VoIP Mailing List Archives

[can_man at gmx.de]

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@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 at g...]

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/1000@1.2.3.4 (1000@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_man@gmx.de (can_man@gmx.de)> wrote:

[can_man at gmx.de]

Hallo,

thank you for your answer Anthony.

[anthony.minessale at g...]

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@gmx.de (can_man@gmx.de)> wrote: