Hi all, i've been using the service of a 3rd party hosted VoIP and thought I'd have a try with by own PBX. As all our phones are Snom's the most logical choice was the Snom ONE pbx!
Now all has been going fine but i've got a problem calling one of the extenstion, it rings, i can answer it but some times there is just no audio and the other phone just remains on "183 Session Progress" I then get an email sayiny snom PBX: User disconnects call:
Rx: udp:86.28.219.71:1217 (1164 bytes)
INVITE sip:1001@77.240.1.41;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:1217;branch=z9hG4bK-j3zinuhxiti1;rport
From: <sip:1002@77.240.1.41>;tag=4wtli7le39
To: <sip:1001@77.240.1.41;user=phone>
Call-ID: 3c26823595d8-dtbbpvekty9n
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:1002@192.168.0.5:1217;line=8vhw1eju>;reg-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom370/7.3.30
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 450
v=0
o=root 467526577 467526577 IN IP4 192.168.0.5
s=call
c=IN IP4 192.168.0.5
t=0 0
m=audio 59200 RTP/AVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZjUEK2WhjujAVLxYtuQnWpV4eImHuRSd9X0wY7yW
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Any one got any pointers for me please? It always seems to be ext 1002 calling 1001 - I don't *think* i've had the issues 1001 calling 1002.
Dan
