vicidial.org

by **Nefariousparity** » Mon Oct 07, 2013 12:22 pm

So I have two servers. One is running a survey. The other is running predictive. Both servers are running the same software as my signature says.

When someone opts in on my survey, the remote agent is instructed to call a inbound number on the predictive server. The inbound is instructed to transfer to a in-group of four agents.

I have the same data in the survey server as I do in the predictive server.

**Goal**
The goal is to get the customer information from the survey server to show up on the predictive server.

So I guess, is the have the customer phone number transfer over from the survey server over to the ingroup on the predictive, and then look for the relevant data and populate the agents screen accordingly.

Thanks in advance!

**EDIT**

Or do I need to like add the servers to each other so that the calls are locally transferred? I think what is happening is that the survey server is transferring with my AC-CID settings, and not the customer number.

by **williamconley** » Tue Oct 08, 2013 12:57 am

You should have the Survey server transfer the call to an ingroup, then use an on-hook remote agent on the survey server to call the other server. The Ingroup on the Survey server will have a setting to allow you to transmit the CUSTOMER phone when dialing to the remote agent. Since the remote agent will be the other server, you can then use CIDLOOKUP to acquire the data on the other server.

The trick is that the Ingroup and On-Hook agent are required to transmit the phone number of the customer. That's the only place this will work when outbound dialing (if at all) since there is no such setting in a campaign, only in an ingroup with on-hook agents.

by **Nefariousparity** » Wed Oct 09, 2013 11:10 pm

William, is there anything you can't do! Thank you so much!

by **Nefariousparity** » Mon Oct 28, 2013 1:28 pm

Looking back....
William how would one have the survey transfer call to to ingroup?

by **williamconley** » Mon Oct 28, 2013 2:06 pm

Since you can't go directly (based on your question, or I'm sure you would not have asked) ... send it to a call menu and drop to the ingroup from there.

by **Nefariousparity** » Mon Oct 28, 2013 2:48 pm

So there are a few ideas I had about this.
First being the callmenu, however I don't want the initially message to be placed twice. (Opening Message from Survey, and if they press 1, sent to call Menu and hear the message again.)

by **williamconley** » Mon Oct 28, 2013 4:26 pm

Not to be condescending, but I would then recommend you leave the message blank in the call menu and just drop it after 1 sec to the ingroup. 8-)

by **Nefariousparity** » Mon Oct 28, 2013 4:44 pm

Hmm I have set all this up before this way. I will try it again.

My issue at the moment, is that when the other server gets the call. It us showing something like "Y0281518330000154869" instead of customer phone number.

*** Edit ***
Also with your last message it almost sounds like you are referring to a singe server setup.

I have two servers.

Server (A)
Has Blended Campaign,
--DID(555-555-5552)
---Ingroup(555-555-5552)
-----Agents

Server (B)
Survey Campaign
--(Remote_Agent)8080(555-555-5551)-OutBound
--(Remote_Agent)9090(555-555-5552)-Inbound

Ingroups are set as customer closer.

Every time a number is transferred, it is not showing customer number, it is showing weird like session numbers. "Y0281518330000154869".

Now if I call in to DID on Server A. It shows my phone number no problem?

by **williamconley** » Mon Oct 28, 2013 8:03 pm

Ingroups have an option for the callerid to show on an on-hook agent's phone. Choose "customer".

by **Nefariousparity** » Tue Oct 29, 2013 3:03 pm

When you say this, are you meaning."Action Transfer CID"

by **williamconley** » Tue Oct 29, 2013 9:24 pm

On-Hook CID - This option is only used for agents that are logged in with phones that have the agent-on-hook feature enabled. This is the caller ID that will show up on their agent phones when the calls are ringing. GENERIC is a generic RINGAGENT00000000001 type of notification. INGROUP will show only the in-group the call came from. CUSTOMER_PHONE will show only the customer phone number. CUSTOMER_PHONE_RINGAGENT will show RINGAGENT_3125551212 with the RINGAGENT as part of the CID with the customer phone number. CUSTOMER_PHONE_INGROUP will show the first 10 characters of the in-group followed by the customer phone number. Default is GENERIC.

by **Nefariousparity** » Wed Oct 30, 2013 2:53 pm

Agent Screen (Server A)

Digram of Call Flow

I had this setup and working fine before, and I have no idea why it is not working now. My assumption is something in A2billing, but what blows my mind is why I can call directly in with my cell phone and it works fine. Customers when they call us back it works fine. It is just not liking the transfers from remote agents in Survey server.

by **Nefariousparity** » Wed Oct 30, 2013 4:16 pm

Something I also notice, is that it is getting my phone number because I could see it in the log of the recording.

AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20131030135625_6062834002_6662934546)

If this option is set to
In-Group Call Handle Method:CID

Then I get the wierd numbers.

If set to anything else(Not Counting VID) I get a blank?

Its like setting the UID as the phone number.

by **williamconley** » Wed Oct 30, 2013 9:14 pm

If you are transferring the calls between two servers and they are not CLUSTERED, this would be good to mention. LOL

So a completely fresh outbound phone call is being generated to different server and you are passing that call through yet another service (A2B) and somewhere in there A2B or the other Vicidial server is grabbing the CallerID Name instead of the CallerID Number for reference.

You'll need to check some sip packets to find out which one.

Also: are you "transferring" or are you "3-way call"ing? If you are 3-way calling, there is a different control in the campaign to determine the callerid of the 3-way call.

As i see your diagram, it seems you are making and taking calls on your primary system and upon qualifying a lead you are sending the call to the secondary system through A2B to invoice a client for the call. This means that the call in the 2nd server is not really a "transfer" in the second system, it's merely an inbound call which was generated as an outbound 3-way call in the primary system using A2B as the carrier which passed the call to the secondary dialer. These two dialers have no inherent data sharing other than linking calls with callerid, so you need to carry the callerid around as a tracker.

by **Nefariousparity** » Wed Oct 30, 2013 11:06 pm

William, you are 100% accurate. This is how I set it up once, and it was working, but not it is not. And I am at a loss. I thought I was brilliant until I could not get it work again.

by **williamconley** » Thu Oct 31, 2013 12:14 am

did you check the 3-way call callerid for the campaign?

also, in some instances, it works to modify the outbound carrier for that outbound 3-way call to NOT include the agi line ...

by **Nefariousparity** » Thu Oct 31, 2013 1:13 am

I tried option one, set it to customer Getting UID still.
The second thing you mention I have no idea of how to go about doing that.

Just dd SIP DEBUG

---

[Klinux-57aw*CLI>
[0K[Oct 30 23:40:25]
<--- SIP read from 192.187.121.218:5060 --->
INVITE sip:9092844002@pd02.baretelecom.com SIP/2.0
Via: SIP/2.0/UDP 192.187.121.218:5060;branch=z9hG4bK10869874;rport
Max-Forwards: 70
From: "V0302340040000154906" <sip:9412844546@192.168.1.218>;tag=as3a4be7a3
To: <sip:9082746002@pd02.baretelecom.com>
Contact: <sip:9512944547@192.187.121.218:5060>
Call-ID: 0987c62225409a866ad1bd612529d815@192.168.1.218:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8.22.0)
Date: Thu, 31 Oct 2013 06:40:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 317

It is definitely doing the "V0302340040000154906" and not the number.

by **williamconley** » Thu Oct 31, 2013 12:35 pm

post the carrier dialplan for the carrier you use to send the call to the a2b machine. i presume a2b is set up as a carrier under admin->carriers ...?

by **Nefariousparity** » Thu Oct 31, 2013 2:59 pm

Yes William, you are correct.

[baretelecom]
username=****
type=friend
secret=****
host=sip.*****
fromuser=****
context=trunkoutbound
allow=ulaw,alaw
trustrpid = yes
sendrpid = yes
canreinvite = no
insecure=port,invite
################################
exten => _71NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _71NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@baretelecom,,tTor)
exten => _71NXXNXXXXXX,3,Hangup
################################

So currently I had my carrier give me another phone number, and terminate at Server Predictive (A). Server (B) is calling, and transferring directly through carrier, and when customer presses 1 going straight to other servers number which comes from carrier. So in short completely bypassing A2B.

Some numbers, seem like they are working correctly, and sometimes I still see, "V0302340040000154906". What I can't discern right now is if the correct showing up numbers were actually transferred or people calling back.

Thanks William!

by **williamconley** » Thu Oct 31, 2013 3:51 pm

you should not be passing these calls through a carrier to get to the 2nd server whether you use A2B or not. just wasting money. your calls CAN go straight to the 2nd vicidial server by setting IT up as the carrier.

But with this method ... Try modifying the dialplan like this:

Code: Select all: exten => _71NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _71NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@baretelecom,,tTor) exten => _71NXXNXXXXXX,3,Hangup exten => _771NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@baretelecom,,tTor)

And use "77" for the 3-way call dial prefix. This will not allow vicidial to link to the call and modify anything, it will merely generate the call. if the campaign is set to use the customer's callerid ... you could get lucky.

by **Nefariousparity** » Thu Oct 31, 2013 4:05 pm

Here we go.... Testing now.

by **Nefariousparity** » Thu Oct 31, 2013 4:15 pm

Hmm through the carrier, it is doing the same thing. And a issue I have using through a2billing is showing customer did's, and not customer number. So of survey calls customer,through a2b, and then upon pressing 1 calls other server through a2b, Its going to show the survey servers DID on the predictive side.

/me starts pulling whats left of my hair out.

by **williamconley** » Thu Oct 31, 2013 7:59 pm

did you change the 3-way call callerid to customer? (and are you using 3-way calling for the agent to transfer the call to the other server?)

by **Nefariousparity** » Fri Nov 01, 2013 12:43 am

Yes, I changed the 3-way callerID. I assume yes? It is a remote agent with the exstension pointing to inbound line on the other server. I have tried a DID running through a2billing and a line that is from the carrier.

by **williamconley** » Fri Nov 01, 2013 2:06 pm

Just because this has already taken an unexpected twist (servers not clustered), it would be good to describe the entire process an path of the call (each button press).

by **Nefariousparity** » Fri Nov 01, 2013 4:22 pm

I totally understand. So here is the break down. I am working feverishly on this. Not because I am under a deadline, well I am but nothing very soon. I like to solve problems, and consider myself to be pretty knowledgeable now with my two years of asterisk, and vicidial under my belt. The fact that this is eluding
me... This is not a step by step, however it is all my configuration settings. Hope this helps, thank you so much William!

Now this is just all the configuration settings for the "SURVEY SERVER". Not the server which is running the predictive campaign with all my agents.
These servers are two separate machines with on different internet. Of course NOT CLUSTERED.

The Goal:
To have the survey server call customer, upon "Press 1" call the inbound line of my other server, and have the customers (callerID PHONE NUMBER) be what is displayed in the agent interface, and match the data so the agent knows the FN LN ADDR ZIP CTY etc. Currently, what is happinging, is not mtater what I do, the (CallerID NAME) which I believe is the UID number(See Post Above) is what is showing up, which of course does not match any records.
I am open to better ways of doing this of course. I have clustered servers before, separate server for asterisk, db, http. I have never done the duel asterisk really. At least not dialer

##What the othe server sees with set debug on##
[Klinux-57aw*CLI>
[0K[Oct 30 23:40:25]
<--- SIP read from 192.187.121.218:5060 --->
INVITE sip:9092844002@pd02.baretelecom.com SIP/2.0
Via: SIP/2.0/UDP 192.187.121.218:5060;branch=z9hG4bK10869874;rport
Max-Forwards: 70
From: "V0302340040000154906" <sip:9412844546@192.168.1.218>;tag=as3a4be7a3
To: <sip:9082746002@pd02.baretelecom.com>
Contact: <sip:9512944547@192.187.121.218:5060>
Call-ID: 0987c62225409a866ad1bd612529d815@192.168.1.218:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8.22.0)
Date: Thu, 31 Oct 2013 06:40:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 317

"Contact is actually my cell phone number"
##########################################

Server(B) Survey Server (Separate Machine Entirely)
||VERSION: 2.8-409a||DB Schema Version:1355||1.8.23.0-vici||BUILD: 130809-1410
########################################################################
-=-=Dial Plan=-=-
[baretelecom]
username=****
type=friend
secret=****
host=sip.*****
fromuser=****
context=trunkoutbound
allow=ulaw,alaw
trustrpid = yes
sendrpid = yes
canreinvite = no
insecure=port,invite

exten => _71NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _71NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@baretelecom,,tTor)
exten => _71NXXNXXXXXX,3,Hangup
exten => _771NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@baretelecom,,tTor)
-=-=Dial Plan=-=-
########################################################################
-=-=Create Campaign=-=-
Campaign ID: 1001
Campaign Name: BroadCast
Campaign Description: BroadCast
Campaign Change Date: 2013-11-01 10:56:16
Campaign Login Date: 2013-11-01 13:14:36
Campaign Call Date: 2013-11-01 11:01:02
8 Day outbound call count for this campaign more summary stats...
date call count
2013-11-01 2
2013-10-31 45
2013-10-30 15
2013-10-29 14
2013-10-28 12
2013-10-27 -none-
2013-10-26 19
2013-10-25 -none-
Active: Y
Admin User Group: --ALL--
Park Music-on-Hold: default
Web Form:
Web Form Two:
Web Form Target: vdcwebform
Allow Closers: Y
Allow Inbound and Blended: Y
Dial Status 1: SVYCLM - Survey sent to Call Menu REMOVE
Dial Status 2: SVYEXT - Survey sent to Extension REMOVE
Dial Status 3: DROP - Agent Not Available REMOVE
Dial Status 4: XFER - Call Transferred REMOVE
Dial Status 5: PM - Played Message REMOVE
Dial Status 6: A - Answering Machine REMOVE
Dial Status 7: AA - Answering Machine Auto REMOVE
Dial Status 8: NA - No Answer AutoDial REMOVE
Dial Status 9: NEW - New Lead REMOVE
Add A Dial Status to Call:
List Order: DOWN
List Order Randomize: N
List Order Secondary: LEAD_ASCEND
List Mix: DISABLED
Lead Filter: NONE -
Drop Lockout Time: 0
Call Count Limit: 0
Call Count Target: 10
Minimum Hopper Level: 100
Automatic Hopper Level: Y
Automatic Hopper Multiplier: 1
Auto Trim Hopper: Y
Hopper VLC Dup Check: N
Force Reset of Hopper: N
Dial Method: RATIO
Auto Dial Level: 10 (0 = off) ADAPT OVERRIDE
Auto Dial Level Threshold: DISABLED agents:
Available Only Tally: N
Available Only Tally Threshold: DISABLED agents:
Drop Percentage Limit: 3%
Maximum Adapt Dial Level:3.0 number only
Latest Server Time: 2100 4 digits only
Adapt Intensity Modifier: 0 - BALANCED
Dial Level Difference Target: 0 --- 0 BALANCED
Dial Level Difference Target Method: ADAPT_CALC_ONLY
Concurrent Transfers: AUTO
Queue Priority: 50 - Higher
Multiple Campaign Drop Rate Group: DISABLED
Inbound Queue No Dial: DISABLED
Auto Alt-Number Dialing: NONE
Next Agent Call: longest_wait_time
Local Call Time: 24hours - default 24 hours calling
State rules defined for this call time: -1
Dial Timeout: 20 in seconds
Dial Prefix:71 for 91NXXNXXXXXX value would be 9, for no dial prefix use X
Manual Dial Prefix: 71
Omit Phone Code: N
Campaign CallerID: 9092844004
Custom CallerID: Y
Routing Extension:8366 (For testing with my cell phone) Typically (8374)
Campaign Rec exten:8309
Campaign Recording: ALLFORCE
Campaign Rec Filename: FULLDATE_CUSTPHONE_AGENT_LEADID
Recording Delay: 2 in seconds
Call Notes Per Call: DISABLED
Agent Lead Search: DISABLED
Agent Lead Search Method: CAMPLISTS_ALL
Script:
Get Call Launch: NONE
Answering Machine Message: vm-goodbye
WaitForSilence Options: 0
AMD send to Action: N
CPD AMD Action: DISABLED
AMD Inbound Group: ---NONE---
AMD Call Menu: ---NONE---
Transfer-Conf DTMF 1:
Transfer-Conf Number 1:
Transfer-Conf DTMF 2:
Transfer-Conf Number 2:
Transfer-Conf Number 3:
Transfer-Conf Number 4:
Transfer-Conf Number 5:
Enable Transfer Presets: DISABLED
Hide Transfer Number to Dial: DISABLED
Quick Transfer Button:N
Custom 3-Way Button Transfer: DISABLED
PrePopulate Transfer Preset: N
Park Call IVR: DISABLED
Park Call IVR AGI:
Timer Action: NONE
Timer Action Message: NONE
Timer Action Seconds: 1
Timer Action Destination:
Alt Number Dialing: N
Scheduled Callbacks:N
Scheduled Callbacks Alert: NONE
Scheduled Callbacks Count: ALL_ACTIVE
Scheduled Callbacks Days Limit: 0
Scheduled Callbacks Hours Block: 0
Scheduled Callbacks Calltime Block: DISABLED
My Callbacks Checkbox Default: unchecked
Drop Call Seconds: 5
Drop Action: CALLMENU
Safe Harbor Exten: 8307
Safe Harbor Audio: buzz
Safe Harbor Audio Field: DISABLED
Safe Harbor Call Menu: 300
Voicemail: 7000
Drop Transfer Group: 1014 - SurveyAgent CustomerCID
Disable Dispo Screen: DISP_ENABLED
Disable Dispo Status:
Wrap Up Seconds: 0
Wrap Up Message: Wrapup Call
Use Internal DNC List: Y
Use Campaign DNC List: Y
Other Campaign DNC:
Agent Pause Codes Active: N
Auto Pause Pre-Call Work: N
Auto Resume Pre-Call Work: N
Auto Pause Pre-Call Code: PRECAL
Campaign Stats Refresh: N
Real-Time Agent Time Stats: CALLS_WAIT_CUST_ACW_PAUSE
Disable Alter Customer Data: N
Disable Alter Customer Phone: Y
Allow No-Hopper-Leads Logins: Y
No Hopper Dialing: N
Owner Only Dialing: NONE
Owner Populate: DISABLED
Agent Display Dialable Leads: N
Agent Screen Labels: ---SYSTEM-SETTINGS---
Status Display Fields: CALLID
Agent Display Queue Count: Y
Agent View Calls in Queue: NONE
View Calls in Queue Launch: MANUAL
Agent Grab Calls in Queue: N
Agent Call Re-Queue Button: N
Agent Pause After Each Call: N
Agent Pause After Next Call Link: DISABLED
Manual Dial Override: NONE
Manual Dial List ID: 998
Manual Dial Filter: NONE
Manual Preview Dial: PREVIEW_AND_SKIP
Manual Call Time Check: DISABLED
Manual Dial API: STANDARD
Manual Dial CID: CAMPAIGN
Phone Post Time Difference Alert: DISABLED
In-Group Manual Dial: DISABLED
In-Group Manual Dial Select: CAMPAIGN_SELECTED
Agent Screen Clipboard Copy: NONE
Agent Screen Extended Alt Dial: N
3-Way Call Outbound CallerID: CUSTOMER
3-Way Call Dial Prefix: 77
Customer 3-Way Hangup Logging: ENABLED
Customer 3-Way Hangup Seconds: 5
Customer 3-Way Hangup Action: NONE
Group Alias Allowed: N
CRM Popup Login: N
CRM Popup Address:
Start Call URL:
Dispo Call URL:
No Agent Call URL:
Extension Append CID: N
Blind Monitor Warning: DISABLED
Blind Monitor Notice: Someone is blind monitoring your session.
Blind Monitor Filename: audio chooser
Allowed Inbound Groups:
1010 - Next-Day-Denise - 0
1011 - Future-Chelsey - 0
1012 - Call-Backs - 0
1013 - Verify-Enrique - 0
1014 - Survey Agent CustomerCID - 0
AGENTDIRECT - Single Agent Direct Queue - 99
Default Transfer Group: ---NONE---
Allowed Transfer Groups:
1010 - Next-Day-Denise
1011 - Future-Chelsey
1012 - Call-Backs
1013 - Verify-Enrique
1014 - Survey Agent CustomerCID
AGENTDIRECT - Single Agent Direct Queue

-=-=Create Campaign (BroadCast)=-=-
##################################################################
##################################################################
-=-=Survey Settings=-=-
Survey First Audio File: 85100008
Survey DTMF Digits: 123
Survey Not Interested Digit: 3
Survey Wait Seconds: 60
Survey Opt-in Audio File:request_has_been_processed
Survey Not Interested Audio File:not_interested_or_dnc
Survey Method: AGENT_XFER
Survey No-Response Action: OPTOUT
Survey Not Interested Status: NI-Not Interested
Survey Third Digit: 3
Survey Third Audio File: US_thanks_no_contact
Survey Third Status: NI
Survey Third Extension: 8300
Survey Fourth Digit:
Survey Fourth Audio File: US_thanks_no_contact
Survey Fourth Status: NI
Survey Fourth Extension: 8300
Survey Response Digit Map: 1-Agent|2-VoiceMail|3-OPTOUT|X-NO RESPONSE|
Survey Survey Xfer Extension: 8300
Survey Campaign Recording Directory: /home/survey
Voicemail: 7000
Survey Call Menu: 3000
Survey Recording: Y
-=-=Survey Settings=-=-
###################################################################

by **williamconley** » Fri Nov 01, 2013 6:33 pm

but i did ask for a step by step ... so you gave me everything BUT the step by step? That's funny. If you worked here I'd pick on you for that for at least a week. LOL

On the other hand, it would appear that there is a phone number in the "From". Quite a bit in the from and to ... are any of them the number that you are trying to send?

Code: Select all: From: "V0302340040000154906" <sip:9412844546@192.168.1.218>;tag=as3a4be7a3

by **Nefariousparity** » Fri Nov 01, 2013 11:12 pm

No the number that I want is the 951 294 4547. That is the number I am using in a test lead I created. Call that number, play the message press 1, and transfer over to inbound\ingroup on the other server.

LOL yeah sorry.

by **williamconley** » Fri Nov 01, 2013 11:59 pm

in which case we have at least determined that the system making the call is generating the wrong callerid. Now we need to figure out why.

agi debug from the generation of the phone call would be useful in that regard. (as would a step by step procedure for how the call was generated)

by **Nefariousparity** » Sun Nov 03, 2013 1:22 am

When you say, AGI debug, are you telling me to "sip set debug"
-------------------------------------------------------------------------
In which case,
1) Login
2) Click List
3) Click add new list
-4) List ID: 1212
-5) List Name: Test List
-6) List Description: Test List
-7) Campaign: 1000 - Broadcast

Click Add a new Lead
9) Set list to Test List "1212 - 1001 Test List"
10) Go down to phone number put my cell phone in
-----------------------------------------------------------
11) Activate remote agent, await phone call and message.
12) Listen to message,
13) Watch Real Time report of other (predictive) server to see what callerID comes in.
---------------------------------------------------------------
AGI-Output of Survey Server
---------------------------------------------------------------
When you say, AGI debug, are you telling me to "sip set debug"
-------------------------------------------------------------------------
In which case,
1) Login
2) Click List
3) Click add new list
-4) List ID: 1212
-5) List Name: Test List
-6) List Description: Test List
-7) Campaign: 1000 - Broadcast

Click Add a new Lead
9) Set list to Test List "1212 - 1001 Test List"
10) Go down to phone number put my cell phone in
-------------------------------------------------------------------------
-------------------------------------------------------------------------
11) Activate remote agent, await phone call and message.
12) Listen to message,
13) Watch Real Time report of other server to see what callerID comes in.
-------------------------------------------------------------------------

-- Executing [719512944547@default:1] [1;36mAGI[0m("[1;35mLocal/719512944547@default-0004898a;2[0m", "[1;35magi://127.0.0.1:4577/call_log[0m") in new stack

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:03] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=1001))

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:03] -- <Local/719512944547@default-0004898a;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:03] -- Executing [719512944547@default:2] [1;36mDial[0m("[1;35mLocal/719512944547@default-0004898a;2[0m", "[1;35mSIP/19512944547@Carrier-X_OutBound,,tTor[0m") in new stack

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:03] == Using SIP RTP CoS mark 5

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:03] Audio is at 16870

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:03] Adding codec 0x4 (ulaw) to SDP

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:03] Adding codec 0x2 (gsm) to SDP

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:03] Adding non-codec 0x1 (telephone-event) to SDP

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:03] Reliably Transmitting (NAT) to *.*.*.*:5060:
INVITE sip:19512944547@*.*.*.*:5060 SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK0e157397;rport
Max-Forwards: 70
From: "V1022305030000154916" <sip:9092844004@*.*.*.*>;tag=as2d8112de
To: <sip:19512944547@*.*.*.*:5060>
Contact: <sip:9092844004@*.*.*.*:5060>
Call-ID: 0ee64ddb79271ab12801e9b015946999@*.*.*.*:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0-vici
Date: Sun, 03 Nov 2013 06:05:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "V1022305030000154916" <sip:9092844004@*.*.*.*>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1278045625 1278045625 IN IP4 *.*.*.*
s=Asterisk PBX 1.8.23.0-vici
c=IN IP4 *.*.*.*
t=0 0
m=audio 16870 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:03] -- Called SIP/19512944547@Carrier-X_OutBound

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:04]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK0e157397;rport=5060
From: "V1022305030000154916" <sip:9092844004@*.*.*.*>;tag=as2d8112de
To: <sip:19512944547@*.*.*.*:5060>
Call-ID: 0ee64ddb79271ab12801e9b015946999@*.*.*.*:5060
CSeq: 102 INVITE
Server: Carrier-X Partner/1.1
Content-Length: 0

<------------->
[Nov 2 23:05:04] --- (8 headers 0 lines) ---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:04]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK0e157397;rport=5060
Record-Route: <sip:*.*.*.*:5060;lr>
From: "V1022305030000154916" <sip:9092844004@*.*.*.*>;tag=as2d8112de
To: <sip:19512944547@*.*.*.*:5060>;tag=xtg-2355719-578489133
Contact: <sip:19512944547@38.102.250.158:5060>
Call-ID: 0ee64ddb79271ab12801e9b015946999@*.*.*.*:5060
CSeq: 102 INVITE
Server: Carrier-X Carrier/1.0
Content-Type: application/sdp
Content-Length: 229

v=0
o=NYMSX2 1716983003 1318585570 IN IP4 199.73.84.76
s=sip call
c=IN IP4 199.73.84.87
t=0 0
m=audio 60028 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
[Nov 2 23:05:04] --- (11 headers 11 lines) ---
[Nov 2 23:05:04] list_route: hop: <sip:*.*.*.*:5060;lr>

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:04] Found RTP audio format 0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:04] Found RTP audio format 101

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:04] Found audio description format PCMU for ID 0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:04] Found audio description format telephone-event for ID 101

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:04] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:04] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:04] Peer audio RTP is at port 199.73.84.87:60028

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:04] -- SIP/Carrier-X_OutBound-00048bb0 is making progress passing it to Local/719512944547@default-0004898a;2

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:06] == Manager 'sendcron' logged on from 127.0.0.1

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:06] == Manager 'sendcron' logged off from 127.0.0.1

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK0e157397;rport=5060
Record-Route: <sip:*.*.*.*:5060;lr>
From: "V1022305030000154916" <sip:9092844004@*.*.*.*>;tag=as2d8112de
To: <sip:19512944547@*.*.*.*:5060>;tag=xtg-2355719-578489133
Contact: <sip:19512944547@38.102.250.158:5060>
Call-ID: 0ee64ddb79271ab12801e9b015946999@*.*.*.*:5060
CSeq: 102 INVITE
Server: Carrier-X Carrier/1.0
Content-Type: application/sdp
Content-Length: 229
Require: timer
Session-Expires: 3600;refresher=uas

v=0
o=NYMSX2 1716983003 1318585570 IN IP4 199.73.84.76
s=sip call
c=IN IP4 199.73.84.87
t=0 0
m=audio 60028 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] --- (13 headers 11 lines) ---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] list_route: hop: <sip:*.*.*.*:5060;lr>
[Nov 2 23:05:11] set_destination: Parsing <sip:*.*.*.*:5060;lr> for address/port to send to
[Nov 2 23:05:11] set_destination: set destination to *.*.*.*:5060
[Nov 2 23:05:11] Transmitting (NAT) to *.*.*.*:5060:
ACK sip:19512944547@38.102.250.158:5060 SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK2c42e0b7;rport
Route: <sip:*.*.*.*:5060;lr>
Max-Forwards: 70
From: "V1022305030000154916" <sip:9092844004@*.*.*.*>;tag=as2d8112de
To: <sip:19512944547@*.*.*.*:5060>;tag=xtg-2355719-578489133
Contact: <sip:9092844004@*.*.*.*:5060>
Call-ID: 0ee64ddb79271ab12801e9b015946999@*.*.*.*:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0-vici
Content-Length: 0

---
[Nov 2 23:05:11] -- SIP/Carrier-X_OutBound-00048bb0 answered Local/719512944547@default-0004898a;2

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] > Channel Local/719512944547@default-0004898a;1 was answered.

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- Executing [8366@default:1] [1;36mPlayback[0m("[1;35mLocal/719512944547@default-0004898a;1[0m", "[1;35msip-silence[0m") in new stack

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- <Local/719512944547@default-0004898a;1> Playing 'sip-silence.gsm' (language 'en')

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- Executing [h@default:1] [1;36mAGI[0m("[1;35mLocal/719512944547@default-0004898a;2[0m", "[1;35magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----8-----0[0m") in new stack

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- Executing [8366@default:2] [1;36mAGI[0m("[1;35mSIP/Carrier-X_OutBound-00048bb0[0m", "[1;35magi://127.0.0.1:4577/call_log[0m") in new stack

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=1001))

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- <SIP/Carrier-X_OutBound-00048bb0>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- Executing [8366@default:3] [1;36mAGI[0m("[1;35mSIP/Carrier-X_OutBound-00048bb0[0m", "[1;35magi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB[0m") in new stack

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20131102-230511_9512944547_VDAD_154916)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:11] -- Playing '85100008' (escape_digits=123) (sample_offset 0)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:12] == Manager 'sendcron' logged off from 127.0.0.1

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:12] -- <Local/719512944547@default-0004898a;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---8-----0 completed, returning 0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:12] == Spawn extension (default, 719512944547, 2) exited non-zero on 'Local/719512944547@default-0004898a;2'

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:33] [1;32mDTMF[0m[4341]: [1;37mchannel.c[0m:[1;37m4151[0m [1;37m__ast_read[0m: DTMF begin '1' received on SIP/Carrier-X_OutBound-00048bb0
[Nov 2 23:05:33] [1;32mDTMF[0m[4341]: [1;37mchannel.c[0m:[1;37m4155[0m [1;37m__ast_read[0m: DTMF begin ignored '1' on SIP/Carrier-X_OutBound-00048bb0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:33] [1;32mDTMF[0m[4341]: [1;37mchannel.c[0m:[1;37m4066[0m [1;37m__ast_read[0m: DTMF end '1' received on SIP/Carrier-X_OutBound-00048bb0, duration 280 ms

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:33] [1;32mDTMF[0m[4341]: [1;37mchannel.c[0m:[1;37m4135[0m [1;37m__ast_read[0m: DTMF end passthrough '1' on SIP/Carrier-X_OutBound-00048bb0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:33] -- Playing 'request_has_been_processed' (escape_digits=) (sample_offset 0)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:35] [1;33mNOTICE[0m[2060]: [1;37mchan_sip.c[0m:[1;37m13719[0m [1;37msip_reregister[0m: -- Re-registration for 1111111111@sip.carrier.com

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:35] REGISTER 11 headers, 0 lines

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:35] Reliably Transmitting (NAT) to *.*.*.*:5060:
REGISTER sip:sip.carrier.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK0f545b7c;rport
Max-Forwards: 70
From: <sip:1111111111@sip.carrier.com>;tag=as60efaf22
To: <sip:1111111111@sip.carrier.com>
Call-ID: 7d378d9b0d3cf91726d8cafb4c292776@*.*.*.*
CSeq: 120 REGISTER
User-Agent: Asterisk PBX 1.8.23.0-vici
Authorization: Digest username="1111111111", realm="asterisk", algorithm=MD5, uri="sip:sip.carrier.com", nonce="61af7b1a", response="6b9d57853a4a16905bd49ce4448869ea"
Expires: 120
Contact: <sip:165378zpwp5apawu4qnf@*.*.*.*:5060>
Content-Length: 0

---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:35]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK0f545b7c;received=*.*.*.*;rport=5060
From: <sip:1111111111@sip.carrier.com>;tag=as60efaf22
To: <sip:1111111111@sip.carrier.com>;tag=as074ab8d9
Call-ID: 7d378d9b0d3cf91726d8cafb4c292776@*.*.*.*
CSeq: 120 REGISTER
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17af546c"
Content-Length: 0

<------------->
[Nov 2 23:05:35] --- (11 headers 0 lines) ---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:35] Responding to challenge, registration to domain/host name sip.carrier.com

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:35] REGISTER 11 headers, 0 lines

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:35] Reliably Transmitting (NAT) to *.*.*.*:5060:
REGISTER sip:sip.carrier.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK06687b4d;rport
Max-Forwards: 70
From: <sip:1111111111@sip.carrier.com>;tag=as7c235e83
To: <sip:1111111111@sip.carrier.com>
Call-ID: 7d378d9b0d3cf91726d8cafb4c292776@*.*.*.*
CSeq: 121 REGISTER
User-Agent: Asterisk PBX 1.8.23.0-vici
Authorization: Digest username="1111111111", realm="asterisk", algorithm=MD5, uri="sip:sip.carrier.com", nonce="17af546c", response="5cbc9f9ef5f3f4465b7c947108197af5"
Expires: 120
Contact: <sip:165378zpwp5apawu4qnf@*.*.*.*:5060>
Content-Length: 0

---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:35]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK06687b4d;received=*.*.*.*;rport=5060
From: <sip:1111111111@sip.carrier.com>;tag=as7c235e83
To: <sip:1111111111@sip.carrier.com>;tag=as074ab8d9
Call-ID: 7d378d9b0d3cf91726d8cafb4c292776@*.*.*.*
CSeq: 121 REGISTER
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: <sip:165378zpwp5apawu4qnf@*.*.*.*:5060>;expires=120
Date: Sun, 03 Nov 2013 06:05:35 GMT
Content-Length: 0

<------------->
[Nov 2 23:05:35] --- (13 headers 0 lines) ---
[Nov 2 23:05:35] Scheduling destruction of SIP dialog '7d378d9b0d3cf91726d8cafb4c292776@*.*.*.*' in 32000 ms (Method: REGISTER)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:35] [1;33mNOTICE[0m[2060]: [1;37mchan_sip.c[0m:[1;37m21597[0m [1;37mhandle_response_register[0m: Outbound Registration: Expiry for sip.carrier.com is 120 sec (Scheduling reregistration in 105 s)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20131102-230511_9512944547_VDAD_154916)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] -- <SIP/Carrier-X_OutBound-00048bb0>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] -- Executing [067*052*125*022*719092844004@default:1] [1;36mGoto[0m("[1;35mSIP/Carrier-X_OutBound-00048bb0[0m", "[1;35mdefault,719092844004,1[0m") in new stack
[Nov 2 23:05:37] -- Goto (default,719092844004,1)
[Nov 2 23:05:37] -- Executing [719092844004@default:1] [1;36mAGI[0m("[1;35mSIP/Carrier-X_OutBound-00048bb0[0m", "[1;35magi://127.0.0.1:4577/call_log[0m") in new stack

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=1001))

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] -- <SIP/Carrier-X_OutBound-00048bb0>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] -- Executing [719092844004@default:2] [1;36mDial[0m("[1;35mSIP/Carrier-X_OutBound-00048bb0[0m", "[1;35mSIP/19092844004@Carrier-X_OutBound,,tTor[0m") in new stack

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] == Using SIP RTP CoS mark 5

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Audio is at 17910

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Adding codec 0x4 (ulaw) to SDP

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Adding codec 0x2 (gsm) to SDP

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Adding non-codec 0x1 (telephone-event) to SDP

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Reliably Transmitting (NAT) to *.*.*.*:5060:
INVITE sip:19092844004@*.*.*.*:5060 SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK65ec598a;rport
Max-Forwards: 70
From: "V1022305030000154916" <sip:9512944547@*.*.*.*>;tag=as23a7af1e
To: <sip:19092844004@*.*.*.*:5060>
Contact: <sip:9512944547@*.*.*.*:5060>
Call-ID: 3d2657e816ec1dce570ebef71dc72c9d@*.*.*.*:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0-vici
Date: Sun, 03 Nov 2013 06:05:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "V1022305030000154916" <sip:9512944547@*.*.*.*>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1670077488 1670077488 IN IP4 *.*.*.*
s=Asterisk PBX 1.8.23.0-vici
c=IN IP4 *.*.*.*
t=0 0
m=audio 17910 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] -- Called SIP/19092844004@Carrier-X_OutBound

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK65ec598a;rport=5060
From: "V1022305030000154916" <sip:9512944547@*.*.*.*>;tag=as23a7af1e
To: <sip:19092844004@*.*.*.*:5060>
Call-ID: 3d2657e816ec1dce570ebef71dc72c9d@*.*.*.*:5060
CSeq: 102 INVITE
Server: Carrier-X Partner/1.1
Content-Length: 0

<------------->

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] --- (8 headers 0 lines) ---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK65ec598a;rport=5060
Record-Route: <sip:*.*.*.*:5060;lr>
From: "V1022305030000154916" <sip:9512944547@*.*.*.*>;tag=as23a7af1e
To: <sip:19092844004@*.*.*.*:5060>;tag=xtg-45462-18446744073675222112
Contact: <sip:19092844004@*.*.*.*:5060>
Call-ID: 3d2657e816ec1dce570ebef71dc72c9d@*.*.*.*:5060
CSeq: 102 INVITE
Server: Carrier-X Carrier/1.0
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 2346 2346 IN IP4 *.*.*.*
s=session
c=IN IP4 *.*.*.*
t=0 0
m=audio 15960 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] --- (11 headers 11 lines) ---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Found RTP audio format 0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Found RTP audio format 101

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Found audio description format PCMU for ID 0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Found audio description format telephone-event for ID 101

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Peer audio RTP is at port *.*.*.*:15960

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] list_route: hop: <sip:*.*.*.*:5060;lr>

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] set_destination: Parsing <sip:*.*.*.*:5060;lr> for address/port to send to

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] set_destination: set destination to *.*.*.*:5060

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] Transmitting (NAT) to *.*.*.*:5060:
ACK sip:19092844004@*.*.*.*:5060 SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK7424974f;rport
Route: <sip:*.*.*.*:5060;lr>
Max-Forwards: 70
From: "V1022305030000154916" <sip:9512944547@*.*.*.*>;tag=as23a7af1e
To: <sip:19092844004@*.*.*.*:5060>;tag=xtg-45462-18446744073675222112
Contact: <sip:9512944547@*.*.*.*:5060>
Call-ID: 3d2657e816ec1dce570ebef71dc72c9d@*.*.*.*:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0-vici
Content-Length: 0

---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:37] -- SIP/Carrier-X_OutBound-00048bb1 answered SIP/Carrier-X_OutBound-00048bb0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:50] Reliably Transmitting (NAT) to *.*.*.*:5060:
OPTIONS sip:sip.carrier.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK03a88560;rport
Max-Forwards: 70
From: "asterisk" <sip:1111111111@*.*.*.*>;tag=as5b129e8a
To: <sip:sip.carrier.com>
Contact: <sip:1111111111@*.*.*.*:5060>
Call-ID: 3e616ac50f5a47676b0da2be21d94ef5@*.*.*.*:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-vici
Date: Sun, 03 Nov 2013 06:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:50]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK03a88560;received=*.*.*.*;rport=5060
From: "asterisk" <sip:1111111111@*.*.*.*>;tag=as5b129e8a
To: <sip:sip.carrier.com>;tag=as3c13cbd6
Call-ID: 3e616ac50f5a47676b0da2be21d94ef5@*.*.*.*:5060
CSeq: 102 OPTIONS
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*.*.*.*:5060>
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 2 23:05:50] --- (12 headers 0 lines) ---
[Nov 2 23:05:50] Really destroying SIP dialog '3e616ac50f5a47676b0da2be21d94ef5@*.*.*.*:5060' Method: OPTIONS

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:50] Reliably Transmitting (NAT) to *.*.*.*:5060:
OPTIONS sip:sip.carrier.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK135b4c23;rport
Max-Forwards: 70
From: "asterisk" <sip:1111111111@*.*.*.*>;tag=as7b19decf
To: <sip:sip.carrier.com>
Contact: <sip:1111111111@*.*.*.*:5060>
Call-ID: 782ab1891751c17503a9c66b72a75298@*.*.*.*:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-vici
Date: Sun, 03 Nov 2013 06:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:50]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK135b4c23;received=*.*.*.*;rport=5060
From: "asterisk" <sip:1111111111@*.*.*.*>;tag=as7b19decf
To: <sip:sip.carrier.com>;tag=as45ca9e28
Call-ID: 782ab1891751c17503a9c66b72a75298@*.*.*.*:5060
CSeq: 102 OPTIONS
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*.*.*.*:5060>
Accept: application/sdp
Content-Length: 0

<------------->

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:50] --- (12 headers 0 lines) ---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:50] Really destroying SIP dialog '782ab1891751c17503a9c66b72a75298@*.*.*.*:5060' Method: OPTIONS

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:50] Reliably Transmitting (NAT) to *.*.*.*:5060:
OPTIONS sip:*.*.*.* SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK6e85a942;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@*.*.*.*>;tag=as655ff32b
To: <sip:*.*.*.*>
Contact: <sip:asterisk@*.*.*.*:5060>
Call-ID: 6232d597431de7e3086fb0b9095f3a04@*.*.*.*:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-vici
Date: Sun, 03 Nov 2013 06:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:50]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK6e85a942;rport=5060
From: "asterisk" <sip:asterisk@*.*.*.*>;tag=as655ff32b
To: <sip:*.*.*.*>;tag=Carrier-X-46912864402032
Call-ID: 6232d597431de7e3086fb0b9095f3a04@*.*.*.*:5060
CSeq: 102 OPTIONS
Server: Carrier-X Carrier/1.0
Content-Length: 0

<------------->

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:50] --- (8 headers 0 lines) ---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:05:50] Really destroying SIP dialog '6232d597431de7e3086fb0b9095f3a04@*.*.*.*:5060' Method: OPTIONS

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:02] == Manager 'sendcron' logged on from 127.0.0.1

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:02] == Manager 'sendcron' logged off from 127.0.0.1

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:02] == Manager 'sendcron' logged on from 127.0.0.1

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:03] == Manager 'sendcron' logged off from 127.0.0.1

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04]
<--- SIP read from UDP:*.*.*.*:5060 --->
BYE sip:9092844004@*.*.*.*:5060 SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK-Carrier-X-NbLTOoHKM8.0
Via: SIP/2.0/UDP 38.102.250.158:5060;branch=z9hG4bK-xcarrier-7362011296513465
From: <sip:19512944547@*.*.*.*:5060>;tag=xtg-2355719-578489133
To: "V1022305030000154916" <sip:9092844004@*.*.*.*>;tag=as2d8112de
Call-ID: 0ee64ddb79271ab12801e9b015946999@*.*.*.*:5060
CSeq: 201 BYE
Max-Forwards: 69
User-Agent: Carrier-X Carrier/1.0
Content-Length: 0

<------------->

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] --- (10 headers 0 lines) ---
[Nov 2 23:06:04] Sending to *.*.*.*:5060 (NAT)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] Scheduling destruction of SIP dialog '0ee64ddb79271ab12801e9b015946999@*.*.*.*:5060' in 6400 ms (Method: BYE)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04]
<--- Transmitting (NAT) to *.*.*.*:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK-Carrier-X-NbLTOoHKM8.0;received=*.*.*.*;rport=5060
Via: SIP/2.0/UDP 38.102.250.158:5060;branch=z9hG4bK-xcarrier-7362011296513465
From: <sip:19512944547@*.*.*.*:5060>;tag=xtg-2355719-578489133
To: "V1022305030000154916" <sip:9092844004@*.*.*.*>;tag=as2d8112de
Call-ID: 0ee64ddb79271ab12801e9b015946999@*.*.*.*:5060
CSeq: 201 BYE
Server: Asterisk PBX 1.8.23.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] -- Executing [h@default:1] [1;36mAGI[0m("[1;35mSIP/Carrier-X_OutBound-00048bb0[0m", "[1;35magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----52-----27[0m") in new stack

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] -- <SIP/Carrier-X_OutBound-00048bb0>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -52-----27 completed, returning 0

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] Scheduling destruction of SIP dialog '3d2657e816ec1dce570ebef71dc72c9d@*.*.*.*:5060' in 6400 ms (Method: INVITE)

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] set_destination: Parsing <sip:*.*.*.*:5060;lr> for address/port to send to

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] set_destination: set destination to *.*.*.*:5060

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] Reliably Transmitting (NAT) to *.*.*.*:5060:
BYE sip:19092844004@*.*.*.*:5060 SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK7a9eb516;rport
Route: <sip:*.*.*.*:5060;lr>
Max-Forwards: 70
From: "V1022305030000154916" <sip:9512944547@*.*.*.*>;tag=as23a7af1e
To: <sip:19092844004@*.*.*.*:5060>;tag=xtg-45462-18446744073675222112
Call-ID: 3d2657e816ec1dce570ebef71dc72c9d@*.*.*.*:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.23.0-vici
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] == Spawn extension (default, 719092844004, 2) exited non-zero on 'SIP/Carrier-X_OutBound-00048bb0'

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04]
<--- SIP read from UDP:*.*.*.*:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP *.*.*.*:5060;branch=z9hG4bK7a9eb516;rport=5060
From: "V1022305030000154916" <sip:9512944547@*.*.*.*>;tag=as23a7af1e
To: <sip:19092844004@*.*.*.*:5060>;tag=xtg-45462-18446744073675222112
Call-ID: 3d2657e816ec1dce570ebef71dc72c9d@*.*.*.*:5060
CSeq: 103 BYE
Server: Carrier-X Carrier/1.0
Content-Length: 0

<------------->

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] --- (8 headers 0 lines) ---

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:04] Really destroying SIP dialog '3d2657e816ec1dce570ebef71dc72c9d@*.*.*.*:5060' Method: INVITE

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:07] == Manager 'sendcron' logged on from 127.0.0.1

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:07] == Manager 'sendcron' logged off from 127.0.0.1

[Klinux-pxvh*CLI>
[0K[Nov 2 23:06:07] Really destroying SIP dialog '7d378d9b0d3cf91726d8cafb4c292776@*.*.*.*' Method: REGISTER

[Klinux-pxvh*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

by **williamconley** » Sun Nov 03, 2013 8:38 pm

Nope. Agi debug is similar to SIP debug, but it ouputs agi data instead of sip data. You can also look in /var/log/astguiclient.

In both cases, you would do well to activate debugging for the server (to the console for agi debug or to FILE for /var/log/astguiclient, or both if you want to be able to use either/both).

This will show the logic being used to make the call and allow you to determine where the phone number goes awry.

Based on your description, this is merely a survey campaign with a remote agent. The remote agent should see the callerid of the client if the call to the remote agent does not use the Vicidial dialplan. It's probably a good idea for the agent to be "on-hook". Is the remote agent an internal number or a number you must reach through a carrier?

by **Nefariousparity** » Mon Nov 04, 2013 2:33 am

Yes, just a remote agent calling a number doing a transfer. I will try with on hook to see what happens.

Here is AGI output.

tail -f agiout.2013-11-03
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi||UPDATE vicidial_list set status='PU' where lead_id='154916' and status NOT IN('CBHOLD','CALLBK','CBHOLD','QCFAIL');|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|-- VDAD vicidial_list PU update: |1|1383550083.892676|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|DAILY STATS UPDATE 1001|1|UPDATE vicidial_daily_max_stats SET update_time=NOW(),max_outbound='1' where campaign_id='1001' and stats_type='CAMPAIGN' and stats_flag='OPEN';|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|-- SURVEY RECORDING STARTED : |1|20131103-232814_9512944547_VDAD_154916|INSERT INTO recording_log (channel,server_ip,extension,start_time,start_epoch,length_in_sec,filename,lead_id,user,location,vicidial_id) values('SIP/Carrier-X-00048bb5','*.*.*.*','9512944547','2013-11-03 23:28:14','1383550094','0','20131103-232814_9512944547_VDAD_154916','154916','VDAD','20131103-232814_9512944547_VDAD_154916','1383550083.892676');|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|Preprocess time: |0.028 (1383550094.807800 - 1383550094.780195)|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi||INSERT INTO vicidial_log (uniqueid,lead_id,campaign_id,call_date,start_epoch,status,phone_code,phone_number,user,processed,alt_dial,list_id,comments) values('1383550083.892676','154916','1001','2013-11-03 23:28:14','1383550094','PU','','9512944547','VDAD','N','NONE','1212','0.028')|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|-- VDAD : |154916|154916|insert to vicidial_log: 1383550083.892676
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|-- VDAD VLE insert: |1|
|INSERT INTO vicidial_log_extended set uniqueid='1383550083.892676',server_ip='*.*.*.*',call_date='2013-11-03 23:28:14',lead_id = '154916',caller_code='V1032328030000154916',custom_call_id='';|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|-- VDAC posttime record: |1|20131103233014|V1032328030000154916|SURVEY|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|interrupt_digit |49| timeout |60|60000|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|digit || TotalDTMF |1|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi||UPDATE vicidial_list set status='PM' where lead_id = '154916';|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|-- VDAD vicidial_list OPT_CODE update: |1|154916|1|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|-- VDAC posttime record: |1|20131103232937|V1032328030000154916|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi||SELECT count(*) FROM vicidial_live_agents where campaign_id = '1001' and last_update_time > '20131103232809';|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi||CONCURRENT TRANSFERS AUTO SETTING: 2 (10)|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '154916' and campaign_id = '1001' and call_time < "2013-11-03 23:28:03";|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|-- VDAD get agent: |1|update of vla table: 1001|*.*.*.*
|UPDATE vicidial_live_agents set status='QUEUE',lead_id='154916',uniqueid='1383550083.892676', channel='SIP/Carrier-X-00048bb5', call_server_ip='*.*.*.*', callerid='V1032328030000154916',comments='AUTO' where status='READY' and lead_id<1 and ring_callerid='' and campaign_id='1001' and last_update_time > '20131103232809' order by last_state_change limit 1;|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi||SELECT conf_exten,user,extension,server_ip,ra_user FROM vicidial_live_agents where status IN('QUEUE','INCALL') and campaign_id='1001' and callerid='V1032328030000154916' and channel='SIP/Carrier-X-00048bb5' order by last_call_time limit 1;|
2013-11-03 23:28:14|agi-VDAD_ALL_outbound.agi|-- REMOTE EXTEN LOG : |1|CID changed: "V1032328030000154916" <9512944547>
|INSERT INTO vicidial_remote_agent_log set callerid='V1032328030000154916',uniqueid='1383550083.892676',ra_user='8080',user='8089',call_time=NOW(),extension='719092844004',lead_id='154916',phone_number='9512944547',campaign_id='1001',processed='N';|
2013-11-03 23:28:41|agi-VDAD_ALL_outbound.agi|-- REMOTE RECORDING STARTED : |1|20131103-232814_9512944547_VDAD_154916|INSERT INTO recording_log (channel,server_ip,extension,start_time,start_epoch,length_in_sec,filename,lead_id,user,location,vicidial_id) values('SIP/Carrier-X-00048bb5','*.*.*.*','9512944547','2013-11-03 23:28:41','1383550121','0','20131103-232814_9512944547_VDAD_154916','154916','8089','20131103-232814_9512944547_VDAD_154916','1383550083.892676');|
2013-11-03 23:28:41|agi-VDAD_ALL_outbound.agi|-- VDAD XFER REMOTE: |1|update of vac table: V1032328030000154916
|UPDATE vicidial_auto_calls set status='XFER', stage='XFER-0', extension='067*052*125*022*719092844004' where callerid='V1032328030000154916';|
2013-11-03 23:28:41|agi-VDAD_ALL_outbound.agi|exiting the VDAD app, transferring call to 067*052*125*022*719092844004
2013-11-03 23:28:41|agi-VDAD_ALL_outbound.agi|XXXXXXXXXX VDAD transferred: start|stop 2013-11-03 23:28:14|2013-11-03 23:28:41

by **williamconley** » Mon Nov 04, 2013 11:15 am

Since I don't know which number is which ... perhaps you could shed some light and determine where your Client number gets lost?

by **Nefariousparity** » Mon Nov 04, 2013 12:14 pm

I think right here.

"agi-VDAD_ALL_outbound.agi|-- REMOTE EXTEN LOG : |1|CID changed: "V1032328030000154916" <9512944547>
|INSERT INTO vicidial_remote_agent_log set "

Survey calls my test number (951 294 4547), I press 1, then transfers to the other server calling the predictive servers inbound line at 9092844004

by **Nefariousparity** » Mon Nov 04, 2013 2:23 pm

***EDIT***

Just did a test with Survey on a different carrier, and then calling number on other carrier. And it worked flawlessly.

by **williamconley** » Mon Nov 04, 2013 11:02 pm

Interesting concept. Is there a difference in the dialplan entry of the two carriers in admin->carriers?

by **Nefariousparity** » Tue Nov 05, 2013 1:35 pm

No, that is what is blowing my mind. My final test is to have the survey server go through A2b, but out one carrier and back in on another carrier to the predictive server.

by **williamconley** » Tue Nov 05, 2013 9:02 pm

Or perhaps just a different carrier. We've had issues with specific carriers in the past. Some temporary, some permanent ...

by **Nefariousparity** » Tue Nov 26, 2013 2:39 pm

All of this and the problem was with the carrier. Uuuuuuuggggg!Thanks for all the help once again WIlliam!

by **williamconley** » Sat Nov 30, 2013 9:13 pm

And yet another vote for "Always check/blame the carrier first ... because it's so simple to check!" LOL

Happy to help. Now stay and help a newbie to pay me back!! 8-)

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