch.org">sip%253A888@conference.freeswitch.org</a>><br>
> > ><br>
> > > <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
> > ><br>
> <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><<a href="mailto:googletalk%253Aconf%252B888@conference.freeswitch.org">googletalk%3Aconf%2B888@conference.freeswitch.org</a>><br>
> ><br>
> <<a href="mailto:googletalk%253Aconf%252B888@conference.freeswitch.org">googletalk%3Aconf%2B888@conference.freeswitch.org</a><<a href="mailto:googletalk%25253Aconf%25252B888@conference.freeswitch.org">googletalk%253Aconf%252B888@conference.freeswitch.org</a>><br>
> > ><br>
> > > pstn:213-799-1400<br>
> ><br>
> > --<br>
> > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate +<br>
> > Telefonanschluss für nur 17,95 Euro/mtl.!*<br>
> > <a href="http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a" target="_blank">http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a</a><br>
> ><br>
> > _______________________________________________<br>
> > Freeswitch-users mailing list<br>
> > <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>
> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
> > UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
> ><br>
><br>
><br>
><br>
> --<br>
> Anthony Minessale II<br>
><br>
> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
> ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
><br>
> AIM: anthm<br>
> <a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a> <<a href="mailto:MSN%253Aanthony_minessale@hotmail.com">MSN%3Aanthony_minessale@hotmail.com</a>><br>
> GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><<a href="mailto:PAYPAL%253Aanthony.minessale@gmail.com">PAYPAL%3Aanthony.minessale@gmail.com</a>><br>
> IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
><br>
> FreeSWITCH Developer Conference<br>
> <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a> <<a href="mailto:sip%253A888@conference.freeswitch.org">sip%3A888@conference.freeswitch.org</a>><br>
> <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
> <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><<a href="mailto:googletalk%253Aconf%252B888@conference.freeswitch.org">googletalk%3Aconf%2B888@conference.freeswitch.org</a>><br>
> pstn:213-799-1400<br>
<br>
--<br>
Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: <a href="http://www.gmx.net/de/go/multimessenger01" target="_blank">http://www.gmx.net/de/go/multimessenger01</a><br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
<br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br>
<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400<br>