Question Concerning Setup.

[Kerry C]

I have been given the project of setting up a predictive dialer.
I read all types of things on this forum and asterisk forum, and understand the hardware i need, we have also acquired a full t1.

My requirements are for 8 agents using a predictive dialer, from a lead list.

I will be using 1 server ( vicidial / asterisk) with hardware specifications of:

2 3.0 dual xeons, 4gb ram, 2 250gb raid 1 hdd, and a 10 /100/1000 nic.

Do i need anything else hardware wise, I am still uncertain on what a t1 card does, or how many sip trunks i need, I would like the predictive to dial at least 8 calls at a time for the lead list, while the agents will be using x lite and a web interface to take the calls the predictive finds. also I would like a sip trunk provider that utilizes a monthly base fee for unlimited out bound / inbound long distance in the use. so far I have found: simplesignal ( dot) com

but they want 30$ a month for unlimited in and out in US per trunk.

any advice or help to my questions would be great!

Kerry

[williamconley]

you do not need to go quite that far. there are several (many in fact) providers out there that charge only per minute for minutes actually used. i even have one that can go well below 1 cent per minute and they have a "startup" company that will start you at 1.5 cents per minute, along with several other companies that range from 1.0 - 1.5 cents per minute. Inbound is the same cost, but also includes a monthly fee of $1 to $3 per DID.

Unless you are getting UNLIMITED MINUTES (with permission to use a dialer on it), do not get any "Monthly Charge SIP Channels".

I'm actually working with one of the companies that has the best price to become a sponsor of this project.

[Kerry C]

Well we use 100,000 minutes + per month, so a unlimited sip trunk would work for each dialer wouldn't it, and does vicidial require multiple trunks to outbound call on the predictive, aka while its dialing on the predictive and I want to choose 8 dial out level does it require multiple channels for that and if im using sip trunks it will not take up multiple channels will it? If i am wrong please educate me, I am trying to understand the best setup for this.

[williamconley]

Ordinarily the "unlimited outbound channel" is for ONE CALL. Also, the contract will ordinarily PROHIBIT "Dialers" of any sort. So for a dialer you will be relegated to pay per minute unless you violate a contract or get VERY lucky.

I have only found one provider of a PRI who would allow unlimited outbound, and he went out of business. I can't imagine why.

Providers like Vonage have unlimited ... but specifically prohibit dialers of any sort and reserve the right to "back-bill" you if they catch you using a dialer on your $24.95/mo unlimited line. Obviously they have a reason for that, they don't want to be ripped off any more than you do.

So Yes, an unlimitied trunk will work, but your providers documentation will explain the rules on usage and how many calls you can fit into the trunk. Generally: 1

This is why we all use providers who bill by the minute and will have a limit of anywhere from 10 to 10000 "channels" per account. Your dialer will be requesting a channel for each simultaneous call. When the provider is billing you for every active call by the minute, they LIKE IT when you use more ...

[Kerry C]

I need to make sure I understand this correctly:

I need 7 agents on calls, the predictive needs a sip trunk for the 4 dial out method ( auto dial) for each agetn, making it 4 x 7 agents = 28 + 7 sip[ trunks for the agents end making it 35. Is this calculation correct.

and also with a t1 is the 729 codec best, i cannot do this with a 711 codec at 90k up.

is this correct?

[williamconley]

You seem to be getting it. Your math is accurate.

If you assume 100 per call (10 calls per Meg) for ulaw, I can see why you may want to compress via gsm or g729. At 35 calls you'd need 3.5Meg to accommodate it, and the average T1 is 1.5Meg. So if your T1 is 1.5 Meg, you are quite correct you will NOT be transmitting 35 calls via ulaw through it.

So you purchae 40 g729 licenses from digium or download from wherever you can get it and now you can fit 35 calls in 1.5 Meg with no serious issues, but now your CPU is being hammered (by all that transcoding of calls). Make sure your CPU is not being overworked. Try to keep your utilization at or below 50% for cruising (one CPU= .5, two CPUs 1.0 ... etc).

That's just a very rought guideline.

For that call volume I've found that a 2.3Ghz Core2Quad is sufficient, but later when you expand you may need more. On the other hand, later when you need more you may also consider getting a second server and a second T1 ...

[Kerry C]

I have acquired 5 g729a codecs from the digium site and followed this readme as follows:

1) Download and execute the 'register' tool to generate a valid license.

1.1) Download the register utility to the root home directory of your
Asterisk server.

** Command-line example for 32-bit Linux:
# Log in as root or use the "su" command to assume root privileges.
cd /root
wget http://downloads.digium.com/pub/registe ... 2/register

1.2) Change the permissions of the /root/register file to r-x------. Change
the user and group ownership of the /root/register file to "root".

** Command-line example:
chmod 500 /root/register
chown root:root /root/register

1.3) Run the register tool and follow the interactive instructions.
Internet access is required from your Asterisk server in order to
register your G.729 key for licensed use. Outgoing network traffic to
TCP port 443 (SSL) must be allowed in order for the register utility to
successfully communicate with Digium's license server and complete the
registration process. The registration utility will prompt you for your
G.729 license key.

** Command-line example:
/root/register



******************

success - Kerry



2) Download and execute the 'benchg729' tool to select the optimum build.

There are various optimized versions of the codec binary available for
different commonly available CPU types, in both x86-32 and x86-64
architectures. To determine which build of the module performs best on
your system, the 'benchg729' utility will run a series of tests and
report which codec module will maximize encoding performance on your
system.

2.1) Download the benchg729 utility to the root home directory of your
Asterisk server.

** Command-line example for 32-bit Linux:
# Log in as root or use the "su" command to assume root privileges.
cd /root
wget http://downloads.digium.com/pub/telepho ... g729-1.0.5 -O benchg729

2.2) Change the permissions of the /root/benchg729 file to r-x------.
Change the user and group ownership of the /root/benchg729 file to
"root".

** Command-line example:
chmod 500 /root/benchg729
chown root:root /root/benchg729

2.3) Run the benchg729 tool and record the 'flavor' that it recommends that
you use on your system.
** Command-line example:
/root/benchg729

***************************************************

Success - by, this was done by inputting the new file structure based on version of link:

Ex:
wget http://downloads.digium.com/pub/telepho ... g729-1.0.5 -O benchg729

wget http://downloads.digium.com/pub/telepho ... g729-1.0.6 -O benchg729-1.0.6





3) Download and install the 'codec_g729' build for your platform.

There are different versions of the codec for various Asterisk releases;
there is a single version for all Asterisk 1.4.x releases, and there is
a version for each Asterisk 1.6.x point release (1.6.0, 1.6.1, etc.).
Note that these modules are *not* loadable in prior releases of Asterisk,
only the specific version they are designed to be used with. Please be sure
that you download the correct version of the codec for your Asterisk version.

In addition, there are frequently updated builds of the codec binary
posted, and each build has a 'version number'. This version number is
part of the filename, and is also included in the copyright/license message
that is displayed when the module is loaded into Asterisk. In this document
build number '3.0.1' has been used as an example, but when you read this
document the current build number may be different (higher).

*****************************************************
I basically used the new

3.1) Download the codec_g729 to the root home directory of your
Asterisk server, replacing 'pentium4m' in the example with the
recommended flavor.

** Command-line example:
wget http://downloads.digium.com/pub/telepho ... m4m.tar.gz

I used the 1.4 directory because i noticed on the vicidial site that the version of the vicidial download had this version of asterisk:

http://www.vicibox.com/server/vicibox-server.html

VICIbox Server includes:

* 2.0.5 astguiclient (VICIDIAL)
* 1.2.30.4 Asterisk
* 1.2.27 Zaptel
* 3.2.7.1 Wanpipe (for Sangoma TDM Cards)
* 1.2.8 Libpri
so the version folder i utilized was:
asterisk-1.4/

** was this incorrect?

3.2) Expand the codec_g729 archive and copy the codec_g729a.so file to the
/usr/lib/asterisk/modules directory.

** Command-line example:
tar xzvf codec_g729a-1.6.0_3.0.1-pentium4m.tar.gz
cp /root/codec_g729a-1.6.0_3.0.1-pentium4m/codec_g729a.so /usr/lib/asterisk/modules

*** did this utilizing the filename of the 1.4 folder, once I did this it just went to the next line.


3.3) Asterisk must be restarted in order to load your new G.729 licensed
channels. (See General Notes to use multiple licenses on one server.)

** Command-line example:
asterisk -rx "restart now"

3.4) Verify that the number of G.729 licensed channels available to Asterisk
matches the number of G.729 licensed channels that you purchased. This
can be verified by issuing "g729 show" in the Asterisk CLI. Take into
consideration any previous G.729 licensed channels that you may have
already had registered to your Asterisk server before verifying this
number.

** Command-line example:
asterisk -rvvv
*CLI> g729 show
0/0 encoders/decoders of 2 licensed channels are currently in use


****when I try to check it at the command prompt of root@vicibox:~#
using command asterisk -rvvv

it says this:

unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)


Can anyone help on making sure these codecs are installed correctly.


***********************************************************

[Op3r]

it seems your asterisk is not running.

type asterisk -g to start your asterisk

then show g729 at asterisk cli to check if your codec is installed.

[Kerry C]

same thing, I used the command asterisk -g
goes to new line, then I typed show g729 and got same response, any thoughts would be great

[Kerry C]

it says the show command is not installed

[Kerry C]

this was done at the prompt
root@vicibox:~#

typed asterisk -g

went back to

root@vicibox:~#

typed show g729

get this:

the program show is currently not installed, you can install it by typing: apt-get install nmh
-bash: show: command not found

[Op3r]

well of course you need to connect to it by

asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvv

then type show g729

[Kerry C]

tried asterisk -rvvvvvvvv

get

unable to connect to remote asterisk does(does /var/run/asterisk.ctl exist?)



its weird because now that i have installed the key for the licenses i am wondering how to backup the directory on a floppy when i try to mount the drive it gives a error, i was thinking of just reinstalling the os and trying agian but any help to keep from doing this would be great!

whats the right command to use to mount the floppy to backup the directory for the lecenses, or how can i get asterisk running again?

[Kerry C]

the read me says this:

===[ Backup Instructions ]=================================================

** It is VERY IMPORTANT to backup the license(s) generated by your system.

The /var/lib/asterisk/licenses directory contains the Host-ID specific
license files for your system. These license files are tied to the MAC
address of all the ethernet devices installed in your system. Creating a
backup of this directory will allow you to restore your G.729 license file
in case you need to reinstall your operating system. This will help prevent
you from needing to contact Digium to request authorization to increment
your G.729 key and from needing to purchase a new G.729 key if you exceed
the maximum number of G.729 key increments allowed.



my linux command syntax is sketchy its been about 7 yrs since i have messed with linux so i am kinda winging it. any help on the right backup syntax would be great.

just to backup to the floppy drive

[williamconley]

I used the 1.4 directory because i noticed on the vicidial site that the version of the vicidial download had this version of asterisk:

1.2.30.4 Asterisk

I don't suppose you noticed the .2.30. between the 1 and the 4?

if you loaded the 1.4 g729 codec for asterisk, it could have crashed your 1.2 asterisk.

as a test you can try:
asterisk -vvvc
this will start asterisk in console mode and likely tell you when and why it crashes. you can also check in /var/log/asterisk. you can also change /etc/asterisk/logger.conf to modify the logging parameters so it will put more information into the /var/log/asterisk folder in a file specified in the logger.conf file.

[Kerry C]

yup thats what happened, which codec license version should i use, and can this be fixed with reinstalling the os

[williamconley]

all you have to do is unload the codec you downloaded and download the correct one. (move it to another folder in case you upgrade to 1.4 later it will be useful)

as soon as you remove the codec from the folder (or change the system to ignore it) your asterisk will start working again.

[Kerry C]

sorry to be a well...dummy, but whats the syntax command to do this, it really helps, to get the answer, me searching the internet takes time at the moment, i tend to have 5 jobs here, and multitasking today and tomorrow will be a well, major thing.

[Kerry C]

I deleted the codec from the modules directory did asterisk - r and now its running again with this in the command area

Vicibox#CLI>

[williamconley]

considering the above example of how you copied it to use it ...

rm /usr/lib/asterisk/modules/codec_g729a.so

this will delete the file, but you have a copy of it according to the above directions.

if you are nervous, of course, you can try

mv /usr/lib/asterisk/modules/codec_g729a.so /usr/src/codec_g729a.so

this will move it so it won't be used, but not delete it.

[Kerry C]

the asterisk .conf configuration is one part, utilizing the administration area in vicidial and setting up a user is another, I have created a agent user Kerry, with the user number 7777 password 1234.

I have setup a campaign with leads loaded and active hopper leads, vicidial is giving me a phone login and password not active on the system error, in the agent login.

I have installed x-lite, when I fill out the sip info and use the servers ip, it gives me a registration 503 service unavailable.

also if i have the g729a codec how do you set up xlite for this also, it does not give the option to apply it in x-lite.

[Kerry C]

Now I have xlite running per voip on a user and pass specified, I have set up a agent in the admin on vicidial to the same username and pass, but per the manager manual it says put no phone user and login, so i did that, got the campaign made, with leads in hopper, but when i go to the username and pass area for the agent aka phone login and pass, I get sorry your phone login and pass is not active on this system, and xlite is connected.

[Kerry C]

set phone in admin area with username and pass, logged into agent area, success, xlite running, auto dial level to 0, started dialing states it is waiting for ring.

VOIP provider sent this log:


-- Executing MeetMe("Local/8600051@default-9eb3,2", "8600051|F") in new stack
> Channel Local/8600051@default-9eb3,1 was answered.
-- Executing AGI("Local/8600051@default-9eb3,1", "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-9eb3,1", "Zap/g2/13525361217||To") in new stack
May 28 13:56:13 NOTICE[13997]: 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-9eb3,1", "") in new stack
== Spawn extension (default, 913525361217, 3) exited non-zero on 'Local/8600051@default-9eb3,1'
-- Executing DeadAGI("Local/8600051@default-9eb3,1", "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 ... ---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-9eb3,2'
-- Executing DeadAGI("Local/8600051@default-9eb3,2", "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 ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
May 28 13:56:42 WARNING[14642]: channel.c:780 channel_find_locked: Avoided deadlock for '0x8186870', 9 retries!
== 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
-- Executing MeetMeAdmin("Local/55558600051@default-457d,2", "8600051|K") in new stack
-- Hungup 'Zap/pseudo-85553373'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-0816d680'
== Parsing '/etc/asterisk/meetme.conf': Found
-- Executing DeadAGI("SIP/cc100-0816d680", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
May 28 13:56:44 NOTICE[14082]: app_meetme.c:2210 admin_exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-457d,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-457d,2'
-- Executing DeadAGI("Local/55558600051@default-457d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("SIP/1000-b765b238", "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("SIP/1000-b765b238", "Zap/g2/16033815352||To") in new stack
May 28 13:57:00 NOTICE[14127]: 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("SIP/1000-b765b238", "") in new stack
== Spawn extension (default, 916033815352, 3) exited non-zero on 'SIP/1000-b765b238'
-- Executing DeadAGI("SIP/1000-b765b238", "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 ... ---------- completed, returning 0


*******************************************************

I sent him 2 numbers i dialed and he says it is not making it to the trunk.

Any help here?

[williamconley]

do you have, and have you read, the manager's manual?

please post the configuration for this box (hardware, software with versions) including any software other than vicidial running.

[Kerry C]

Hardware I am Using:
• Dell Poweredge 2650
• 2x Intel Xeon 2.4 GHZ Processors
• 2x 73GB 15K RPM Hard Drives with Trays
• DVD-ROM Drive
• 2GB Memory (2x 1GB)
• Power Cord
• Dual Power Supplies
• Floppy Drive
• CD-ROM
Switch:
Dell PowerConnect 3424 PoE Switch, Power over Ethernet
• 24 10/100BASE-T auto-sensing Fast Ethernet switching ports
• Additional 2 Copper GbE ports PLUS 2 optional Fiber GbE via SFP transceivers
• Integrated Copper GbE ports provide resilient stacking
• Auto-negotiation for speed, duplex mode and flow control
• Auto MDI/MDIX
• Port mirroring
• Broadcast storm control


* 2.0.5 astguiclient (VICIDIAL)
* 1.2.30.4 Asterisk
* 1.2.27 Zaptel
* 3.2.7.1 Wanpipe (for Sangoma TDM Cards)
* 1.2.8 Libpri
* g729a codec x5

using xlite on separate computer

I have the manual and have read it.

[williamconley]

Have you followed the installation instructions for setting up vicidial? (Where did you stop?)

[Kerry C]

setup vicidial to the t, got to the point of instaling the codec, installed wrong version deleted codec, reinstalled new one for asterisk 1.2 registered, set up conf files for voip per provider specificaions.

got into admin area of vicidial, made campaign, loaded leads, went into agent area started dialing but waits for call to pick up, not sure whatt to do next, voip said this:

gzpxyj



Joined: 22 Mar 2009
Posts: 3 Posted: Mon May 25, 2009 11:22 pm Post subject: Need to configure trunk


________________________________________
If your print out is right, you have not configure your trunk yet. The registration string is not right and your vicidialnow has not been able to talk to your voip service provider yet, could not register your extension/voip number. Also, as someone posted, using the gui to configure carrier will not work.

__________________________________________________________


What this tells me is that I already had configured the trunk in the command line but I removed what I did because you have the GUI installed and usually it is recommended to use the GUI when there is one available.

What I need to do now is reconfigure the trunk via the command line and try again.

then after that he asked me to send him screen shots of the dialer screens which i did.

[williamconley]

Your best bet would be to begin at the beginning of the tutorial and work straight through until you have your FIRST problem ... then stop and fix that problem before continuing. Ask for help if you need it. Continuing when you have ANYTHING wrong ... will generally result in trying to fix the wrong problem, often for a LOOOONG TIME.

So, if you started at TUTORIALS ... A. ... with a fresh system ... you will likely meet with success (or minor failure which can be resolved). Reset all your configuration files to the defaults that came with the system and try again. Works much better that way. Start at the beginning, work towards the end ... if you have ANY flaws STOP ... and fix them before continuing.

[Kerry C]

Based on what you said I went through the tutorial got to

34. The customer's information should now appear and you will hear the call ringing.

page 11.

I never hear the call ringing

it says waiting for ring.

[williamconley]

In "A" before step "1" ...

If your phones and/or carrier trunks are not set up, then go
to Tutorials I and L after you have finished step 3 of this tutorial

Did we do this?

[Kerry C]

it seems that we are having a dial plan issue:

the dial plan is at 9

here is the extensions.conf

[outbound]
exten => _NXXNXXXXXX,1,goto(outbound,1${EXTEN},1)
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@XXXXXXXXX-outbound)
exten => _011.,1,Dial(SIP/${EXTEN}@XXXXXXXXX-outbound)
[inbound]
exten => xxxxxxxxxx,1,Answer

the x's mafeer @ and the inbound x's i did, but do we put 9 in the front exten

exten => _NXXNXXXXXX,1,goto(outbound,1${EXTEN},1)
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@XXXXXXXXX-outbound)

the reason why i say this is because if we put x in the vicidial instead of nine it gives me a extensions error in the agent login

[williamconley]

it seems that we are having a dial plan issue:

the dial plan is at 9

here is the extensions.conf

[outbound]
exten => _NXXNXXXXXX,1,goto(outbound,1${EXTEN},1)
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@XXXXXXXXX-outbound)
exten => _011.,1,Dial(SIP/${EXTEN}@XXXXXXXXX-outbound)
[inbound]
exten => xxxxxxxxxx,1,Answer

the x's mafeer @ and the inbound x's i did, but do we put 9 in the front exten

exten => _NXXNXXXXXX,1,goto(outbound,1${EXTEN},1)
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@XXXXXXXXX-outbound)

the reason why i say this is because if we put x in the vicidial instead of nine it gives me a extensions error in the agent login

You know, I'd really like to help you ... but i don't understand what the dial plan is at 9 means (since none of the extensions listed start with 9 which was my only guess) and the rest of the above doesn't seem to resemble anything in the tutorials.

I don't remember an "outbound" or "inbound" context in extensions.conf, all the vicidial standard systems i've set up use default (certainly in "beginner" mode). This looks a little more like a few pieces of several extensions tossed together. You really should go back to the tutorials and just go through it slowly using the examples built into the system (but that's just my opinion ...).

If this is your first system, it is considered a best practice to change as little as possible and use the given examples to get a "functional" system before adding cool things you know about asterisk (that may not work in vicidial) and certainly before attempting to learn asterisk dialplan programming on the fly.

Just a thought.

[Kerry C]

This is the settings that were emailed to me from the voip:

Kerry, please setup your SIP with the following settings.

This is in your /etc/asterisk/sip.conf

[bestvoiopusa-inbound]
type=friend
dtmfmode=auto
host=xx.x.xxx.xx
context=inbound
allow=all
insecure=very
canreinvite=no
[bestvoipusa-outbound]
type=friend
dtmfmode=auto
host=xx.x.xxx.xx
allow=all
canreinvite=no

The following is in extensions.conf

[outbound]
exten => _NXXNXXXXXX,1,goto(outbound,1${EXTEN},1)
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@bestvoipusa-outbound)
exten => _011.,1,Dial(SIP/${EXTEN}@bestvoipusa-outbound)
[inbound]
exten => 9047582433,1,Answer

We are ready to process call and route calls to your server with the TEMP TEST inbound DID 904-758-2433

[Kerry C]

in the campaigns area i have the Dial Prefix at:9

Dial Prefix: for 91NXXNXXXXXX value would be 9, for no dial prefix use X HELP

when I put x it says cannot find extension, when i put 9 it gets no ring and waits on ring

when i set up the leads i input 1 in phone code cause my lead list have no 1's in front of them.

"12. If you want to load all of the leads with the same phone_code or you did not put a phone_codes into
your lead file, then you should enter the phone_code in the Phone Code Override Field. In the USA and
Canada this field should be 1."

[Kerry C]

I asked my voip this:

here is a example of a working configuration for vicidial, why does my configuration not have a username or pass, why is there a outbound and inbound? If you are use to giving this configuration info to the user its because asterisk uses it, but with vicidial it is different;

register => andrew:XXXXX@207.182.133.26
[SIPtrunk]
disallow=all
allow=G729
type=friend
username=andrew
secret=XXXXXX
host=207.182.133.26
dtmfmode=inband
qualify=1000






SIPtrunk = SIP/SIPtrunk

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIPtrunk/${EXTEN:1}@SIPtrunk,55,o)
exten => _91NXXNXXXXXX,3,Hangup


Here is the information based on the manual that I need from the sip provider,


Server Carriers(Trunks)
Server Carrier trunks are how your ViciDial server connects to the phone network or other Asterisk or
VOIP servers.
Carrier ID - This field needs to be at least 2 characters in length and no more than 15 characters in
length, no spaces. This is the ID that will be used to identify the carrier for this specific entry
throughout the system.
Carrier Name - This is the descriptive name of the carrier entry.
Registration String - This field is where you can enter in the exact string needed in the IAX or SIP
configuration file to register to the provider. Optional but highly recommended if your carrier allows
registration.
Template ID - This optional field allows you to choose a conf file template for this carrier entry.
Account Entry - This field is used if you have not selected a template to use, and it is where you can
enter in the specific account settings to be used for this carrier. If you will be taking in inbound calls
from this carrier trunk you might want to set the context=trunkinbound within this field so that you can
use the DID handling process within VICIDIAL.
Protocol - This field allows you to define the protocol to use for the carrier entry. Currently only IAX
and SIP are supported.
2009-04-02 version 103 ©2009 Matt Florell
Globals String - This optional field allows you to define a global variable to use for the carrier in the
dialplan.
Dialplan Entry - This optional field allows you to define a set of dialplan entries to use for this carrier.
Server IP - This is the server that this specific carrier record is associated with.
Active - This defines whether the carrier will be included in the auto-generated conf files or not.


Even in the extensions.conf it gives examples on what type of info is needed and it states it thoroughly in the comment brackets.


They responded with this:

Kerry, the registration string is only for trunks that require a registration. We have configured you for IP authentication so that is not needed.

The other thing to remember is that when you are attempting to make those outbound calls you are not making it to the trunk at all during out testing. What this means is, if it were the registration string that was causing the calls to fail then that is the error we would see in the CLI. When we are watching the CLI the call is not making it to the trunk in the earlier testing.

The reason for an inbound trunk is for calls to make it to your server we are sending them from the INBOUND side. Again, this is not affecting the calls in our out making it to the trunk.

This is from our switch:
It is strongly recommended that you do not register with us and that you use IP-based authentication instead.
Your IP is registered with our switch.
Registration is mostly with IAX trunks and will not provide the robust connection required to engineer your setup. It can be done with SIP but we do not recommend it that is why we are IP authenticated. Again this is not your error from earlier testing but based on your info above I understand why you are asking.

Does this help?


Any help on this ????????????????

[williamconley]

In "A" before step "1" ...

If your phones and/or carrier trunks are not set up, then go
to Tutorials I and L after you have finished step 3 of this tutorial

Did we do this?

Did you complete Tutorials I and L?

[Kerry C]

yes and this was the string my voip provided:

for the L tutorial

[bestvoiopusa-inbound]
type=friend
dtmfmode=auto
host=xx.x.xxx.xx
context=inbound
allow=all
insecure=very
canreinvite=no
[bestvoipusa-outbound]
type=friend
dtmfmode=auto
host=xx.x.xxx.xx
allow=all
canreinvite=no

The following is in extensions.conf

[outbound]
exten => _NXXNXXXXXX,1,goto(outbound,1${EXTEN},1)
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@bestvoipusa-outbound)
exten => _011.,1,Dial(SIP/${EXTEN}@bestvoipusa-outbound)
[inbound]
exten => 9047582433,1,Answer


In the I tutorial I have it set up as follows:


extension: cc100
Dial Plan Number: 100
Voicemailbox: 100
Outbound Caller ID: 00000000
No Phone Ip
There is a computer ip but it generated it not me
the sever ip is there
login is correct, pass is correct
status is active
account active
full name is correct
protocal is sip
gmt -5
manager user and pass is correct

[williamconley]

Let's try this:

First, forget the "inbound" until you have "outbound" working (different tutorial!).

Second, let's add the carrier for the outbound calls. So click on "Add New Carrier".

Carrier ID: BESTVOIPUSA

Carrier Name: BestVoipUSA

Registration String: register=> (contact your provider for this string)

Account Entry:
[bestvoiopusa]
disallow=all
allow=all
type=friend
username=(contact your provider for this)
secret=(contact your provider for this)
auth=(contact your provider for this)
host=xx.x.xxx.xx
dtmfmode=rfc2833
context=trunkinbound
insecure=very
canreinvite=no
qualify=500

Protocol: SIP (? I assume)

Globals String:
DIAL8TRUNK = SIP/bestvoipusa

Dialplan Entry:
exten => _81NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81NXXNXXXXXX,2,Dial(${DIAL8TRUNK}/${EXTEN:1},,To)
exten => _81NXXNXXXXXX,3,Hangup

Server IP: Choose your server from the dropdown (should already be there)

Active: Yes (after you submit the first time, this will become available)

After you've changed Active to "Yes" you can go to "ADMIN->Servers" and select your server. Then go to the bottom and verify that "Rebuild conf files" is "Y" (change it if it isn't, but it should be) and submit. Then you should wait one to two minutes for the system to regenerate the configuration files and reload asterisk, then you are ready to test. If you watch the Asterisk CLI while doing all this, you can see it happen and test immediately upon reload.

Your register string and username/pwd/authtype may not be necessary if you have provided the VOIP carrier with your IP address. In these cases authentication is handled already and no longer necessary, in which case those items may be entirely omitted.

I have used the standard "Dial()" function, but if your CLI with SIP DEBUG on indicates that your carrier is rejecting the call based on the format of the Dial command, you may need to modify it to fit their needs. Be sure to include the necessary options. DO NOT omit 1,AGI and 3,Hangup and do not alter those lines from your dial code.

Please note: This dial plan is set up for DIAL 81NXXNXXXXXX not just dialing the number, dialing 1+ number or dialing 91+number. It's for 81+number for 10 digit numbers within the US.

To test, you create a campaign, create a list, add a few leads to it (like your cell phone) using the previous tutorials in the managers manual. Then you Log In as an Agent (you MUST hear "you are the only person in this conference, or you are NOT logged in), then click on "Resume" and watch your Asterisk CLI.

[Kerry C]

with your settings I recieved this from the cli once i was in the agent area and tried to dial:

== 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
> Channel SIP/201-0820c670 was answered.
-- Executing MeetMe("SIP/201-0820c670", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Refreshing DNS lookups.
== 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-5b40,2", "8600051|F") in new stac k
> Channel Local/8600051@default-5b40,1 was answered.
-- Executing AGI("Local/8600051@default-5b40,1", "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-5b40,1", "Zap/g2/12033338500||To") in new stack
May 30 13:38:53 NOTICE[26074]: 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-5b40,1", "") in new stack
== Spawn extension (default, 912033338500, 3) exited non-zero on 'Local/860005 1@default-5b40,1'
-- Executing DeadAGI("Local/8600051@default-5b40,1", "agi://127.0.0.1:4577/c all_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new sta ck
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----1 6-----CHANUNAVAIL---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@def ault-5b40,2'
-- Executing DeadAGI("Local/8600051@default-5b40,2", "agi://127.0.0.1:4577/c all_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----0 --------------- completed, returning 0



*******************

still no ring.

[Kerry C]

this was taking off the 8's you had, the 9 is in the campaigns, and 1 was set in the load leads setup.