It's the dreaded "invalid extension"

[rgc]

Hi all,

This is my first time posting here so I hope I've included all the info you need. Any help is much appreciated because the other similar forum posts didn't help.

System:
I downloaded the demo version of vicidial a few weeks ago (perhaps 3) as an ISO and installed it straight into a VMWare (v5.0.0 build-812388) virtualization. My machine is a Core 2 duo 2.5GHz with 4GB RAM and SSD hard drive. The Asterisk CLI tells me that this is Asterisk 1.2.30.2 and Ubuntu is running 4 year old Gnome v2.22.2 .

At present I'm just trying to make outbound calls, which has been fine when I dial directly from XLite. I am on page 20 (item 34) in the manual and when I attempt an outbound call I receive a "Sorry that is not a valid extension."
The number I am dialing is a regular 13 digit cell phone number with DialCode: 1. My phone is named "phoneextension" with extension 666. I have attempted dialing manually and with a lead list. From my point of view, its an issue with the dial plan so I'll post that first, followed by carrier settings and .confs.

In the "exten" entries below I've tried firstly applying the default settings in the manual, secondly with a global string, and finally with no leading 9s in front of the number. Although I restart the whole Asterisk server each time (is this necessary?), all attempts result in the same thing. I'm losing hair here... thanks in advance for keeping me young.

P.S. Excuse the randomly scattered white space... just trying to defeat your incessant "spammy" protection. Turns out it doesn't like .n3t


Global String:

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TRUNKVIT=username:passw d @ in bound. ********.n3t: 5060

Dialplan Entry:

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exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1: 4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(TRUNKVIT/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup


These are my Carrier Settings:

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register => username:pass wd @ inbo und. ********.n3t:5060

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[vitelity]
disallow=all
allow=ulaw
type=friend
username=username    <== I used the correct uname and passwd here
secret=passwd
host=inbound. ******** .n3t
dtmfmode=auto
context=trunkinbound


Sip Show Peers:

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v it e l i t y/username         66.241.99.27         N      5060     Unmonitored

CLI output:

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=========================================================================
Connected to Asterisk 1.2.30.2 currently running on rob-desktop (pid = 6812)
Verbosity is at least 21
    -- Remote UNIX connection
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
       > Channel SIP/phoneextension-08214810 was answered.
    -- Executing MeetMe("SIP/phoneextension-08214810", "8600051|F") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
    -- Playing 'conf-onlyperson' (language 'en')
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing MeetMe("Local/8600051@default-94d4,2", "8600051|F") in new stack
       > Channel Local/8600051@default-94d4,1 was answered.
  == Starting Local/8600051@default-94d4,1 at default,**********,1 failed so falling back to exten 's'
  == Starting Local/8600051@default-94d4,1 at default,s,1 still failed so falling back to context 'default'
    -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-94d4,1
    -- Executing Playback("Local/8600051@default-94d4,1", "invalid") in new stack
    -- Playing 'invalid' (language 'en')
Nov  6 22:09:13 WARNING[2262]: file.c:1045 ast_waitstream: Unexpected control subclass '-1'
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Spawn extension (default, i, 1) exited non-zero on 'Local/8600051@default-94d4,1'
    -- Executing DeadAGI("Local/8600051@default-94d4,1", "agi://127.0.0. 1 :4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-94d4,2'
    -- Executing DeadAGI("Local/8600051@default-94d4,2", "agi://127.0.0. 1 :4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1


extensions.conf

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[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
TRUNK=Zap/g1                                    ; Trunk interface
TRUNKX=Zap/g2                                   ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569    ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569   ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.c0m   ; IAX trunk interface
SIPtrunk=SIP/1234:PASSWORD@sip.provider.n3t     ; SIP trunk
TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569   ; used for blind monitoring
TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing

#include extensions-vicidial.conf

[trunkinbound]
; agent dial-in:
exten => 2345,1,Answer          ; Answer the line
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup

; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})



[default]
include => vicidial-auto

; VICI-GROUP DIRECT SUPPORT LINE (VICIHELP[84244357])
exten => _84244XXX,1,Dial(IAX2/vicihelp/${EXTEN:5})

; Local agent alert extensions
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
; Local blind monitoring
exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)


;;;;;;;;;; BEGIN Voicemail and Prompts Section ;;;;;;;;;;;;;;;;;;;;;;;
; Give voicemail at extension 8500
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
[color=#FF0000]...etc[/color]


extensions-vicidial.conf

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KPIAF = IAX2/ViciBox:test@192.168.X.x:4569
TRUNKVIT=username:passwd@inbound.******.n3t:5060

[vicidial-auto]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG--------------------)

; Local Server: 127.0.0.1
exten => _127*000*000*001*.,1,Goto(default,${EXTEN:16},1)
; Remote Server VDAD extens: PIAF 192.168.2.10
exten => _192*168*XXX*xxx*.,1,Dial(${TRUNKPIAF}/${EXTEN:16},55,o)
; VICIDIAL Carrier: Vitelity - Vitelity
exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)^M
exten => _1NXXNXXXXXX,2,Dial(TRUNKVIT/${EXTEN:2},,tTor)^M
exten => _1NXXNXXXXXX,3,Hangup

...
exten => 666,1,Dial(SIP/phoneextension)
exten => 666,2,Voicemail,u666


[Kumba]

does dahdi_test run?

It doesn't look like you attempted to dial. It looks like you tried to login and got the error. That's usually a DAHDI timer error.

[rgc]

does dahdi_test run?

It doesn't look like you attempted to dial. It looks like you tried to login and got the error. That's usually a DAHDI timer error.

Thanks for the fast reply! The code below is a snippet that is generated everytime I "dial" while the MeetMe() is from my answering the internal call via XLite. The stars are the cell number I'm trying to reach. Hopefully this constitutes dialing.

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    -- Executing MeetMe("Local/8600051@default-94d4,2", "8600051|F") in new stack
       > Channel Local/8600051@default-94d4,1 was answered.
  == Starting Local/8600051@default-94d4,1 at default,**********,1 failed so falling back to exten 's'
  == Starting Local/8600051@default-94d4,1 at default,s,1 still failed so falling back to context 'default'
    -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-94d4,1
    -- Executing Playback("Local/8600051@default-94d4,1", "invalid") in new stack
    -- Playing 'invalid' (language 'en')

It will be interesting to see if dahdi_test runs okay so I'll give it a try as soon as I'm back and connected to the wired network.

[striker]

dialplan not updated or wrong

as per your dialplan entry

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1: 4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(TRUNKVIT/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup

as per your extensions-vicidial.conf
; VICIDIAL Carrier: Vitelity - Vitelity

exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)^M
exten => _1NXXNXXXXXX,2,Dial(TRUNKVIT/${EXTEN:2},,tTor)^M
exten => _1NXXNXXXXXX,3,Hangup

*** 9 is missing in the dialplan

[williamconley]

13 digit cell phone number

cell phones are not 13 digits. Your dialplan expects 10 digits (possibly with a 1 in front of it from a "dial code" entry ... and the campaign will add a 9 ... this totals 12 digits, and you only dialed 10 of them ...).

I note that the number itself is missing and that the "dial" command does not appear in your examples (which are very well organized ... cool ... but missing the initiation of the outbound call being made! LOL).

I also note that you have not been entirely descriptive about the action being taken in the call (ie: I entered 9195551212 in the manual dial field in the agent screen and pressed dial and this is what happened in the CLI).

I also note that you mention "the demo version" but did not give the name of the .iso in question (helpful for clarity ...). Here's a sample suggestion I drop on "Lots" of new users (has good info in it, pick through but read it all ...):

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600
_____________

And even more importantly: The "Demo" version is not meant to be "installed". It was designed as a "live CD" to be booted and run without installation (and is Very Old!). If you are going to Install, use Vicibox 4.0 full version and you'll have a much better experience and a much more recent product.

[rgc]

Thanks for the replies. Apologies for the newby questions.

Lets clear up questions about the system. I used the vicidial demo "vicibox-live-demo-1.2.0-i386.is0"; sorry for the omission and the misuse of "installed" I'd love to start using Vicibox 4.0 full version but sadly I'm not up to scratch in linux and ended up mysteriously breaking the installation several times before reverting to something that I could be sure would work, i.e. the demo. So far I've tried Ubuntu and CentOS(for the easy install of Asterisk), but the Vicibox 4.0 install went a awry. I failed to coaxing OpenSuse to install Asterisk. Any updated demo versions on the horizon?

Kumba, I'm having a bit of trouble locating dahdi_test in this version of the demo. Isn't it in /usr/sbin? This is probably another case for using the full version of Vicibox. Regardless of how we get the system up and running, I'd still like to be able to run dahdi_test for completeness.

striker, good catch! I actually took the screen shot for extensions-vicidial.c0nf after I made a single change to the dialplan, which was to remove the 9. All other code windows are consistent. I have tried both with and without the 9. I did not think it made much difference because vicidial should update extensions-vicidial.c0nf automatically no?

williamconley,

13 digit cell phone number

= I simply cannot count *sigh*. I indeed used a regular 10* digit number.
Despite reading the manual and voip.0rg several times, your post is the first one to make it clearer regarding what is happening with a dialed number during dial(). THANK YOU! If I understand things correctly, by starting with a lead that has DialCode 1 and Phone Number 9871234567, if I click dial it should pass that number across as 19871234567. Then the campaign adds a leading 9 which makes the global variable EXTEN contain 919871234567. This is used as a parameter in dial() along with the channel. Does this sound right? Also where does the channel get set?

By the way, to "Manual Dial" I began in the agent screen with "1" in the DialCode field, and a 10 digit cell number like 9871234567 in the Phone Number field and then pressed "Dial Now". I also tried 919871234567 in this field with the DialCode field blank, which actually produced a dial() event. See CLI code below. I also tried the "Dial Override" field with 19871234567 and 919871234567. The former produced no dial() event, while the latter produced the same CLI output as shown below. FYI the dialplan begins with a 9 in this case.

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=========================================================================
Connected to Asterisk 1.2.30.2 currently running on desktop (pid = 6698)
Verbosity is at least 38
    -- Remote UNIX connection
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing MeetMe("Local/8600051@default-3b30,2", "8600051|F") in new stack
       > Channel Local/8600051@default-3b30,1 was answered.
    -- Executing AGI("Local/8600051@default-3b30,1", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing Dial("Local/8600051@default-3b30,1", "Zap/g2/19871234567||To") in new stack
Nov  7 20:27:20 NOTICE[20771]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Hangup("Local/8600051@default-3b30,1", "") in new stack
  == Spawn extension (default, 912064996023, 3) exited non-zero on 'Local/8600051@default-3b30,1'
    -- Executing DeadAGI("Local/8600051@default-3b30,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL---------- completed, returning 0
  == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-3b30,2'
    -- Executing DeadAGI("Local/8600051@default-3b30,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1


Is Zap/g2 being set in the outbound calling group? Are there any clues in there or shall I get other output?

[rgc]

UPDATE! But not fixed yet

Managed to fix the 'Zap/g2' channel issue I think. Found that in extensions.conf there was a global variable TRUNKX referring to Zap/g2. I changed TRUNKX to point to my provider SIP/username:passwd@outbound.vitelity.n3t:5060

CLI output when a manual call is initiated by using DialOveride: 919871234567 and clicking "Dial Now." The new error is "is circuit-busy"

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-- Executing MeetMe("Local/8600051@default-a634,2", "8600051|F") in new stack
       > Channel Local/8600051@default-a634,1 was answered.
    -- Executing AGI("Local/8600051@default-a634,1", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing Dial("Local/8600051@default-a634,1", "SIP/username:passwd@outbound.vitelity.n3t:5060/19871234567||To") in new stack
    -- Called username:passwd@outbound.vitelity.n3t:5060/19871234567
    -- SIP/outbound.vitelity.n3t:5060/19871234567-0823fc10 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Hangup("Local/8600051@default-a634,1", "") in new stack
  == Spawn extension (default, 919871234567, 3) exited non-zero on 'Local/8600051@default-a634,1'
    -- Executing DeadAGI("Local/8600051@default-a634,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION---------- completed, returning 0
  == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-a634,2'
    -- Executing DeadAGI("Local/8600051@default-a634,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1


*CLI>sip show peer vitelity

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vitelityout/username      64.2.142.188         N      5060     OK (33 ms)
vitelity/username         64.2.142.187         N      5060     OK (33 ms)


[williamconley]

In which case your issue is that you are modifying this vicidial system as if it were a plain-vanilla asterisk system ... modifying conf files directly, etc.

You should put all your conf files back where they were (unedited) and create your carrier under Admin->Carriers according to the standards and you'll be fine.

There are lots of samples online, including (i'm pretty sure) specific samples for Vitelity.

The Global Variable (defined in the globals string field) should be used to "point" to the sip context created in the account entry. The globals entry is then used in the dialplan instead of the actual sip account entry (no user or pass required, as those are stored in the sip account and used automatically with this method). If you created your carrier in admin->carriers, all the ".conf files" are edited automatically by vicidial perl scripts and maintained for you.

Also I note you still have not posted your system specs Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

Have you read the Vicidial Manager's Manual?

[rgc]

Thank you William, I have updated my sig to reflect my setup and have indeed read the manual. Thanks also for the info about the globals string. This kind of information is missing from the manual.

I might start with a fresh install of the live cd demo. I have the following sample settings from vitelity, which have not worked directly (as their disclaimer suggested). It wants them in sip.conf and extensions.conf but it is not clear for a beginner like myself where to put these settings in vicidial. I would think it all goes in Admin->Carrier so fingers crossed.

Add the following at the bottom of your sip.conf

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[inbound]
type=friend
dtmfmode=auto
host=inbound.vitelity.n3t
context=inbound
username=*
secret=*
allow=all
insecure=port,invite
nat=yes

[outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.n3t
context=outbound
username=*
fromuser=*
trustrpid=yes
sendrpid=yes
secret=*
allow=all
nat=yes



Please add the following configuration to your /etc/asterisk/extensions.conf
Add the following to the bottom of your extensions.conf

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[outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/$\{EXTEN}@outbound)
exten => _011.,1,Dial(SIP/$\{EXTEN}@outbound)

; e911 must be enabled. see DIDs > NPANXXNXXX > Action > e911
exten => _911,1,Dial(SIP/911@outbound)

[inbound]

[williamconley]

the samples sent by vitelity for sip.conf go in the account entry, except:

context=inbound
becomes
context=trunkinbound

and
context=outbound
becomes
context=trunkinbound

there is NO reason to have a "context=outbound" as this value is NEVER used for an outbound call. Any call received through any sip connection should be routed to trunkinbound, and that is the only use of this value.

The extensions.conf entries can be merely discarded and you must use a standard vicidial extensions.conf configuration. The reason for this is many faceted: First is the requirement for the agi line which is in the stock samples. Without this line the dialer simply won't work. Another reason is to ensure you use the "o" switch in the dial command. Necessary for many functions to operate properly. And the hangup line must also exist to ensure things go well ...

This standard setup should also include a "variable" that links to your "Globals String" defined variable which can allow all your "dialplan entries" to be virtually identical to one another, except for the "Dial Prefix". In some of the examples the exten entry will be "_91NXXNXXXXXX" in which the "9" is the Dial Prefix. This allows creating several carriers that only differ in the "9" becoming (for instance) an "8" or a "7", and this is how you will choose your carrier in each Campaign (the "9" goes in the "Dial Prefix" field in the campaign). There are ordinarily only two differences between each standard dialplan entry: the Dial Prefix and the Global String variable.

Check the available samples and you'll see what I mean. Beware the :2 or :1 inside a variable. This represents "stripping off" a digit at the front of the exten. So "913522690000" becomes "3522690000" if the variable has :2 at the end. Strips off the first two digits. And what happened to the "_", you ask? LOL That is only there to allow the extension to be a "pattern" instead of an actual number. So that "NXX" can be any three digit number (the N can be 2-9 and the "X" can be 0-9).

you make no entries in sip.conf or extensions.conf. Sip.conf "stuff" goes in "Account Entry" (which Vicidial then places in "sip-vicidial.conf" ... then voip carrier provided "Extensions.conf" information is discarded in favor of the Vicidial standard version of what goes in "Dialplan Entry". Vicidial then places this in "extensions-vicidial.conf" for you after you save your carrier settings (at the one minute mark, which is when the script that saves those files runs).

All of these samples have been posted many times, and there are some "interesting" examples in your Admin->Carriers menu. But be careful about the dialplan entry (999 is not viable for an operational dialplan entry, needs to be changed ...) and the sample globals string will need to be rewritten to actually connect to the Account Entry you set up.

The Globals String should be something like:
DIAL9TRUNK = SIP/carrier
and is used like:
exten=>_91NXXNXXXXXX,n,Dial(${DIAL9TRUNK}/${EXTEN:1},,To)

where "carrier" is replaced with whatever you put inside the [brackets] in your account entry. They must match because you are creating a "variable" that will be replaced with that value. When the dial command is executed, it will replace "DIAL9TRUNK" with "SIP/carrier" inside the dial command ... so if that carrier doesn't exist the dial command will fail. Must be exact (case sensitive ...). And includes "SIP/" only if it is a SIP carrier (if it's IAX2 then it would be "IAX2/carrier" ... right?). In the end, when the dial command is executed it will end up like: dial (SIP/carrier/13522690000) even though the number dialed was "913522690000" because the :1 removed that first digit. When you watch your asterisk command line, you can even see this process happen as one line will show 913522690000 executing and a couple lines later, you'll see dialed SIP/carrier/13522690000 ... so you can actually watch it work. And bring us any results that seem wrong.

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[vitel-inbound]
type=friend
dtmfmode=auto
host=zzzzzzzzz
context=trunkinbound
username=xxxxx
secret=yyyyy
allow=all
insecure=very
canreinvite=no


[vitel-outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
username=xxxxx
fromuser=xxxxx
secret=yyyyy
trustrpid=yes
sendrpid=yes
allow=all
canreinvite=no

Code: Select all

DIAL9TRUNK = SIP/vitel-outbound

Code: Select all

exten=>_91NXXNXXXXXX,1,Set(CALLERID(num)=9999999999)
exten=>_91NXXNXXXXXX,n,AGI(agi://127.0.0.1:4577/call_log)
exten=>_91NXXNXXXXXX,n,Dial(${DIAL9TRUNK}/${EXTEN:1},,To)
exten=>_91NXXNXXXXXX,n,Hangup


[rgc]

This is the most I've learned about Vicidial and dialplans than from any other source. Is there such thing as a sticky-post on the forum because all that information should be available to new users, if not in the manual itself!

I'll try it tomorrow

[williamconley]

Actually, I've done that quite a few times. I thought it was stickified once. I have not, however, taken the time to put it on a wiki yet. Maybe I should. LOL

[williamconley]

http://www.viciwiki.com/index.php/Add_a_SIP_Carrier

[rgc]

Thank you William, everything is working flawlessly and thanks too for adding it to the wiki. For the benefit of anyone else, I'll share one other stumbling block along the road for me. These are just basics but the manual won't help you troubleshoot them easily.

The first time I followed your instructions William, I was able to dial out but could not hang up on people. It turned out that,
1) If you don't have leads in the hopper and also try to dial manually, the hangup button can be disabled and you can't hangup unless you use the link in 2) below. Just make sure you have leads in the hopper by adding a campaign and lead list as per the manual. If your lead list is short like mine was, make sure you reset the leads so that you can dial them again: Lists -> List ID -> Reset Lead-Called-Status for this list.
2) You can always hangup on people by using a link at the bottom of the agent screen that shows the conference participants (hidden by default I believe).

I also ran into issues during testing because, if your campaign is set to dial new leads only ( like mine was) and ignore the leads you have called already and set the disposition for, then after you've gone through a lead list once you will not be able to call any more leads from that list. The way I fixed this was to go to Campaigns -> Add a Dial Status, and allow the campaign to dial No Answer (N), Disconnected Number (DC) and a few other Dial Statuses.

Many thanks to all who helped out.

[williamconley]

You would do well to read through the manual. It has a fair amount of exercises for new users.

Excellent post-back by the way.

Now stay and help out! You're obviously good at it ...