Unable to place a call

[robertofmlx]

I have a phone in our network that is unable to place a call. The client is a regular phone connected to a Linksys SPA2102. There are two phone on that box and line 2 works fine, line 1 fails when placing calls. On the other hand, line 1 is able to register just fine and can receive calls. Out of my whits!

I looked at the server and it looks like a proxy-authentication header is missing when line 1 places a call, but I have checked both client and server sides and both line 1 and 2 are the same (other than some basic info). Can someone give me a clue?

As you can see, the call is internally from extension to extension in this case. Needless to say calls to external #s fail too.

TIA

13:14:11.309 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.123.11:5060;branch=z9hG4bK-5b2cfca1
Max-Forwards: 70
Contact: "Anonymous"<sip:[email protected]:5060>
To: <sip:[email protected]>
From: "Anonymous"<sip:anonymous@localhost>;tag=f992913e6f2bf999o0
Call-ID: d08441e2-ef536145@localhost
CSeq: 101 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 0
Remote-Party-ID: Real UserName <sip:[email protected]>;screen=yes;privacy=full;party=calling

 

[leejor]

Are you certain that the second line has port 5061 (or at least NOT 5060) assigned?

 

[robertofmlx]

Yes, the two lines go against 5061 (in fact, I changed line 1 from 5060 to 5061 because line 2 worked). The weird thing is that all the other boxes go to 5060. But why would Line 2 work?? I'll run a small test on this.

 

[leejor]

When you have a SIP "box" (this could be a Gateway or an ATA) with two or more devices/lines/trunks inside, it is common to address each with its own port number as the "box" has only one IP. 3CX sees each as unique as it will register as the IP + port number. I'm not sure what else you may have changed on the ATA, but, if you have made a number of changes, and didn't keep track, you may want to consider a 'factory" reset and fresh start.

With the 2102, from a factory fresh start, the only differences between the two lines will be the extension number/name and password. (that assumes you have left the ports alone and line 1 is at 5060 and line 2 is at 5061) The dialplan can be cut and pasted from the working line to the other. RTP Packet Size can be changed to 0.020 and you may want to consider giving the device a fixed IP, if you haven't already. NTP server(s) can be datafilled to provide the correct time/date on CID of incoming calls.

That should give you a basic working ATA. Other settings may require changes for additional features, or regional differences as the defaults (depending on the version) are generally for north America.

 

[robertofmlx]

Thanks for the explanation Leejor... good to learn something new. As you might guess, I'm sort of a part time IT "department" at my company. I'm back at it!

I changed line 1 to 5060 and line 2 remained on 5061. I still have the same problems.

Not sure if this helps, but the bad line (line 1) is able to receive calls, but not place them. Also, on the server side things look fine. I'm placing the good (x140) and bad (x138) line invites below to see if you can spot anything obvious. One difference is that the call-id/from/call-id revolved around anonymous and localhost. Could that be the problem? How do I change that?

Good phone to x117
12:40:54.045 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.123.11:5060;branch=z9hG4bK-2ad45411
Max-Forwards: 70
Contact: "John Durian"<sip:[email protected]:5060>
To: <sip:[email protected]>
From: "John Durian"<sip:[email protected]>;tag=6236d3c25d0e4cc5o1
Call-ID: [email protected]
CSeq: 102 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Proxy-Authorization: Digest username="140",realm="3CXPhoneSystem",nonce="414d535c08e9fc2584:f2c916a593034bb48e5109cbed046a9c",uri="sip:[email protected]",algorithm=MD5,response="212b43a9dc9d800ec20fa8aeb9592577"
Supported: x-sipura
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 0
Remote-Party-ID: John Durian <sip:[email protected]>;screen=yes;party=calling


Bad phone to x117
12:48:43.170 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.123.11:5060;branch=z9hG4bK-c02e21f5
Max-Forwards: 70
Contact: "Anonymous"<sip:[email protected]:5060>
To: <sip:[email protected]>
From: "Anonymous"<sip:anonymous@localhost>;tag=285c711531feeff4o0
Call-ID: 5545cc69-757b2cb0@localhost
CSeq: 102 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Proxy-Authorization: Digest username="138",realm="3CXPhoneSystem",nonce="414d535c08e9fdfb63:33f4762c7273d43a808872e1fe580e02",uri="sip:[email protected]",algorithm=MD5,response="ac1988d19a8db126e22e72a5407b9877"
Supported: x-sipura
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 0
Remote-Party-ID: Mi Johnson <sip:[email protected]>;screen=yes;privacy=full;party=calling

12:48:43.045 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.123.11:5060;branch=z9hG4bK-190e0859
Max-Forwards: 70
Contact: "Anonymous"<sip:[email protected]:5060>
To: <sip:[email protected]>
From: "Anonymous"<sip:anonymous@localhost>;tag=285c711531feeff4o0
Call-ID: 5545cc69-757b2cb0@localhost
CSeq: 101 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 0
Remote-Party-ID: Mi Johnson <sip:[email protected]>;screen=yes;privacy=full;party=calling

12:48:43.045 [CM302001]: Authorization system can not identify source of: SipReq: INVITE [email protected] tid=-190e0859 cseq=INVITE [email protected]:5060 / 101 from(wire)

 

[leejor]

I suspect that there is still something amiss in the settings for Line 2

Contact: "John Durian"<sip:140@192.168.123.11:5060>
To: <sip:[email protected]>

This line, in the successful call should look the same as a call from line 2 with the exception of the highlighted items. The one that really concerns me is the port number. it should be showing as 5061.

If you power down the ATA, then re-power and check the 3Cx log, does it show both extensions registering correctly? The extension numbers should be there along with the same IP but each using a different port.

I'm not certain if some other features have been modified on the ATA and that is causing the issue.

You may want to consider a factory reset, and start from scratch, this time only changing the server name, extension number(s) and password(s), to start with. Or, you can post the Line 2 settings.

 

[robertofmlx]

Got it! I'm back at this... and finally discovered the difference

This is what I found to be different on the bad line:

Block CID Setting = Yes
CID Setting = No

They were the opposite on the other line. Not sure why these would affect placing a call?