Possible problems with SPA3102 on 7.1

[boffin]

I note that the SPA3102 gateway has been re-written for v7.1 and think this might be the reason for my problems.

I have just configured a new 3CX server on v7.1 in an identical way as an earlier v7.0, with a Linksys SPA3102 providing a PSTN Gateway.

Using an identical configuration as on the SPA3102 as I had historically used, I was unable to get incoming calls to even register in the 3CX system. To confirm that I wasn't going silly, I even swapped a SPA3102 from the 7.0 system to the 7.1.

What I eventually found in the SPA3102 configuration, I had to set the PSTN Answer Delay: to either 0 or 1 to allow incoming calls to register. Any higher and it wouldn't work. The downside to this is that CID no longer works, as it need a setting of at least 2 to give the system long enough to extract this data.

Also another unearthed couple smaller faults. When you create a new SPA3102 gateway, the 3CX defaults to port 5062. This is not open, even when the configuration is saved, but port 5061 is. I had to manually reset the port to 5061 on the 3CX to make it work. (setting both 3CX and SPA3102 to 5062 isn't any good. Also I noted that the generated config file, still persists in using port 5062 - even though its been set to 5061 on the 3CX

Can anyone else confirm my findings? Can an update be provided soon Please!

So in summary, with a little manual tweaking, I now have incoming calls from the PSTN, but no CID being delivered.

John

 

[William400]

Hi

Issue noted on the PSTN answer delay. We are setting this to '0' by default in the current template. What value were you finding on yours after you provision? What country and telco are you using? This may need to vary per tone set to get it right.

Also, the Linksys port is set to 5062 on the line port since the phone port uses 5060. The 3CX install uses 5061 for Secure SIP, hence the reason for it being open. Are you encountering issues due to a firewall running on the 3CX machine?

Await you reply to review templating possibilities.

 

[boffin]

Thanks for the comment William.

I am manually configuring the SPA3102. However I checked the template generated, and this is 0

I am in New Zealand, and tried both Custom (but nothing configurable) and our close cousins Australia (which it is currently set to). I have found that I have needed historically to set the delay to at least 2, to allow for the CID information to be transferred to the phone. As stated, if set to 2 or higher, the incoming call isn't delivered to the phone system at all.

I currently have the Extension on port 5060 and the PSTN on 5061 and its all working. I have tried 5060 and 5062, but PSTN would not register. Device in on local LAN and Windows firewall is disabled. Even tried to telnet to the 3CX server, using port 5062 and was told that the port was not open but without any changes on the 3CX, I tried port 5061 and it was open.

 

[boffin]

After more experimenting, I have found that if I set the delay to 3, I get the following error in the log file

22:32:20.562 [CM102001]: Authentication failed for SipReq: INVITE <[email protected] tid=-9f7077e5 cseq=INVITE [email protected]:5061 / 102 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings

22:32:20.562 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:%[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.3.5:5061;branch=z9hG4bK-9f7077e5
Max-Forwards: 70
Contact: "Ferry PSTN"<sip:[email protected]:5061>
To: <sip:%[email protected]>
From: "Ferry PSTN"<sip:[email protected]>;tag=4ab42041a72eb25o1
Call-ID: [email protected]
CSeq: 102 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Proxy-Authorization: Digest username="10000",realm="3CXPhoneSystem",nonce="414d535bffd2a22465:af7032d330065eb3dc1f033c90da6ad3",uri="sip:<[email protected]",algorithm=MD5,response="223e223e1831c94bd59b71b7d42f9501"
Supported: x-sipura, replaces
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Remote-Party-ID: Ferry PSTN <sip:[email protected]>;screen=yes;party=calling

22:32:20.562 [CM302002]: Authentication failed due to unidentified source of: SipReq: INVITE <[email protected] tid=-9f7077e5 cseq=INVITE [email protected]:5061 / 102 from(wire)

22:32:20.453 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:%[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.3.5:5061;branch=z9hG4bK-605c5e85
Max-Forwards: 70
Contact: "Ferry PSTN"<sip:[email protected]:5061>
To: <sip:%[email protected]>
From: "Ferry PSTN"<sip:[email protected]>;tag=4ab42041a72eb25o1
Call-ID: [email protected]
CSeq: 101 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Remote-Party-ID: Ferry PSTN <sip:[email protected]>;screen=yes;party=calling

22:32:20.453 [CM302001]: Authorization system can not identify source of: SipReq: INVITE <[email protected] tid=-605c5e85 cseq=INVITE [email protected]:5061 / 101 from(wire)

Ext 51 doesn't ring

However if I set the delay to 1 then the following appears in the logs

22:36:35.109 [CM505001]: Ext.51: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA942-6.1.3(a)] Transport: [sip:192.168.3.4:5060]
22:36:35.093 [CM503002]: Call(87): Alerting sip:[email protected]:5060
22:36:34.921 [CM503004]: Call(87): Calling: Ext:Ext.51@[Dev:sip:[email protected]:5060]
22:36:34.921 [CM503010]: Making route(s) to <sip:[email protected]:5060>
22:36:34.921 [CM505002]: Gateway:[Ferry PSTN] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.7(GW)] Transport: [sip:192.168.3.4:5060]
22:36:34.906 [CM503001]: Call(87): Incoming call from 10000@(Ln.10000@Ferry PSTN) to <sip:[email protected]:5060>

Strange, but true

and call is delivered to Ext 51

 

[Bob Denny]

I have this exact same problem. Logs are similar. Any ideas? I am in the US. The problem appears to be related to whether the CID is picked up by the 3102. If it is, 3CX refuses the call:

15:03:22.583  [CM102001]: Authentication failed for SipReq:  INVITE [email protected] tid=-c88aca68 cseq=INVITE [email protected]:5063 / 102 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings

15:03:22.583  [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
  INVITE sip:[email protected] SIP/2.0
  Via: SIP/2.0/UDP 192.168.2.101:5063;branch=z9hG4bK-c88aca68
  Max-Forwards: 70
  Contact: "10002"<sip:[email protected]:5063>
  To: <sip:[email protected]>
  From: "STEPHENS S"<sip:[email protected]>;tag=fe6f71ccbab96a6co1
  Call-ID: [email protected]
  CSeq: 102 INVITE
  Expires: 240
  Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  Proxy-Authorization: Digest username="10002",realm="3CXPhoneSystem",nonce="414d535c0040beaa18:780b5f0259bbfdf6b3445542de93a69e",uri="sip:[email protected]",algorithm=MD5,response="5d1ad8e39bc9b92ab5988bb1eb4a8808"
  Supported: x-sipura, replaces
  User-Agent: Linksys/SPA3102-5.1.7(GW)
  Content-Length: 0
  Remote-Party-ID: STEPHENS S      <sip:[email protected]>;screen=yes;party=calling

  
15:03:22.583  [CM302002]: Authentication failed due to unidentified source of: SipReq:  INVITE [email protected] tid=-c88aca68 cseq=INVITE [email protected]:5063 / 102 from(wire)
15:03:22.474  [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
  INVITE sip:[email protected] SIP/2.0
  Via: SIP/2.0/UDP 192.168.2.101:5063;branch=z9hG4bK-d51f696c
  Max-Forwards: 70
  Contact: "10002"<sip:[email protected]:5063>
  To: <sip:[email protected]>
  From: "STEPHENS S"<sip:[email protected]>;tag=fe6f71ccbab96a6co1
  Call-ID: [email protected]
  CSeq: 101 INVITE
  Expires: 240
  Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  Supported: x-sipura, replaces
  User-Agent: Linksys/SPA3102-5.1.7(GW)
  Content-Length: 0
  Remote-Party-ID: STEPHENS S      <sip:[email protected]>;screen=yes;party=calling
  

15:03:22.474  [CM302001]: Authorization system can not identify source of: SipReq:  INVITE [email protected] tid=-d51f696c cseq=INVITE [email protected]:5063 / 101 from(wire)

With the 3102 set to answer after 0 rings, and no other changes!!:

14:52:39.177  [CM503008]: Call(64): Call is terminated
14:52:39.162  [CM503008]: Call(64): Call is terminated
14:52:26.958  [CM503004]: Call(64): Calling: Ext:Ext.103@[Dev:sip:[email protected]:5060]
14:52:26.958  [CM503010]: Making route(s) to <sip:[email protected]:5060>
14:52:16.958  [CM503007]: Call(64): Device joined: sip:[email protected]:40600;rinstance=9d5f0749e218ea9c
14:52:16.943  [CM503007]: Call(64): Device joined: sip:[email protected]:5063
14:52:16.927  [CM505001]: Ext.800: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Voice Mail Menu;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] Transport: [sip:127.0.0.1:5060]
14:52:16.896  [CM503002]: Call(64): Alerting sip:[email protected]:40600;rinstance=9d5f0749e218ea9c
14:52:16.412  [CM503004]: Call(64): Calling: Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=9d5f0749e218ea9c]
14:52:16.412  [CM503010]: Making route(s) to <sip:[email protected]:5060>
14:52:16.396  [CM505002]: Gateway:[Bob's land line] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.7(GW)] Transport: [sip:192.168.2.20:5060]
14:52:16.380  [CM503001]: Call(64): Incoming call from 10002@(Ln.10002@Bob's land line) to <sip:[email protected]:5060>
14:52:16.380  [CM503012]: Inbound office hours rule (unnamed) for 10002 forwards to DN:800

I tried adding a DID rule but that didn't do it. What now?

 

[Bob Denny]

I solved this one! I needed to add a Source Identification rule to allow it to ID the port by the Contact : Host Part with my SPA3102's IP/Port in it (192.168.2.101:5063). The error messages are pretty obscure, but I tried this on a whim and it worked.

Would there be a better/preferable way to ID it?

 

[boffin]

Thanks for the feedback, however I don't fully understand the solution

Bob ( or anyone else), can you explain in simple terms the solution please

Thanks
John

 

[Bob Denny]

Under PSTN Devices, select the gateway for your SPA3102. Then click the Inbound Parameters tab. In the Call Source Identification area, SIP Field, select Contact : Host Part. For Variable, enter the IPort of your SPA3102, for example 192.168.2.101:5063.