TIPS How to confirm CNAM on contact header

[rsarceno]

I'm having Outbound Called ID issue (onlye number / no cname)

My trunk provider (Bandwidth) look into one of my INVITEs and they saw that the From header had "name" on it,
but the Contact header only had the originating number.

How can I confirm this? I can see my "name" on the log file

My vendor recommend I update the Contact header, and add a p-asserted-identity header to the INVITE with your caller ID information.

 

[Halea]

If I understand you correctly, your PBX is sending out your Caller ID phone number correctly but not the correct Caller ID name.
If that's the case, the first thing to check is the CDR reports window/page which shows the actual outbound caller id, both phone number and name, as it was sent upstream to your trunk provider for every particular call.
If that info is inaccurate, figure out where it's being set as it could be at the trunk level, outbound route level or extension level, and fix it there.
The typical format is "CallerIDname"<CallerIDphonenumber>, including the "" and <> characters.

 

[rsarceno]

The CDR contains the CNAM and number


Ext 102 is pjsip. Outbound CID is "VoIPAccess" <9162392320>
Ext 103 is sip. Outbound CID is "VoIPAccess" <9162392320>

Outbound Routes > Route CID is blank and Override Extension is No

Trunk > Hide CallerID is No

Trunk > Outbound CallerId is blank

Trunk > CID Option is Allow Any CID

Trunk > Sip Settngs > peer details > sendrpid=pai

Thanks

 

[kenn10]

So the PJSIP extension works correctly but the chan_sip extension does not or do neither work? Also what version of Asterisk are you using and what version of FreePBX?

On the extension advanced settings, try setting like this:


Also, on the extension caller ID settings make sure there is no space between "NAME" and <NUMBER> in the outbound caller ID. It should look like "VoIPAccess"<9162392320> and not "VoIPAccess" <9162392320>

Note that most caller-id name to the called party is provided by the end Central Office. The receiving Central Office does a database lookup from the LIDB database when processing an incoming call. If the caller-id name of your numbers has not been set in the LIDB, the called party may not receive any caller-id name from your users whether you send it in your outbound header or not.

If you were to call me from your number above, the caller-id name is showing as "VOIPACCESS."

 

[tbrummell]

Note that most caller-id name to the called party is provided by the end Central Office. The receiving Central Office does a database lookup from the LIDB database when processing an incoming call. If the caller-id name of your numbers has not been set in the LIDB, the called party may not receive any caller-id name from your users whether you send it in your outbound header or not.

Unless they are in Canada, then what is sent, is carried through.

 

[kenn10]

Unless they are in Canada

Noted. However, they are in California. Canuk-ary does not apply.

 

[Samot]

My trunk provider (Bandwidth) look into one of my INVITEs and they saw that the From header had "name" on it,
but the Contact header only had the originating number.

Bandwidth uses From Header, PAI and RPID to pull the CallerID information. The only time I see Bandwidth use the contact header for things is the Dynamic Location Routing. I send all my callerID to them over PAI and I've never had a single issue with CallerID.

 

[rsarceno]

It does not work with both PJSIP and CHANSIP

I've try different Outbound CID format and it didn't work.
"VOIPACCESS"<9162392320>
"voipaccess"<9162392320>
"VoIPAccess"<9162392320>
"VoIPAccess" <9162392320>
"VoIPAccess"<19162392320>
"VoIPAccess"<+19162392320>

I've also try setting Outbound Routes > Route CID and Trunk > Outbound CallerId to "VOIPACCESS"<9162392320> and it didn't work

Both PJSIP (ext 102) and SIP (ext 103) Extension > Advance > Send RIPD are set for "Send P-Asserted-Identify Header"

LIDB is set and CNAM works when I query using third party to verify like calleridtest dot com but. I did confirm with Bandwidth that it uses Header, PAI and RPID to pull the CallerID informatio. As I mention on my on my initial post, Bandwidth see that the From header had "VoIPAccess" on it,
but the Contact header only had the originating number.

I have several PBX servers and non of the outbound caller ID works
Asterisk Version 16.13.0 Incredible PBX 2020 15.0.17.55
Asterisk Version 13.32.0 Freepbx 15.0.17.68

I appreciate all the tips but I'm still missing something

 

[atsak]

Unless they are in Canada, then what is sent, is carried through.

Ha - sometimes. ILEC's tend to do LIDB lookups a lot of the time (Bell / Telus) IME, the CLEC's (including the big ones) and Rogers Wireless phones will take what they get. Sometimes. CID is very weird up here.

 

[kenn10]

I appreciate all the tips but I'm still missing something

You appear to have everything set correctly. I suspect the issue lies with Bandwidth.com. They are not known for being extremely helpful because they expect the customer to be an expert. I believe @Samot on this forum has experience with Bandwidth.com and may be able to advise you further.

Have you tried calling someone on a land line with caller-id who is off your system to see if they are getting the caller-id? Cell phones are not a good option for receiving calling name and that is what you were calling in your examples above.

 

[rsarceno]

Bandwidth uses From Header, PAI and RPID to pull the CallerID information. The only time I see Bandwidth use the contact header for things is the Dynamic Location Routing. I send all my callerID to them over PAI and I've never had a single issue with CallerID.

Samot, can you think of anything I missed? Any ideas?

Can we compare bandwidth trunk sip setting?
type=peer
qualify=yes
progressinbound=yes
outboundproxy=x.x.x.x
nat=no
host=x.x.x.x
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=from-trunk
canreinvite=no
sendrpid=pai

I'm not sure if Bandwidth offers trunk pjsip. But if your using, do you mind sharing your setting

Thanks

 

[Samot]

I'm not sure if Bandwidth offers trunk pjsip. But if your using, do you mind sharing your setting

PJSIP is SIP and SIP is SIP. They support it because it is SIP. As for my Asterisk configs, I can't share that with you as I don't connect individual PBXes directly to Bandwidth. I have softswitches between Bandwidth and my network.

Here is how a call looks being sent from one of my softswitches to Bandwidth, which sourced though an Asterisk system running Chan_PJSIP. Nothing in the contact name field and CallerID is delivered fine.

INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5082;rport;branch=z9hG4bKPj1d1849ce-844a-4bf6-b7b6-86400340b05c From: "MY CNAM" <sip:[email protected]>;tag=d285fdb6-5e19-4365-80dc-0b03e9d48c33 To: <sip:[email protected]> Contact: <sip:[email protected]:5082> Call-ID: 7e6e808c-66e2-4cbc-a8cf-7fb50b1d042d CSeq: 15160 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "MY CNAM" <sip:[email protected]> Max-Forwards: 70 User-Agent: CVN-Cloud-3.0 (18.11.3) Content-Type: application/sdp Content-Length: 241

 

[rsarceno]

Samot, thanks for the info and sample report. Its nice to have a good report to compare with.

My pjsip history shows
<— History Entry 0 Received from 10.0.0.18:5060 at 1652306908 —>
INVITE sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.18:5060;received=10.0.0.18;branch=z9hG4bK4071026270
From: “PBX1 101” sip:[email protected];tag=2104256583
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.82.0.30
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 286
Content-Type: application/sdp
Content-Length: 286

The caller ID is set on the Extension > Outbound CID “myname”<9165551212>
I don't know why it shows “PBX1 101” sip:101@10....
PBX1 is the hostname and 101 is the extension number
The ext display name is set to myname

I also don't see p-asserted-identity on my history so I'm assuming its not set.

Any ideas?