[SR-Users] rtpproxy and kamailio doesnt work out from the box.

MingHon gminghon at gmail.com
Tue Jun 28 04:25:01 CEST 2011


Hi,

i registered 3 uac behind same nat successfully but when i try to call each
other i didnt get any audio.
but if i use uac 102 and 103 to call into the voicemail i heard the audio
but not for 101.
kamailio is listening 60.48.218.61 and 192.168.2.3
rtpproxy is running.
asterisk is at 192.168.2.23.

here is my ul show.

AOR:: 102
                Contact:: sip:102 at 175.136.221.60:5062 Q=
                        Expires:: 3110
                        Callid:: 721498432 at 175.136.221.60
                        Cseq:: 2
                        User-agent:: T22 7.3.0.50
                        Received:: sip:175.136.221.60:1024
                        State:: CS_SYNC
                        Flags:: 0
                        Cflag:: 192
                        Socket:: udp:60.48.218.61:5060
                        Methods:: 16383
        AOR:: 103
                Contact:: sip:103 at 175.136.221.60:5062 Q=
                        Expires:: 3114
                        Callid:: 1499738216 at 175.136.221.60
                        Cseq:: 2
                        User-agent:: Yealink SIP-T18 18.0.0.70
                        Received:: sip:175.136.221.60:1025
                        State:: CS_SYNC
                        Flags:: 0
                        Cflag:: 192
                        Socket:: udp:60.48.218.61:5060
                        Methods:: 16383
        AOR:: 101
                Contact:: sip:101 at 175.136.221.60:5062 Q=
                        Expires:: 3097
                        Callid:: 166053301 at 175.136.221.60
                        Cseq:: 2
                        User-agent:: T20 9.41.0.80
                        State:: CS_SYNC
                        Flags:: 0
                        Cflag:: 0
                        Socket:: udp:60.48.218.61:5060
                        Methods:: 16383

and may i know why uac 101 did not have the received: field?

please some one could give a hand on this? the audio really cant get thru i
really have no idea.

thank you

-- 
Regards,

MingHon
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20110628/1f54f30a/attachment.htm>


More information about the sr-users mailing list