Remote Phone having trouble making call

[craigreilly]

I plan on doing a remote session with this user later, to see if something is wrong in the phone settings. But maybe someone can shed some light. I see that no digits are coming across in the log. But user says they dial the digits then DIAL. Extension is using an Asastra 6753i (I know not officially supported but Templates are available to download in v11)
Any help would be appreciated.

15-08-2012 08:04:36.408 L:3075.1[Extn]: Terminating targets, reason:
15-08-2012 08:04:36.408 Leg L:3075.1[Extn] is terminated: Cause: 200 BYE from PBX
15-08-2012 08:04:36.408 L:3075.1[Extn] Sending: OnSendResp Send 404/INVITE from 0.0.0.0:0 tid=d474231724ced1831 Call-ID=2c2703493b958095:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 76.176.140.44:5196;branch=z9hG4bKd474231724ced1831;rport=5196
To: <sip:66.x.x.112:5060;user=phone>;tag=181e0919
From: "4929"<sip:[email protected]:5060>;tag=0e4dcfdc10
Call-ID: 2c2703493b958095
CSeq: 15069 INVITE
Warning: 499 AAVC-PHONE "Not found"
Content-Length: 0
15-08-2012 08:04:36.384 [CM503020]: Call(C:3075): Normal call termination. Call originator: Extn:4929. Reason: Not found
15-08-2012 08:04:36.384 [CM503016]: Call(C:3075): Attempt to reach <sip66.x.x.112:5060> from Extn:4929 has failed. Reason: Not Found
15-08-2012 08:04:36.384 [CM503014]: Call(C:3075): No known route from Extn:4929 to target: <sip66.x.x.112:5060>
15-08-2012 08:04:36.384 [CM503010]: Call(C:3075): Making route(s) from Extn:4929 to <sip66.x.x.112:5060>
15-08-2012 08:04:36.384 Remote SDP is set for leg L:3075.1[Extn]
15-08-2012 08:04:36.382 [CM500002]: Call(C:3075): Info on incoming INVITE from Extn:4929:
Invite-IN Recv Req INVITE from 76.x.x.44:5196 tid=d474231724ced1831 Call-ID=2c2703493b958095:
INVITE sip:66.x.x.112:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 76.x.x.44:5196;branch=z9hG4bKd474231724ced1831;rport=5196
Max-Forwards: 70
Route: <sip:66.x.x.112;lr>
Contact: "4929"<sip:[email protected]:5196;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D18DDFA>"
To: <sip:66.x.x.112:5060;user=phone>
From: "4929"<sip:[email protected]:5060>;tag=0e4dcfdc10
Call-ID: 2c2703493b958095
CSeq: 15069 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="4929",realm="3CXPhoneSystem",nonce="414d535c063c4b8443:f5f9dd204ae7145137aa97479124296b",uri="sip:66.x.x.112:5060;user=phone",response="cd6220c01c6ca45e5cf2d1686d4fca96",algorithm=MD5
Supported: path, 100rel, replaces
User-Agent: Aastra 53i/3.2.2.2044
Allow-Events: aastra-xml, talk, hold, conference, LocalModeStatus
Content-Length: 309

v=0
o=MxSIP 0 1 IN IP4 76.176.140.44
s=SIP Call
c=IN IP4 76.176.140.44
t=0 0
m=audio 3000 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=silenceSuppff - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
15-08-2012 08:04:36.382 [CM503001]: Call(C:3075): Incoming call from Extn:4929 to <sip66.xxx.xxx.112:5060>

 

[craigreilly]

By the way, I have another site using the same setup without issue, so I am guessing it might be a Firewall Issue/DSL Router at the remote site from the local TelCo.

 

[lneblett]

What I don't see is the destination:
Contact: "4929"<sip:[email protected]:5196;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D18DDFA>"
To: <sip:66.x.x.112:5060;user=phone>
From: "4929"<sip:[email protected]:5060>;tag=0e4dcfdc10

Where are they really trying to get to?

 

[craigreilly]

He says he is dialing his cell phone a 480xxxxx87
Now, as a test I tried changing 'PBX Delivers Audio' and I see 3 of the digits he is dialing - 480.

So that leads me to a firewall issue, since I have another Aastra using a different router (Netgear N300) working just fine in a test environment.

 

[lneblett]

So, he is dialing from external extension 4929 trying to be routed out to so that he can reach an external number 480XXXXXXX; can he reach an internal extension?

 

[craigreilly]

Another DOH! Moment. :shock:
For some reason I set the registration under Global SIP instead of Line 1. I may have been in a hurry the day I set it up and put it in the wrong place. So when he tried to make a call, no line was being used... I don't really understand how the Aastras work. How the Global vs Line 1-6 works. On the Yealinks it is just Account 1, Account 2, etc.

 

[craigreilly]

Oh and if I had used the template no problem. It puts the info into Line 1 automatically. But I programmed it manually because 3cx doesn't support remote registration for that handset.