Howto make OBi accept SIP registration
weiqj:
Set it to ph, same problem.
So that tutorial doesn't work?
RonR:
weiqj,
I'm not an expert down at the SIP protocol level. Looking at the syslog entry you provided, it's not clear to me what the OBi doesn't like about the INVITE from the 3CX server. It's very similar to calls successfully initiated from my PAP2 to the OBi. Normally, such calls get routed via the InboundCallRoute, which in this case, should ring the PHONE Port.
weiqj:
Appreciate your help very much. Below is everything recorded from syslog, including the request. I have no idea why it won't work. (I only replaced the phone number with XXXXXXXXXX).
As a side note, OBi doesn't provide a way to trigger in response to events (incoming call, phone status etc). It is a feature some people, including me, are looking for. For example, they would want to automatically pause XBMC playback on incoming phone call.
Somebody wrote a perl script to keep track of the events. But it sends HTTP query for port status XML every second, which puts a lot of load on both client and OBi box.
I think the perfect solution is to watch syslog, it is triggered by UDP packets and will be the best way to solve the problem. I may build something like when I have time and share it with people. Now it's just some thoughts.
<7> RxFrom:c0a80bfe:5060
INVITE sip:XXXXXXXXXX@192.168.11.26:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.254:5060;branch=z9hG4bK-d8754z-e9620904787a5d65-1---
d8754z-;rport
Max-Forwards: 70
Contact: <sip:10001@192.168.11.254:5060>
To: <sip:XXXXXXXXXX@192.168.11.26:5061>
From: "Home"<sip:10001@192.168.11.254:5060>;tag=8f7e9d6d
Call-ID: YWE4MjNlZjRiZDcwMTdkMDhjNzk1ZGY4Y2ZlNjE3ZWQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, IN
FO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 10.0.22539.0
Content-Length: 410
v=0
o=- 12969595059572250 1 IN IP4 192.168.11.13
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.11.13
t=0 0
a=ice-ufrag:54a623
a=ice-pwd:87158d76464039cf541a5507397aa163
m=audio 55036 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.11.13 55036 typ host
a=candidate:1 2 UDP 659134 192.168.11.13 55037 typ host
<7> sendto c0a80bfe:5060(337)
SIP/2.0 404 Not Found
Call-ID: YWE4MjNlZjRiZDcwMTdkMDhjNzk1ZGY4Y2ZlNjE3ZWQ.
CSeq: 1 INVITE
Content-Length: 0
From: "Dad"<sip:10001@192.168.11.254:5060>;tag=8f7e9d6d
To: <sip:XXXXXXXXXX@192.168.11.26:5061>
Via: SIP/2.0/UDP 192.168.11.254:5060;branch=z9hG4bK-d8754z-e9620904787a5d65-1---
d8754z-;received=192.168.11.254;rport=5060
<7> RxFrom:c0a80bfe:5060
ACK sip:XXXXXXXXXX@192.168.11.26:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.254:5060;branch=z9hG4bK-d8754z-e9620904787a5d65-1---
d8754z-;rport
Max-Forwards: 70
To: <sip:XXXXXXXXXX@192.168.11.26:5061>
From: "Home"<sip:10001@192.168.11.254:5060>;tag=8f7e9d6d
Call-ID: YWE4MjNlZjRiZDcwMTdkMDhjNzk1ZGY4Y2ZlNjE3ZWQ.
CSeq: 1 ACK
Content-Length: 0
weiqj:
Thanks a lot for your help. It now works. Problem is that II didn't have a chance to check what did I do wrong. It just worked all of a sudden.
joseacarras:
Could you provide the final configuration you use so anyone could replicate it on the future?
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