maybe that phone does not support early media<br><br>try adding the answer application to your dialplan<br><br><br><div class="gmail_quote">On Mon, Mar 30, 2009 at 3:33 PM, <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;">Hallo,<br>
<br>
thank you for your answer Anthony.<br>
<div class="im"><br>
><br>
> starting at line 192 you seem to be sending yourself a notify, not sure<br>
> how you did that.<br>
<br>
</div>That is indeed strange, I have looked at the MjSip code but haven't found the cause yet.<br>
<div class="im"><br>
> you are not by any chance trying to call a registered endpoint using 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>
</div>For testing I at the moment send the incoming call to the voicemail of user 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 FreeSwitch">\n'''\<br>
'''<context name="public">\n'''\<br>
'''<extension name="voicemail%s">\n'''\<br>
'''<condition field="destination_number" expression="^(%s)$">\n'''\<br>
'''<action application="voicemail" data="default $${domain} %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 the 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>
<div><div></div><div class="h5"><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 network<br>
> and<br>
> > when using x-lite, but when it comes to making calls from MjSip to an<br>
> > outside FS server I don't hear any voice - seems to be a NAT problem 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 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 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>
> "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes",<br>
> ><br>
> ><br>
> "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068",<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>
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> ><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>
</div></div>> <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>
<div class="im">> IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
><br>
> FreeSWITCH Developer Conference<br>
</div>> <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>
<div class="im">> <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
</div>> <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>
<font color="#888888"><br>
--<br>
</font><div><div></div><div class="h5">Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss für nur 17,95 Euro/mtl.!* <a href="http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a" target="_blank">http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a</a><br>
<br>
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</div></div></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>