vicidial.org

Hello Everyone,

Need your inputs.

How can we remove the UID or the one highlighted below on FROM header when doing predictive and manual dialling. We just want to make sure that we are sending only the correct name/number to our trunks/called party and not the UID. I saw an article that the UID is being used on the entire vicidial system.

[May 12 20:59:44] INVITE sip:12027621069@52.26.213.75 SIP/2.0
[May 12 20:59:44] Via: SIP/2.0/UDP 197.253.13.77:5060;branch=z9hG4bK7fc9ba71;rport
[May 12 20:59:44] Max-Forwards: 70
[May 12 20:59:44] From: "M5122059440051743154" <sip:+12024101721@197.253.13.77>;tag=as0251215e
[May 12 20:59:44] To: <sip:12027621069@52.26.213.75>
[May 12 20:59:44] Contact: <sip:+12024101721@197.253.13.77:5060>
[May 12 20:59:44] Call-ID: 0807b411737b248902832fbf745915c5@197.253.13.77:5060
[May 12 20:59:44] CSeq: 102 INVITE
[May 12 20:59:44] User-Agent: Asterisk PBX 13.29.2-vici
[May 12 20:59:44] Date: Wed, 13 May 2020 00:59:44 GMT
[May 12 20:59:44] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[May 12 20:59:44] Supported: replaces, timer
[May 12 20:59:44] Remote-Party-ID: "M5122059440051743154" <sip:+12024101721@197.253.13.77>;party=calling;privacy=off;screen=no
[May 12 20:59:44] Content-Type: application/sdp
[May 12 20:59:44] Content-Length: 257
[May 12 20:59:44]
[May 12 20:59:44] v=0
[May 12 20:59:44] o=root 791153939 791153939 IN IP4 197.253.13.77
[May 12 20:59:44] s=Asterisk PBX 13.29.2-vici
[May 12 20:59:44] c=IN IP4 197.253.13.77
[May 12 20:59:44] t=0 0
[May 12 20:59:44] m=audio 14816 RTP/AVP 0 101
[May 12 20:59:44] a=rtpmap:0 PCMU/8000
[May 12 20:59:44] a=rtpmap:101 telephone-event/8000
[May 12 20:59:44] a=fmtp:101 0-16
[May 12 20:59:44] a=ptime:20
[May 12 20:59:44] a=maxptime:150
[May 12 20:59:44] a=sendrecv


Regards,

Jessie

set the did in the campaign....looks like its set to: 2024101721

We are using AC-CID. We want to completely eliminate the UID as FROM header. Any idea how?

Regards,

Jessie

AKA: Caller ID Name. This is covered in a multitude of posts on this site.

What is your goal? IE: WHY do you want to change the Caller ID Name? Is it annoying your grandmother or are calls failing or are there complaints or extra charges or ...?

The loopback dialplan method allows generating a second call which has a fully editable callerid name value since that second call is not a vicidial-linked call. You can also pass the call through a different SIP server and use that second server to modify the CID Name field without penalty in Vicidial.

Hi Williamconley,

Yes, you are correct. I am referring to the Caller ID Name.

Well, basically our ultimate goal is just to present the caller id number and totally remove the caller ID name. We are getting some complaints because of the UID as caller id name. If we will use the loopback, what will be the cons using it? eg. additional overhead on the server's load.

Also, where on the dialplan it is located?

Thanks in advance.

Regards,

Jessie

This creates a second call. I'd say the overhead there is obvious. It' in the manual. Have you downloaded the manual?

Hello,

Apparently the only way to reliably do this without creating more load on the server by having to route back a call internally to make the call (which creates a tracking nightmare in the call log / outbound log) is to use a OpenSIPS Server to send the calls and remove the header there.

OpenSIPs is not required. Any asterisk server can handle passthrough calls. OpenSIPs is a bit ... involved for most call centers.

Your Carrier can also alter your callerID name if they are set up correctly, but then you lose the ability to manage it yourself.

Logs shouldn't present an issue. No clients to date have complained that any of the logs had extraneous entries in them. But that depends on how you generate that second call. It does not need the agi() line, for instance. But it does definitely add load to the server.