Campaign Caller ID same as Session

[iboam]

Hi everyone i hope all you guys are safe at home, im having an issue dialing out, yesterday talked with a customer and he said that my number was SCAM LIKELY then i called myself and notice that caller id wasn't a number, it was the the vicidial session (M3250414130000119058) how may i fix this ???

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register => blablabla:bleble@sip.trunk.com/blablabla

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[TRUNK]
username= blablabla
type= peer
secret= bleble
rfc2833compensate= yes
qualify= yes
nat=force_rport
insecure= port,invite
host= sip.trunk.com
dtmfmode= rfc2833
context= from-pstn-toheader
canreinvite= no

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UNIK = SIP/TRUNK

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exten => _151251NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _151251NXXNXXXXXX,n,Dial(${UNIK}/${EXTEN:5},,To)
exten => _151251NXXNXXXXXX,n,Hangup

exten => _15125NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _15125NXXNXXXXXX,n,Dial(${UNIK}/1${EXTEN:5},,tTo)
exten => _15125NXXNXXXXXX,n,Hangup

[williamconley]

http://www.vicidial.org/VICIDIALforum/v ... =4&t=38023

[iboam]

didn't work for me im dialing local numbers and still with the issue, it is random but happens 6 from 10 manual calls not sure if the campaign as ratio has the same behavior

[williamconley]

I have no idea what you just said or how it relates to the post I put in. The previous post contains a method to change your caller ID name. It also has a discussion about how "Where you are dialing" affects the Caller ID name in some areas.

However: If you are marked as "Scam Likely" the next question is whether the call recipients actually see the random string caller ID name sometimes and "scam likely" other times ... or whether they see "scam likely" sometimes and just your phone number (no random string) other times. This makes a huge difference, and you can NOT go by what YOU see when you call yourself if you are using a telephone company to receive the calls that is VOIP, since those are often related to one another and pass caller ID name very different than "Real" telephone companies (to which your call recipients will actually belong).

What Country do you dial? What do they see for caller ID name?

You can also usually just change the phone number by buying a new one every few days (it's only a couple bucks ...). Reduces the scam likely persona unless you are in one of those areas such as Canada where caller ID name is actually passed through. In that situation you must use the loopback to change your caller ID name unless your carrier has a method to alter it for you.

[iboam]

Fixed !!!!!!! just added a line to my carrier dial plan

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exten => _91NXXNXXXXXX,1,Set(CALLERID(name)=${CALLERID(num)})
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(${UNITRUNK}/1${EXTEN:2},,To)
exten => _91NXXNXXXXXX,n,Hangup

exten => _9NXXNXXXXXX,1,Set(CALLERID(name)=${CALLERID(num)})
exten => _9NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXNXXXXXX,n,Dial(${UNITRUNK}/1${EXTEN:1},,tTo)
exten => _9NXXNXXXXXX,n,Hangup

but i have a question, i had to change the carrier prefix to 9 and i have multiple carriers some reliable at termination and some not, how may i keep the prefix that i used to have before (5 numbers) ??

[iboam]

Sorry It's not fixed 100% when dial manual calls connect but still

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STATUS: Calling: (786)123-4561 UID: M3251820230000119061   Waiting for Ring... 15 seconds

im able to talk but no transfer the call it's like a dead/live call

Any suggestion ???

[williamconley]

Sorry It's not fixed 100% when dial manual calls connect but still

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STATUS: Calling: (786)123-4561 UID: M3251820230000119061   Waiting for Ring... 15 seconds

im able to talk but no transfer the call it's like a dead/live call

Any suggestion ???

Usually a version issue with Asterisk in the /etc/astguiclient.conf file and/or in admin->servers.

If the version is wrong in astguiclient.conf, you may have the wrong .conf files and need to run install.pl again ... and re-edit any customizations in the .conf files.

[iboam]

my question is if the dial plan is ok ???

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exten => _91NXXNXXXXXX,1,Set(CALLERID(name)=${CALLERID(num)})
exten => _91NXXNXXXXXX,n,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(${UNITRUNK}/1${EXTEN:2},,To)
exten => _91NXXNXXXXXX,n,Hangup

exten => _9NXXNXXXXXX,1,Set(CALLERID(name)=${CALLERID(num)})
exten => _9NXXNXXXXXX,n,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXNXXXXXX,n,Dial(${UNITRUNK}/1${EXTEN:1},,tTo)
exten => _9NXXNXXXXXX,n,Hangup

[williamconley]

Yes, but:

On autodialed calls, it won't actually affect the caller id name field. The Vicidial agi script will change the callerid name back to the random string. The previous link I posted should explain why you can't easily change that behavior, but it is changeable with the loopback dial method for which samples are supplied. But this is still based on your calling area and all that other stuff you didn't answer. lol

[iboam]

Im dialing local USA numbers mostly Florida but when dial manual call does not last, is like considered as dead call because all buttons are disabled even talking with the person, im not able to transfer calls

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exten => _91NXXNXXXXXX,1,Set(CALLERID(name)=${CALLERID(num)})
exten => _91NXXNXXXXXX,n,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(${UNITRUNK}/1${EXTEN:2},,To)
exten => _91NXXNXXXXXX,n,Hangup

exten => _9NXXNXXXXXX,1,Set(CALLERID(name)=${CALLERID(num)})
exten => _9NXXNXXXXXX,n,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXNXXXXXX,n,Dial(${UNITRUNK}/1${EXTEN:1},,tTo)
exten => _9NXXNXXXXXX,n,Hangup

when i remove the this first line

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exten => _91NXXNXXXXXX,1,Set(CALLERID(name)=${CALLERID(num)})

it dial fine as normal but continue the issue with the caller id as Session ID

[williamconley]

gonna suggest you go back and read that link i posted before. The caller ID name field is NOT used in the US after the call goes to the first telco. If you are actually using that same VOIP telco to GET the call, they may pass it. So that's not a real test. Call your cell phone instead.

[iboam]

Call connect, im able to talk but in the agent screen call never get active ....

[williamconley]

Show asterisk CLI of a test call. Not 3000 lines of unrelated code. One call, beginning to end. Best if there's no other traffic on the server while you run the test and capture the output. Of course you can modify any phone numbers or domain names or IPs to fake ones "to protect the innocent", lol.

[iboam]

Dial timed out, contact your system administrator

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[Mar 26 00:50:57]   == Manager 'sendcron' logged on from 127.0.0.1
[Mar 26 00:50:57]     -- Called 8600051@default
[Mar 26 00:50:57]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000023;2", "8600051,F") in new stack
[Mar 26 00:50:57]     -- Local/8600051@default-00000023;1 answered
[Mar 26 00:50:57]     -- Executing [91XXXXXXXXXX@default:1] Set("Local/8600051@default-00000023;1", "CALLERID(name)=1XXXXXXXXXX") in new stack
[Mar 26 00:50:57]     -- Executing [91XXXXXXXXXX@default:2] Dial("Local/8600051@default-00000023;1", "SIP/TRUNK/1XXXXXXXXXX,,tTor") in new stack
[Mar 26 00:50:57]   == Using SIP RTP CoS mark 5
[Mar 26 00:50:57]     -- Called SIP/TRUNK/1XXXXXXXXXX
[Mar 26 00:50:59]   == SRTCP unprotect failed because of authentication failure
[Mar 26 00:50:59]        > 0x7fa3e4011670 -- Strict RTP learning after remote address set to: 208.69.83.149:64194
[Mar 26 00:50:59]     -- SIP/TRUNK-0000002c is making progress passing it to Local/8600051@default-00000023;1
[Mar 26 00:51:01]     -- SIP/TRUNK-0000002c answered Local/8600051@default-00000023;1
[Mar 26 00:51:01]     -- Channel SIP/TRUNK-0000002c joined 'simple_bridge' basic-bridge <40b515bf-57bb-4c2c-ac54-73172a2432d3>
[Mar 26 00:51:01]     -- Channel Local/8600051@default-00000023;1 joined 'simple_bridge' basic-bridge <40b515bf-57bb-4c2c-ac54-73172a2432d3>
[Mar 26 00:51:01]        > 0x7fa3e4011670 -- Strict RTP switching to RTP target address 208.69.83.149:64194 as source
[Mar 26 00:51:02]   == SRTCP unprotect failed because of authentication failure
[Mar 26 00:51:04]        > 0x7fa3e4011670 -- Strict RTP learning complete - Locking on source address 208.69.83.149:64194
[Mar 26 00:51:04]   == SRTCP unprotect failed because of authentication failure
[Mar 26 00:51:30]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/8600051@default-00000023;2
[Mar 26 00:51:30]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000023;2'
[Mar 26 00:51:30] WARNING[22104][C-00000051]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Mar 26 00:51:30]     -- Executing [h@default:1] AGI("Local/8600051@default-00000023;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
[Mar 26 00:51:30]     -- <Local/8600051@default-00000023;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
[Mar 26 00:51:30]     -- Channel Local/8600051@default-00000023;1 left 'simple_bridge' basic-bridge <40b515bf-57bb-4c2c-ac54-73172a2432d3>
[Mar 26 00:51:30]     -- Channel SIP/TRUNK-0000002c left 'simple_bridge' basic-bridge <40b515bf-57bb-4c2c-ac54-73172a2432d3>
[Mar 26 00:51:30]   == Spawn extension (default, 91XXXXXXXXXX, 2) exited non-zero on 'Local/8600051@default-00000023;1'
[Mar 26 00:51:30]     -- Executing [h@default:1] AGI("Local/8600051@default-00000023;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----32-----32-----SIP 200 OK)") in new stack
[Mar 26 00:51:30]     -- <Local/8600051@default-00000023;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----32-----32-----SIP 200 OK) completed, returning 0
[Mar 26 00:51:34]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/8421-0000002a
[Mar 26 00:51:34]     -- Hungup 'DAHDI/pseudo-1078862578'
[Mar 26 00:51:34]   == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/8421-0000002a'
[Mar 26 00:51:34]     -- Executing [h@default:1] AGI("SIP/8421-0000002a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK)") in new stack
[Mar 26 00:51:34]     -- <SIP/8421-0000002a>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK) completed, returning 0
[Mar 26 00:51:34]     -- Called 55558600051@default
[Mar 26 00:51:34]     -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000024;2", "8600051,K") in new stack
[Mar 26 00:51:34] WARNING[22231][C-00000053]: app_meetme.c:5253 admin_exec: Conference number '8600051' not found!
[Mar 26 00:51:34]     -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000024;2", "") in new stack
[Mar 26 00:51:34]   == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000024;2'
[Mar 26 00:51:34] WARNING[22231][C-00000053]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Mar 26 00:51:34]     -- Executing [h@default:1] AGI("Local/55558600051@default-00000024;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Mar 26 00:51:34]     -- <Local/55558600051@default-00000024;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0


[williamconley]

Is this a manual dial call? Has this system ever worked?

Is the IP correct in admin->servers? Have you altered the IP after installation at all? Is the asterisk version correct in /etc/astguiclient.conf and in admin->servers?

Did you follow the steps in the Vicidial Manager's Manual starting at page one and not stopping until you had a fully functional system? Or did you head off in your own direction?

[iboam]

System is fully functional, it's a fresh install, i'm having this caller id issue always, im having this issue using Manual Dial and Ratio, sometimes customers says that they don't get to much calls but when you see a call with a caller id M345837457394857984 you usually don't get the call and the only one method i have to test it is dialing myself and it hapen every time (almost), today i called a customer and his phone number answer me that is not able to take calls from blocked numbers then i added the string to the dial plan and call connect but after 18 secs i got the error (Dial timed out, contact your system administrator)

[williamconley]

sometimes customers says ...

vs

but when you see a call with a caller id M345837457394857984

You've made a logical jump here without stating that a client has told you that is what they see.

his phone number answer me that is not able to take calls from blocked numbers

Once again, you don't seem to be testing this on a regular phone line such as a cell phone company or NON-voip line to find out what is actually being seen by the call recipient.

All the logic in the world, all the decisions and hair pulling can't replace a real-world test.

It could just be that your system is broadcasting the agent's phone number (which is often 0000000000) or broken in some other way. Test it real world. And remember that Manual Dial and Autodial must be handled separately for testing. They are not related.

[iboam]

I mean no there is not 0000000000 as caller number, it's the caller name, i have T-Mobile and when i dial my number in 10 calls i have at least 4 with caller name with a string (Capital letter followed by numbers) i believe is something related to the carrier, i tried a different carrier today and didn't happen in 20 calls to my cell phone, then i back to same provider and happen again.

[williamconley]

I'm going out on a limb and suggesting that your VOIP carrier has a direct link to TMobile.

In which case for T-Mobile calls, use the previous link to manually set your callerID name with the Loopback method.

That is why I provided it.

[mflorell]

One option is that you can strip that out by using OpenSIPs as a SIP gateway. We've actually had to set this up for several clients recently anyway because of some other SIP carrier issues relating to outbound phone calls through those carriers.

[iboam]

how does it works ???

[mflorell]

We basically use VICInoc(http://vicinoc.com/) as the framework, then use OpenSIPs rules to remove the CallerIDname as calls are sent out to the carriers.

[williamconley]

how does it works ???

Requires a second server running SIP server software. Pass all calls through the second server and it can alter Caller ID information on the way through.

[iboam]

I have used 3 different carriers in the last week but the issue by itself is in Vicidial, T-Mobile have a service included for Metro PCS and T-Mobile for enhanced called id, that if the caller name is not a number or a verified caller name call won't connect, when i include this code in my dial plan it force the caller name to be same as caller id on the campaing, no matter if i dialing manual or ratio or custom caller id by state or area code

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exten => _91NXXNXXXXXX,1,Set(CALLERID(name)=${CALLERID(num)})

but all features on the agent page are disabled (conference, park) while the actual call is connected. Any solution for force the caller name same as caller id plaease

[williamconley]

One more time: the original link shows a method to modify the callerid name (to whatever you want).

[iboam]

I just need that the Caller Name on my customer phone be the same as the caller id, i just called someone manually and the caller name was M4021829080000119308 even when he call me back was a phone number but caller name was the same UID: M4021829080000119308

[iboam]

Hi everyone and thanks for all your help and support, found this dial plan entry exten => _03261NXXNXXXXXX,n,NoOP(${CALLERID(name)}) and got this

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[Apr  3 17:14:43]     -- Executing [03261XXXXXXXXXX@default:2] NoOp("Local/03261XXXXXXXXXX@default-00000057;2", "V4031714430000044889") in new stack
[Apr  3 17:14:43]     -- Executing [03261XXXXXXXXXX@default:3] Dial("Local/03261XXXXXXXXXX@default-00000057;2", "SIP/TRUNK/1XXXXXXXXXX,,tTor") in new stack


any help