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How do I get the "Called Number" to display on Handsets?

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Advoc8tr

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We've been using 3CX PABX for quite a while and it has been fabulous. I have 1 x VOIP trunk and 1 x Lynksys SPA adapter.

Caller ID has always worked straight out of the box. We used to get the Ring Group name on top line of display and the callers phone number on the second line. (using Polycom 301 phones)

The only changes I have made is that I recently changed over my firewall from pfsense to astaro and implemented the SIP proxy - but I can't see how that has anything to do with the problem we now face which is ...

all incoming calls (from both the VOIP provider and the PSTN device) now show up as the Ring Group on the top line of the display and "sip:0417XXX192@IPAddress" on the second line where it used to show "0417XXX192" If I return a call from the missed call log on the phone it seems to work fine ??? however as the display is truncated I would prefer it show just the phone number like it used to?

Any ideas on what has changed? Could it be a setting that has been updated in the provisioning templates by 3cx and if so which one?

I have tried changing the inbound SIP field mapping but it didn't make any difference?
 
Re: Inbound Caller ID suddenly changed

do you have an opportunity to try the old router to see if it changes back ?
 
Yes, I reinstated the original firewall and the behavior persisted?

I have determined that caller ID is working - by unchecking the relevant global setting and stopping the Ring Group from being prepended to the Caller ID I can get the Caller ID to show (It was being truncated on the first line of the screen) It appears that previously it was 'wrapping' onto the second line and I could see both?

So now I think that whatever sip field is being displayed on the second line of the phone display was previously not getting passed which allowed the first line to 'wrap'?

Does anyone else use polycom phones and can tell me what they usually get on the 2 lines of display when a call comes in? I am getting the Caller ID on the first and the full sip address on the second? It doesn't look like it is possible to isolate the ring group to a variable and pass it to the phone to show on the second line of the display.

Essentially we are trying to display the incoming line / trunk and the caller ID (I have each incoming line going to its own ring group)
 
I have determined that there is in fact nothing wrong with the caller ID function per se ..... After removing the global option to append/prepend ring group the caller ID shows up fine - further after adding the numbers to the company phone book the callers name shows up INSTEAD of the number :D

What I really would like help with is trying to determine what field / variable the polycom phones are showing on the second line of the display? It seems that no matter what I change in the inbound parameters for the voip provider the phone shows sip:0417XXX192@ipaddress:5060 on the second line .

We need to be able to determine what number was called when answering the phone (different numbers for different business names)

I think it is coming from the sip contact field but cannot work out how to send the CalledNum to the phone instead of what the VOIP provider sends in the contact field? If you try to map the contact: host part field there are only 2 options instead of the usual list that includes all the variables and the CalledNum variable is not one of them?

Any Ideas?
 
Rather than looking at phone settings, it may be possible to look the other way, and adjust the SIP info that is being sent to the phone and used in that particular display field ...
Have you done a Wireshark trap on the incoming call???

http://www.3cx.com/blog/docs/use-wireshark-to-capture-network-traffic/

Wireshark will show you 'to' and 'from' SIP header info that should show you exactly what parameter is being displayed at the phone.

You may then be able to edit the 'Inbound Parameters' in the 3CX VoIP Provider edit page, so that the appropriate 'SIP Field' is showing the 'Variable' that you need to see.

Note however, that the potential to upset [& kill] the connection in doing this kind of edit is extremely high!!! So please do take a back-up before you start changing things, and take note of existing SIP Field -> Variable settings.
 
What you suggest about adjusting the SIP mapping is what I am trying to do - thanks for the tip regarding wireshark.

As you can see from the capture below the sip from field is what I need to modify. It appears the handset is showing the "display" part of the "from" field on the top line and the rest of the field on the bottom line.

<code>Via: SIP/2.0/UDP 125.213.160.81:5060;branch=z9hG4bK17dd5a051ll550c4fINV5eb1e55b13cb
Max-Forwards: 35
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "0732947229"<sip:[email protected]>;tag=5eb1e55b-co5067-INS001
Call-ID: 6329-500-05201323620-BCPD_IMG01-11-125.213.174.82
CSeq: 506701 INVITE
Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO
Content-Type: application/sdp
User-Agent: ENSR3.0.6
Content-Length: 384
Diversion: <sip:[email protected]>;reason=unavailable



sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.44.5:5060;branch=z9hG4bK-d8754z-9f6e1114dd60c346-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: "Home Phone"<sip:[email protected]:5060;nf=e>;tag=3b39124d
Call-ID: NjdmZGMyMTRlM2M1OWU2MGY2MTUzYjVkMWJlYTcyNmU.
CSeq: 1 INVITE
Alert-Info: <http://www.notused.invalidtld>;info=external
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 11.0.28392.820 (27565)
Content-Length: 357</code>

I need it to be From: "Home Phone"<09475814>;tag=3b39124d (The second part of the field taken from the "to" address)

any suggestions on how to achieve this?
 
OK all the subtle issues appear to have been caused by the fact I was using multiple IP's on the LAN interface (so I could have an isolated IP for the VOIP services and another for FTP services running on same machine)

I picked this up from the wireshark capture ... an IP associated with the FTP was showing up in the sip fields?

By removing the additional IP addresses and using a single LAN IP on the server all of a sudden the display on the phone went back to how I remembered it and further if I make changes to the inbound parameters sip mapping I see them show up on the phone display - which wasn't happening before ???

Moral of the story is stick to a single LAN IP assigned to the network interface of the server running 3CX

Thanks for the help. :D
 
Thanks for the closure - and yes - have struggled with that one early on so can agree whole heatedly with that ...

3CX and multiple NIC's just do not seem to get along at all well!!!

best regards
 
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