Solved: Unless PSTN call is picked up

Pradeep Hapuarachchi · ‎09-05-2014

Hi Guys

Recently I discovered if the PSTN Line is having with CLI/CID enabled then PSTN to VOIP Call is not establishing with the standard parameters. But other-way (VOIP to PSTN) works well OK. Basically it works well with the PSTN lines without CLI/CID. I guess when with CLI it will have different voltage levels in the line.? Would you let me know what are the parameters I need to set in order to SPA to work with PSTN line which enabled CLI/CID please ? This situation is under Sri Lanka Telecom PSTN lines. At the moment I have standard Parameters ON as it comes with the SPA3102.

Appreciate your comments and Help Here..

Howard Wittenberg · ‎06-18-2015

Pradeep,

The documentation you posted shows Dial Plan 8:
(S0:cc_userid>in.callcentric.com)
the dial plan should be:
(S0:cc_userid@in.callcentric.com)

where cc_userid is the callcentric number to be called

With that change your configuration should work. I tested something similiar with my SPA3102.

Your posted SPA3102 INFO tab shows an incoming caller id number but does not show an outgoing "Last Called VoIP Number" which would be consistent with a faulty dial plan. ~~The faulty dial plan though should have also caused problems with the pstn line without the caller id.~~

Edit: The reason your original dial plan worked with the pstn line without caller id is with the original dial plan the call was then sent as a regular call to the proxy that you have setup on the PSTN Line Tab which was callcentric.com. When the call is sent in this manner the call is authenticated. When a call is sent to in.callcentric.com the call is sent as an incoming sip uri call and there is no authentication of the caller.

There is a setting PSTN CID for Voip CID: No/Yes. The default setting is NO. You have this setting set to YES to send the CID coming in on your PSTN line call out with the bridged outgoing voip call. When you do this it impacts certain fields in the outgoing sip call. If one of the settings that it changed is used by the voip provider in their call authentication and the call fails authentication, the call will fail. When you had no caller id coming in there was no field to be changed and the authentication did not fail.

Edit: There is an additional setting on the Line 1 Tab that you need to make:
Enable IP Dialing: YES

If the change does not solve the problem then a sip debug trace is needed to see what is going on inside the SPA3102. A sip debug trace will show receiving the incoming caller id number, will show the outgoing call's sip invite and will show CallCentric's response, if any, to your outgoing call.

You can setup a sip debug trace by installing a syslog program on a local pc, then putting the local pc's ip address on the SPA3102 System Tab under Debug Server, then setting the Debug Level to 3 on the System Tab, then on the PSTN Line Tab setting the Sip Debug Option to Full. You can use any syslog program on your local pc. A simple windows pc syslog program is available for download here:
https://supportforums.cisco.com/document/36921/using-slogsrvexe-utility

View solution in original post

Dan Lukes · ‎06-18-2015

Syslog&debug can save so many time ...

Line with no CLI:

INVITE sip:17772327245@callcentric.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK-a16a89a0
From: PradeephSL <sip:17772695735@callcentric.com>;tag=4cc44dbb58f1a944o1
To: <sip:17772327245@callcentric.com>
Remote-Party-ID: PradeephSL <sip:17772695735@callcentric.com>;screen=yes;party=calling
Call-ID: f4e23447-33d41428@192.168.1.2
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="17772695735" realm="callcentric.com" nonce="684082550c6f8ce50da482371a591df7" uri="sip:17772327245@callcentric.com" algorithm=MD5 response="01c15ae601d9d9f5af10023f908a1a4c"	opaque=""
Contact: PradeephSL <sip:17772695735@192.168.1.2:5061>
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 441
Allow: ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS REFER
Supported: x-sipura replaces
Content-Type: application/sdp

...

This INVITE is accepted by proxy.

Line with CLI:

INVITE sip:17772327245@callcentric.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK-ea75ea0
From: PradeephSL <sip:777536172@callcentric.com>;tag=2d4fafa5d94faa2co1
To: <sip:17772327245@callcentric.com>
Remote-Party-ID: PradeephSL <sip:777536172@callcentric.com>;screen=yes;party=calling
Call-ID: 2d4fc3d9-98388@192.168.1.2
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="17772695735" realm="callcentric.com" nonce="fd5a5e637b6e56fea56886ee25aababa" uri="sip:17772327245@callcentric.com" algorithm=MD5 response="b6b1aa7104d1be764de5c74f369ef5be"	opaque=""
Contact: PradeephSL <sip:777536172@192.168.1.2:5061>
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 441
Allow: ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS REFER
Supported: x-sipura replaces
Content-Type: application/sdp

...

This INVITE is rejected by proxy with "403 Incorrect Authentication"

But Howard already hit and explained it ...

View solution in original post

Dan Lukes · ‎09-05-2014

I'm not sure what you mean saying "Call is not establishing with the standard parameters".

Unfortunately, I don't know the correct parameters for Sri Lanka Telecom lines. Most critical parameters are line impedance and input (output) gain. But all of them needs to be set correctly or calls may have either distorted sound (it may cause CID not to be recognized) or may not be established at all.

Sri Lanka Telecom should be able to tell you the parameters of line provided to you.

Despite created for different hardware, the following document may help you somewhat:

Analog Voice Port Best Match Impedance Setting Choice

Pradeep Hapuarachchi · ‎09-05-2014

Hi Dan, Thanks for the reply.... Let me explain..

'Standard parameter' means the settings comes along with the SPA3012 without changing anything specially like 'Voice tab > Regional tab > Miscellaneous >' Voice tab > PSTN Line > International Control >' etc..

With that when I make a PSTN to VOIP call it works well and other end SIP phone rings OK. This is OK if the PSTN line is NO CLI/CID enabled.

When I plug a PSTN line to the LINE port of the SPA3102 which CLI/CID enabled, then VOIP call fails. I can hear PSTN line rings but VOIP part is not establishing thru. In the SPA3102 Info page shows 'VOIP Call failed'

Pradeep Hapuarachchi · ‎09-05-2014

I mean in SPA 3102 'PSTN Line Status' there is a sub status call 'Last PSTN Disconnect Reason:' it shows as 'VOIP Call failed'

Dan Lukes · ‎09-08-2014

Unless PSTN call is picked up, no VoIP can be established. Well, you may mean that PSTN call is not picked up ...

Well. I suspect that only syslog&debug messages may disclose informations important to us. So turn them on and catch them ...

Pradeep Hapuarachchi · ‎09-09-2014

Yes Correct, NO PSTN Line is physically picked up as this PSTN line is dedicated for Call termination purpose though this is for private usage.

I am not very sound in technical, But once I setup a Syslog, but I am not sure how to debug it. Let me see whether I can recall how to setup and get the log out from here.

Whatever it is 100% confirm this problem comes for the PSTN lines with CLI/CID. Moment you connect a Non CLI/CID enabled line all works as normal...

Dan Lukes · ‎09-09-2014

I have no SPA3102. Following document may help you (as they describe setup for similar device):

Debug and syslog Messages from SPA Analog Telephone Adaptors

Pradeep Hapuarachchi · ‎06-17-2015

Hi Dan

Basically I tested all the available parameters under 'FXO Port Impedance' and NO luck.

Please see the images attached.

Do you recommend me to setup a Wire-shark. ? I m not 100% familiar how to do that.

Can you recommend some parameter combinations.

Dan Lukes · ‎06-18-2015

I'm unable to recommend you parameter combination suitable for particular telco provider. There is nothing like generic answer that's fit for all.

But I'm still unsure about the issue. OK, the PSTN is ringing. But is it picked up by SPA3102 or not ? If not, then "Ring *" options may not be set correctly. Or Line-In-Use voltage.

Do you recommend me to setup a Wire-shark.

Yes. Then configure the computer's IP as a target for syslog and debug messages in SPA3102 configuration.

Pradeep Hapuarachchi · ‎06-18-2015

Hi Dan

Let me explain with some diagrams. Your help and understanding is VERY appreciated. For the moment let me try to tell like this as SysLog/Wireshark way I m not very comfortable. This has 4 attachments/diagrams/images for better reference.

As per the ‘Manual Diagram.PDF’ my setup is connected @ Location A
My SIP provider is ‘callcentric.com’
Configurations done as per ‘SPA-3102 Config Pages.Doc’
As per this, under PSTN Tab of the SPA-3102, I am forwarding the PSTN call to far end ‘Location B’ SIP registered ID as per the Dial Plan 8.
Under ‘PSTN to VOIP Gateway Setup’ I am using the Dial Plan 8 (PSTN Caller Default DP : )
Remaining Parameters you may refer as per the Full page shown.

As per the setup like this, if any PSTN Call comes IN to the PSTN Number (Local Telco Number @ Location A) will straight ring the Location B’s SIP phone configured (as per the ‘SPA-112 Line 1 Port.JPG’ image).

This setup works totally well & Fine since ‘Location A’ PSTN Line has NO CLI enabled.

As soon as I plug a CLI enabled PSTN line @ Location A’s Line Port of the SPA3102, then the PSTN Call not transfer to the Location B far end.

Caller at Location A side whom trying to reach Location B can hear phone rings but PSTN port not picks the call. When I check the SPA-3102 infor page I can see its “Ringing” status and also later I can see “Last PSTN Caller” also get recorded (Refer ‘PSTN Status.JPG & PSTN Status 2.JPG’ images).

Many Thanks

Dan Lukes · ‎06-18-2015

since ‘Location A’ PSTN Line has NO CLI enabled

I'm unsure about your conclusion. I see no reason to assume the CLI is the only difference. If a PBX in location B is CLI capable but PBX in location A is not CLI ready, then the simplest explanation is - it's different hardware. Thus other parameters may be different as well.

Well. I'm unsure I have a further advice to you. I even don't know which type of Caller ID is used in Bangladesh (type I or type II). And with no syslog&debug I'm out of advices.

So just some blind advices ...

1. SPA3102 is rather old equipment. May be the SPA112 will work even with CLI line at location A. And SPA3102 will work at location B like a charm. E.g. swap devices between locations.

2. Ask local telco company to turn off CLI on line

Pradeep Hapuarachchi · ‎06-18-2015

See my reply in Blue aligning with your comments below.

I'm unsure about your conclusion. I see no reason to assume the CLI is the only difference. If a PBX in location B is CLI capable but PBX in location A is not CLI ready, then the simplest explanation is - it's different hardware. Thus other parameters may be different as well.

Here NO PBX at all as the setup between 2 houses @ Location A & B.

Always I am talking about the CLI (Local Telco provider’s CLI enablement) @ Location A only. There is NO relationship whether CLI is there or NOT at location B. Because at Location B, there is NO Call Termination out. Calls ends at the SPA-112 level as a SIP/VOIP call. That’s it. In my diagram I have drawn the PSTN line was for you to understand, even location B is using ADSL internet (comes via PSTN Line). Also as I said, as per the manual diagram my setup works as I expected today (with NO CLI enabled line @ Location A). Problem is, as soon as I hook up a CLI enable PSTN line to SPA-3102’s Line port then it stops flowing the PSTN call to expected destination (which is SPA 112 with SIP registered destination)

Well. I'm unsure I have a further advice to you. I even don't know which type of Caller ID is used in Bangladesh (type I or type II). And with no syslog&debug I'm out of advices.

Hmm let me try my best to provide the Syslog details with CLI enabled and without CLI enabled situation. Honestly I am not sure what Sri Lanka Telcom’s CLI types.

So just some blind advices ...

1. SPA3102 is rather old equipment. May be the SPA112 will work even with CLI line at location A. And SPA3102 will work at location B like a charm. E.g. swap devices between locations.

No there is NO possibility to swap the devices because SPA112 has NO ‘PSTN/Line’ port to hook the PSTN line. My requirement is all the external PSTN Calls comes/generates at put side of Location A to bring down to Location B. Hence SPA3102 must be only at Location A.

2. Ask local telco company to turn off CLI on line

YES I did, as I said if I use a line which doesn’t have CLI ON, this works totally fine. That is why I am saying this is because CLI enablement causing some different Voltage to the device/setup which I am not sure whereto adjust. I tried different Values at “Line-In-Voltage” of SPA-3102 but NO luck.

Dan Lukes · ‎06-18-2015

SPA112 has NO ‘PSTN/Line’

True, it is the SPA122. Sorry.

CLI enablement causing some different Voltage to the device/setup which I am not sure whereto adjust.

Unless I missed something, there is slight difference in voltage only. It can't explain the issue.

Pradeep Hapuarachchi · ‎06-18-2015

Hi Dan & Howard

Here you may find attached SysLog results.

1. SysLog SPA3102 - NO CLI PSTN Test - Works OK

2. SysLog SPA3102 - With CLI PSTN Test - NOT working as needed.

3. Status Page With CLI PSTN Line.JPG - Also for more reference

Kind Regds

PradeepH

Dan Lukes · ‎06-18-2015