Hallo, thank you for your answer Anthony.
starting at line 192 you seem to be sending yourself a notify, not sure how you did that.
That is indeed strange, I have looked at the MjSip code but haven't found the cause yet.
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 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.
For testing I at the moment send the incoming call to the voicemail of user 1000 with this code: return '''<?xml version="1.0" encoding="UTF-8" standalone="no"?>\n'''\ '''<document type="freeswitch/xml">\n'''\ '''<section name="dialplan" description="RE Dial Plan For FreeSwitch">\n'''\ '''<context name="public">\n'''\ '''<extension name="voicemail%s">\n'''\ '''<condition field="destination_number" expression="^(%s)$">\n'''\ '''<action application="voicemail" data="default $${domain} %s"/>\n'''\ '''</condition>\n'''\ '''</extension>\n'''\ '''</context>\n'''\ '''</section>\n'''\ '''</document>''' % (didNumber, didNumber, id) Works fine with a normal SIP client. I have captured more output with debug enabled and have also captured the SIP messages originating from MjSip. FS: http://pastebin.freeswitch.org/8045 MjSip: http://pastebin.freeswitch.org/8046 Thank you very much for your help. Best wishes, Phil
On Sun, Mar 29, 2009 at 5:09 PM, <can_man@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@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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