Am trying to connect via Direct SIP using SJPhone SIP client. Looking at the 3CX activity log, the SIP INVITE gets to the 3CX system, but it's wanting an authentication? Here is the log entry:
11-Jan-2016 08:49:33.532 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
Invite-UNK Recv Req INVITE from 10.1.243.84:5060 tid=0a01f3540000006e5693dd1e00004cf100000016 Call-ID=4097048036014CE7AFF8A6999D25A5B00x0a01f354:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.243.84;branch=z9hG4bK0a01f3540000006e5693dd1e00004cf100000016;rport=5060
Max-Forwards: 70
Contact: <sip:10.1.243.84>
To: <sip:[email protected]>
From: "John Doe"<sip:10.1.243.84>;tag=92e85f8d75a
Call-ID: 4097048036014CE7AFF8A6999D25A5B00x0a01f354
CSeq: 1 INVITE
Content-Type: application/sdp
Supported: replaces, norefersub, timer
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 362
v=0
o=- 3661519773 3661519773 IN IP4 10.1.243.84
s=SJphone
c=IN IP4 10.1.243.84
t=0 0
m=audio 49164 RTP/AVP 3 97 98 8 0 101
c=IN IP4 10.1.243.84
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv
On the 3CX, I have simply created an extension (1001). No phone is registered or connected to that extension at this time. Under network setting -> FQDN, I have gc.com as the local SIP domain and I have checked the "Allow calls from/to external SIP URIs). I also have a DNS A record for sip.gc.com that points to the 3CX IP and a SRV record for gc.com that's pointing to the 3CX IP per http://www.3cx.com/blog/voip-howto/direct-sip/.
Does anyone know why this is not working?
11-Jan-2016 08:49:33.532 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
Invite-UNK Recv Req INVITE from 10.1.243.84:5060 tid=0a01f3540000006e5693dd1e00004cf100000016 Call-ID=4097048036014CE7AFF8A6999D25A5B00x0a01f354:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.243.84;branch=z9hG4bK0a01f3540000006e5693dd1e00004cf100000016;rport=5060
Max-Forwards: 70
Contact: <sip:10.1.243.84>
To: <sip:[email protected]>
From: "John Doe"<sip:10.1.243.84>;tag=92e85f8d75a
Call-ID: 4097048036014CE7AFF8A6999D25A5B00x0a01f354
CSeq: 1 INVITE
Content-Type: application/sdp
Supported: replaces, norefersub, timer
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 362
v=0
o=- 3661519773 3661519773 IN IP4 10.1.243.84
s=SJphone
c=IN IP4 10.1.243.84
t=0 0
m=audio 49164 RTP/AVP 3 97 98 8 0 101
c=IN IP4 10.1.243.84
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv
On the 3CX, I have simply created an extension (1001). No phone is registered or connected to that extension at this time. Under network setting -> FQDN, I have gc.com as the local SIP domain and I have checked the "Allow calls from/to external SIP URIs). I also have a DNS A record for sip.gc.com that points to the 3CX IP and a SRV record for gc.com that's pointing to the 3CX IP per http://www.3cx.com/blog/voip-howto/direct-sip/.
Does anyone know why this is not working?
