You should press f8 to get more detailed output from FS. you also should capture more of the call,
starting at line 192 you seem to be sending yourself a notify, not sure how you did that. 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...@1.2.3.4 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. otherwise, enable debugging with f8 and reproduce your issue and capture *all* the output. 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> 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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