[Freeswitch-users] FS - MjSip no voice

[Anthony Minessale]

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 at 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 at 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:puli at 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
>
> --
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-- 
Anthony Minessale II

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