SPA3102 Extension

[fiachra]

Hello. I have a SPA3102 and have set it up as per guide. The ports 5060 and 50061 are configured correctly and are showing as registered on the ATA and showing green indicator on the 3CX portal. I set up 2 separate devices on the 3CX SIP Trunk (one for PSTN port and a second for the Line 1 port). I am not sure if this is correct.
I can dial out on the 3102 PSTN line and when I receive PSTN calls they are forwarded to the 3CX system, and ring my iPhone.
How can I get the calls to ring my house phone also (Line 1 on the 3102)?
But, when I try to dial an outbound international number which is routed via the SIP line, it does not work. My iPhone rings instead with a call from 3CX.
What have I done wrong?
Should I have set up the 3CX SIP as only one device? How would I list both ports in that event?
Should there be an extension or phone added to 3CX for the SPA3102 device ?
Thank you in advance for any advice. I sense that Im nearly there....maybe Im wrong!!

 

[fiachra]

Everything seems to work now except caller ID on FXS phone. I have set answer delay to 4 in order to pick up the caller ID. Caller ID appears perfectly on the iPhone extensions.
I also have

​

PSTN CID For VoIP CID set to YES
There are some error message on the Activity Log. See below.
Any ideas?

 

[leejor]

Do you have the FXS ringing as a 3CX extension, or directly from the PSTN port ? (3102 option....PSTN Ring Thru Line 1) If other extensions are getting caller ID, then you know it is reaching 3CX and should reach the extension using the FXS port..

I find that the Answer Delay works just fine (Bellcore Type) set at 3 seconds. The attachments are not really of any help.

Post the PSTN Tab from the 3102, if you still have issues then I can see if there are settings that need to be changed.

 

[fiachra]

PSTN ring thru set to off.
FXS phone rings by dialling the extension number from iPhone app and caller ID/Name appears on phone connected as Line 1. Also, the caller ID appears on the 3CX iPhone app. It just doesn't appear on the FXS phone when a call is sent to the PSTN line. Could it be a SIP problem ?
I'll attach the PSTN tab..its actually a Cisco 232D, but i believe that they are the same.
Thanks again for your help.

 

[leejor]

Settings all look to be correct.
Does the set show caller ID when being called from another extension? Have you tried another caller ID display device / telephone ? As I said previously, if incoming calls, on other devices show the caller ID, then 3CX is receiving it from the 3102.

 

[fiachra]

Thanks, caller ID does show when called from another another extension.
Also 3CX shows that the caller ID is received from the 3102 when the call is forwarded.
Also if i dial my Voip.ms number (which is linked to 3CX)...my 3102 rings and displays the caller ID of the original caller, so 3CX is sending that OK.
So it seems that 3CX is receiving the caller ID from the PSTN section of the 3102 and has some problem sending it back on the SIP line. Perhaps it is a different format or something?
My PSTN line comes from my Telco's gigabit modem/router....maybe that is creating a problem?
Last piece of the puzzle.

 

[leejor]

You might want to check the 3CX Activity Log, and compare a call coming in on the 3102 and one from voip.ms, in particular, the caller ID.
So it is working from the 3102 FXO to all other phones on 3CX, and it is fine from all phones, and the voip.ms trunk to the FXS on the 3102, correct?

You did use a different port on the FXS and FXO , FXO = 5060, and FXS= 5062 ?

 

[fiachra]

I used FXS 5062 and FXO 5063. I’ll compare the activity log for Voip.ms and 3102.
thank you

 

[leejor]

Also, look at the Invite from the 3102 FXO and the Invites being sent back to the FXS as well as all other extensions (in the ring group). If the others are getting caller ID and the FXS port on the 3102 is not, then there should be something different.

 

[fiachra]

The 3CX call history shows the received numbers from Voip.ms which send successful caller ID with the origin number in brackets (1353404xxxxxxxxxx).
However the received numbers from the 3102 show as 353404xxxxxxxxxx (1353404xxxxxxxxxx) and do not show a caller ID. I have attached a screenshot.
Thank you for the suggestion.

The activity log (verbose) shows invites with strange looking caller ID

12/07/2020 6:37:51 PM - L:1.1[Line:10000<<1353872520590] Sending: OnSendResp Send 180/INVITE from 0.0.0.0:0 tid=-4188d0e2 [email protected]: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.205:5062;branch=z9hG4bK-4188d0e2 Contact: <sip:[email protected]:5060> To: <sip:[email protected]>;tag=7c94a451 From: "353872520590"<sip:[email protected]>;tag=f6aed19a28f623a4o2 Call-ID: [email protected] CSeq: 102 INVITE Content-Length: 0 12/07/2020 6:37:51 PM - SendMsg from <sip:[email protected]>;tag=7c94a451 to "353872520590"<sip:[email protected]>;tag=f6aed19a28f623a4o2 12/07/2020 6:37:51 PM - L:1.2[Extn:10] got Provisional Recv 180/INVITE from 192.168.1.205:5063 tid=2eb0f44aa430ba43 Call-ID=iNYKaE-7zyKFDEufi5-ZjQ..: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK-524287-1---2eb0f44aa430ba43 Contact: "Ext 10" <sip:[email protected]:5063> To: <sip:[email protected]>;tag=288eed294676a58ei0 From: "353872520590"<sip:[email protected]:5060;nf=e>;tag=812e0c5d Call-ID: iNYKaE-7zyKFDEufi5-ZjQ.. CSeq: 1 INVITE Server: Cisco/SPA232D-1.4.1(002_282) Remote-Party-ID: "Ext 10" <sip:[email protected]>;screen=yes;party=called Content-Length: 0

 

[leejor]

the received numbers from the 3102 show as 353404xxxxxxxxxx (1353404xxxxxxxxxx) and do not show a caller ID. I have attached a screenshot.

So, I'm a bit confused. Previously you said that "other" extensions were getting caller ID on external calls, coming in on the 3102 FXO. If you are not seeing caller ID in the Invite then I don't see how that is possible. If there is no caller ID, then... is the CID method correct? Is there a CID option on the PSTN line? Does a CID box, plugged into the same line get caller ID?

The Event Log is not helpful for diagnosing SIP issues. The Activity Log shows what is happening with the call as it passes through the PBX.

Invite info...
https://www.3cx.com/blog/voip-howto/sip-invite-header-fields/

 

[fiachra]

Thanks again.
I have restricted inbound calls to the ATA line 1for this test. The ATA PSTN port is 5062.
I have also changed the SPA232d Line 1 port to 5060, but problem remains.
I have made 1 call to my local PSTN number and attached the activity log associated with that call.The caller id does not work with the FXS on this call.
I have also made a call to my voip.ms number and attached the log associated with that. This call works with caller ID on my FXS phone.
I have de-identified and inserted random phone numbers and 3cx id in each log.
192.168.1.191 is 3CX
192.168.1.205 is ATA

 

[leejor]

12/07/2020 8:02:32 PM - Call from
<sip:35340455566677@50.31.115.149>;tag=as48e20169 to
<sip:[email protected]>;tag=2abd9244


12/07/2020 7:54:13 PM - Call from
"35340455667788"<sip:135340455667788@192.168.1.191>;tag=888178167cf1721ao2 to
<sip:[email protected]>;tag=fea8fd79

So the first is from VoP.MS (I assume), the second from the 3102. Both show that 3Cx has received caller ID, the 3102 is also sending caller ID name (in quotes), which is the number again. So the issue would seem to be 3CXs ability to send caller ID back to the ATA, on the 3102, or, the set plugged into the FXS port.
Voip.ms to the set..

Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---ff7f8f17e14d1111;rport
Max-Forwards: 70
Contact: <sip:35340455566677@192.168.1.191:5060>
To: <sip:[email protected]>
From: <sip:35340455566677@192.168.1.191:5060;nf=e>;tag=34908e1a
Call-ID: 5F0ZtWo-nGzx_nZSYs5xoA..
CSeq: 1 INVITE

3102 to the set...

Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---7bb72f2023691a70;rport
Max-Forwards: 70
Contact: <sip:135340455667788@192.168.1.191:5060>
To: <sip:[email protected]>
From:
"35340455667788"<sip:135340455667788@192.168.1.191:5060;nf=e>;tag=a3c49471
Call-ID: iYaskPNoRozDZ2hhmNhNCw..
CSeq: 1 INVITE


So the only difference seems to be that there is also a name display being sent on incoming calls from the 3102. If you unplug the set from the FXS and put it directly into the PSTN line (unplug the FXO of the 3102), then try a call to the PSTN number, does the display work?

 

[fiachra]

Thank you.
Set plugged directly into PSTN home line. Caller ID works perfectly, however in Ireland the Telco caller ID only sends phone number and never sends name display. Perhaps this is what is confusing it.
Is there a way to filter out the name display being sent from 3102? This may solve the problem if that is the only difference?
The 232D has SIP caller ID header options of PAID-RPID-FROM, PAID-FROM, RPID-PAID-FROM, RPID-FROM,FROM. i'm not sure if that will affect things....iy is currently set for PAID-RPID-FROM.
I have tried different options, but they don't seem to change things.

 

[leejor]

It may be an issue wit the caller ID device you are using. If only numbers are sent in Ireland then there is a possibility that the local telco is using Single Data Message Format (SDMF), and the set is only capable of making use of that. The standard in North America (Bellcore), or areas that also send a name is Multiple Data Message Format (MDMF). An older SDMF format display will generally not read an MDMF message. I know no way of changing that on the ATA, (or removing the name) as SDMF is considered obsolete. I have no idea why the set will show number from the Voip.ms trunk, it makes me think that it is capable of MDMF.. You may want to see if you can borrow a call display device, or set to see if using a different device, works.

I know that some places like Australia and New Zealand also use Bellcore , but, as I understand it, do things a bit differently.
Does the Linksys have the latest firmware?

 

[fiachra]

The ATA has the latest firmware. I even rolled it back a few editions to see if it would help.
Its bizarre because the analog phone caller ID works for all calls received from wall PSTN and from inbound Voip.ms via the ATA. It just won't work with calls which originated from the PSTN section (FXO) of the ATA and then returned to the FXS section. I have tried various changes to the 'caller ID method' section and to the FXS input/output gain, but they don't make any difference.
Just wondering, under SIP settings, should 'handle via' and 'Insert via' be yes/no?
I may just have to accept that if I use 3CX that caller ID won't work
Thank you ever so much for all of your help in getting me this far.

 

[leejor]

Because you are actually using 232 and not a 3102, I don't know if there are any additional settings. Caller ID has always worked "out of the box", in the set plugged into the FXS port. The set does receive caller Id from other sources which mean that the FSK circuit is working. Did you ever try using a different set/display? If you have another ATA, you might plug the PSTN port into that then call the extension associated with, to see if caller ID works. (replace the PSTN line with an ATA extension from 3CX).
Short of doing a factory reset, and starting all over again, I can't think of anything else to try. THe log shows that 3CX is sending it back to the 3102.

 

[fiachra]

Just wanted to give an update.
I did a full reset on the 232D, and completed the info to pair with the 3CX system. The caller ID was still not working.
I went to the Regional settings in the 232D menu and tried each caller ID possibility one at a time. I discovered that by using the setting 'ETSI DTMF after ring' and V23 that the caller ID worked. I have not idea why this is the case, but I'm very happy that it did.
The Irish setting is ETSI FSK and V23.
Thank you again for all of your help.

 

[leejor]

Glad you figured it out.
I had assumed that the Caller ID was set to the correct standard used in Ireland, as you had said that incoming calls, through the FXO were showing on other sets , and other (VoIP) incoming calls were showing on the set using the FXS. I know that in the UK, some providers use the Bellcore standard, and not what is used by British Telecom.

 

[fiachra]

Yes, it was strange that when the PSTN calls were routed through the ATA to the FXS port caller ID worked on the Irish standard of ETSI FSK, but when the PSTN calls were routed via the 3CX and returned to the FXS port, they only work with ETSI DTMF!
Would you suggest that I base my dial plan using the 3CX inbound rules, or by using the Cisco....or a combination of both?
if ax extension number is '10' should you just arrange for a number of '10' to route through to the VOIP section, or is there a prefix required?

 

[leejor]

The dialplan for the FXS port should be constructed to allow the the immediate sending of any extension or known length/format number, and I would route all dialled calls though 3CX, then out to whichever route 3CX is set to use, even if that is back though the FXO port. This way all extensions consistently follow the same rules. If you look at the example I attached earlier, the only reason that i had numbers starting with 808 (or 911 and toll free) routing out on the FXO is that I was remote to the PBX, and was using a local PSTN number there. It made more sense to keep local calls...local. I also had any incoming local PSTN calls ring the FXS port, but, if unanswered after a set time, would deflect to my voicemail on 3CX.