CLI Not Passing When Using "Fromuser" parameter for supplier

[manoj.9494]

Hi All,

My supplier need Fromuser parameter in peer configuration, but due to that CLI is not passing, its sending account name in "contact" header in invite message and in from header its sending RemotePartyID, i need to send caller id in "From" header, kindly suggest

INVITE sip:12127773456@xx.xx.xx.xx;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.172:5060;branch=z9hG4bK15a66dd0;rport
From: "M2232320270000000002" <sip:testom@192.168.1.172>;tag=as29533504
To: <sip:12127773456@xx.xx.xx.xx;cpd=on>
Contact: <sip:testom@192.168.1.172>
Call-ID: 5b97bf7d4f1a942423275a133493306c@192.168.1.172
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M2232320270000000002" <sip:15522557894@192.168.1.172>;privacy=off;screen=no
Date: Sun, 24 Feb 2019 04:20:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 399

Kindly suggest me on this, i want to send CLI in from header in this format From: "15068571245" <sip:testom@192.168.1.172>;tag=as382e9e68 but its sending in this format From: "M2250037370000000013" <sip:testom@192.168.1.172>;tag=as382e9e68
i am sending 15068571245 as a cli from campaign

[ambiorixg12]

add this to your carrier configuration sendrpid = no and also make sure the fromuser parameter is set

[manoj.9494]

I tried it, its still not working,
i tried changing trustrpid & sendrpid both but still getting Remote-Party-ID in from parameter, it look like my peer configuration is ok some adjustments need in goautoddial setting, I am using Goautdial 3.0

[ambiorixg12]

show your carrier configuration,

[manoj.9494]

[testom11]
type=peer
allow=all
sendrpid=no
secret=secret
username=testom
fromuser=testom
host=xx.xx.xx.xx
callerid="15522557894" <15522557894>
canreinvite=no
insecure=very
qualify=yes
nat=yes

[ambiorixg12]

add the callerid on fromuser,

[manoj.9494]

Still its not working, below are the traces for different scenario


When trustrpid=no & sendrpid=yes
INVITE sip:12127773456@xx.xx.xx.xx;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.172:5060;branch=z9hG4bK47c0c8c2;rport
From: "M2270009130000000028" <sip:testom@192.168.1.172>;tag=as60ca7853
To: <sip:12127773456@xx.xx.xx.xx;cpd=on>
Contact: <sip:testom@192.168.1.172>
Call-ID: 0db66c5d0b85dcc2003fc5d50e0d4861@192.168.1.172
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M2270009130000000028" <sip:15522557894@192.168.1.172>;privacy=off;screen=no
Date: Wed, 27 Feb 2019 05:09:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 399

When trustrpid=yes & sendrpid=yes
INVITE sip:12127773456@xx.xx.xx.xx;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.172:5060;branch=z9hG4bK2cd2cd28;rport
From: "M2270019450000000032" <sip:testom@192.168.1.172>;tag=as4cf1ce3f
To: <sip:12127773456@xx.xx.xx.xx;cpd=on>
Contact: <sip:testom@192.168.1.172>
Call-ID: 4da167665f9c71067ba8ed0c433bc3e5@192.168.1.172
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M2270019450000000032" <sip:15522557894@192.168.1.172>;privacy=off;screen=no
Date: Wed, 27 Feb 2019 05:19:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 399


When trustrpid=no & sendrpid=no
INVITE sip:12127773456@xx.xx.xx.xx;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.172:5060;branch=z9hG4bK0a5f9b4d;rport
From: "M2270014280000000030" <sip:testom@192.168.1.172>;tag=as4dd88586
To: <sip:12127773456@xx.xx.xx.xx;cpd=on>
Contact: <sip:testom@192.168.1.172>
Call-ID: 6b4cd05a40fa33334afb399760fa1a32@192.168.1.172
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 27 Feb 2019 05:14:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 399


When Fromuser= number i want to display, its changes the contact header value which failing calls as my provider need account name in contact header
INVITE sip:12127773456@xx.xx.xx.xx;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.172:5060;branch=z9hG4bK36e2b0aa;rport
From: "M2270023420000000033" <sip:15522557894@192.168.1.172>;tag=as4a72e9e9
To: <sip:12127773456@xx.xx.xx.xx;cpd=on>
Contact: <sip:15522557894@192.168.1.172>
Call-ID: 364f03ab0f2895e77baf6ede4b5a1033@192.168.1.172
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M2270023420000000033" <sip:15522557894@192.168.1.172>;privacy=off;screen=no
Date: Wed, 27 Feb 2019 05:23:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 399

[williamconley]

Do you have an example from your carrier of what they expect?

[manoj.9494]

Here is a example what my carrier expecting from me

INVITE sip:12127773456@xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.113:7026;branch=z9hG4bK-d87543-c678650126348173-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:testom@192.168.1.113:7026>
To: <sip:12127773456@xx.xx.xx.xx>
From: "15522557894"<sip:testom@xx.xx.xx.xx>;tag=e7424963
Call-ID: 522f19375107695e@REVTS1RPUC04OEhCTUJT
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: eventlist
User-Agent: eyeBeam AudioOnly release 3014w stamp 26704
Content-Length: 252

[williamconley]

i don't see a "fromuser" parameter in their example.

If you refer to:

Code: Select all

From: "15522557894"<sip:testom@xx.xx.xx.xx>;tag=e7424963

where the caller id name/number is displayed, that's just "from" and you can set the callerid Number in the Campaign, but the Name will always be a random string (part of Vicidial). To set the caller id name, you must use the TRUNKloop / loopback-no-log method to generate a second call on which you can set the caller id name.

http://eflo.net/VICIDIALforum/viewtopic.php?f=4&t=38023

[ambiorixg12]

Here is a example what my carrier expecting from me

INVITE sip:12127773456@xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.113:7026;branch=z9hG4bK-d87543-c678650126348173-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:testom@192.168.1.113:7026>
To: <sip:12127773456@xx.xx.xx.xx>
From: "15522557894"<sip:testom@xx.xx.xx.xx>;tag=e7424963
Call-ID: 522f19375107695e@REVTS1RPUC04OEhCTUJT
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: eventlist
User-Agent: eyeBeam AudioOnly release 3014w stamp 26704
Content-Length: 252

They just require you to add the 15522557894 as the display name on the from header, just simply set the following

fromuser=testom

and on the dial plan set this

same=>n,Set(CALLERID(name)=15522557894)


your from header will be

[Feb 27 17:14:59] From: "15522557894" <sip:testom@149.28.42.44>;tag=as246e6511

[williamconley]

Editing the callerid name field breaks vicidial. how long have you been here?

[ambiorixg12]

Editing the callerid name field breaks vicidial. how long have you been here?

Now you say that.I do remember, I readed a post where caller id name has special meaning for vicidial, but when you say it breaks, vicidial what it will exactly brake ?

[williamconley]

Editing the callerid name field breaks vicidial. how long have you been here?

Now you say that.I do remember, I readed a post where caller id name has special meaning for vicidial, but when you say it breaks, vicidial what it will exactly brake ?

I'm glad you asked. lol

Vicidial uses the CallerID Name field as an identifier for scripted control of the calls. Since these calls are migrated from place to place and "broken" during the process, asterisk standard variables get lost in the mix. Even the UniqueID gets changed occasionally. But the CallerID remains constant when these calls are shadowed from place to place. Since the caller id Number is visible to the public, but the caller id NAME is NOT visible to the public, that name field is where the identifier (a random character field after a fashion) is stored. In Canada this presents a problem since they allow passing this field to the public, but in the US the name field is dropped by all major carriers in favor of the caller id name field from their shared callername database (this is to prevent random callers from showing "Free Cell Phone" as their name, for instance).

The solution if you NEED to set the callerid Name field is to call yourself. Then generate a new call (not Vicidial controlled, just generated by Asterisk) which can have caller id name of anything you need. Adds load to the server, though, obviously. Alternately you can pass all your calls through a gateway of some sort and allow it to change the caller id name. Either way you're using more power to manage that name value.

Short version: CallerID Name is used to control the calls for autodialed calls. It's the handle used to grab the call and register it for call management. Without it ... there is no autodialed call since the ID never shows up to "get grabbed".

[ambiorixg12]

Thanks for the information William,