Inbound DID not working

[ccetech20]

Hi

First of all, thanks a tonne for the community to have such high participation and developers for making such extraordinary effort towards call center and telephony.

I have just go the setup installed and customized to our crm software. We have installed
GoAutoDial CE 2.1
Kernel 2.6.18-238.9.1.el5.goPAE (SMP)

ON:
Processors 4
Model Intel(R) Xeon(R) CPU E5504 @ 2.00GHz
CPU Speed 2 GHz
Cache Size 4.00 MB
System Bogomips 16000.2
Memory 7.33 GB Available


I read the manager's manual as well before posting. We have requirement to have inbound traffic from existing TOLLFREEFORWARDING(dot)COM.
Now as per my talk with outbound provider, he says if you have asterisk you can point your inbound traffic directly to your own server.

We have static IPs as well. However, I'm clueless as to how to enable my installation to have inbound calls land to our dialer straight away. I read that I need to create inbound group and DID. Which I copied using "copy from" command which is in installation itself.

I am unable to proceed as it is confusing me which dial plan to use? Also, in our case we do not have a provider for inbound but want that the calls from 7IP addresses that the tollfreeforwarding have given us to be received by the static IP of our server.


Please help.


Clarke

[striker]

go to asterisk cli and make sure your call is landing .

in vicidial trunkinbound is the the context used to receive the incomming call to land into dialler.

but you are dialling via static ip , so i hope it will come under the default context. if it is a registred trunk then u can mention the context.

so if the calls is landing inside the asterisk , then put the dialplan under the default context and create necessary DID route,ingroup,campaing

[default]
exten => _DIDnumber,1,AGI(agi-DID_route.agi)

[ccetech20]

Hi Striker

Thanks for your response.

I have created in-group and campaign to test.

However, what I am unable to understand is that while creating a TRUNK what should I be doing? For inbound calls, we are not using any voip provider as the current provider said that the calls can be terminated directly to your servers since you have a static IP in Dialer. Also, tollfreeforwarding is giving us 7 static IP from where calls will originate and land in our system.

Now we want the ability to have it routed to agents screen with relevant information if at all it is in CRM or ELSE a new entry. [I guess CID should work as route handling in this case].

[striker]

as you said call directly terminated to the ip ,
so i hope call will land directly to the default context.


put the below line under the default context in /etc/asterisk/extensions.conf
exten => _DIDnumber,1,AGI(agi-DID_route.agi)

go to asterisk cli command : asterisk -vvvvvvr
and dial the incomming tollfree no, you can see some call flow , if not then u need to check with your provider.

[ccetech20]

So this is what I follow from above post:

at the tollfree provider end I put username@static ip so that it can be forwarded.

At our end:
Make Campaign -- Done
Make DID number with DID Extension as tollfree number
exten => _TOLLFREENUMBER,1,AGI(agi-DID_route.agi)

Right?

[ccetech20]

anyone?


TIA

[ctc_olsen]

Hi bud. I was in the same spot as you months ago, as William helped me, now I'm returning the favor of helping somebody too. The idea is to set up a a dial plan entry so VICIDIAL would know what to do once a call comes in. I can only show you the fundamentals, but you need figure this out on your own. You need to

1. Admin - Carriers - Set up Account entry, this will create a peer to your provider. More like a session. Our exact config below. Search for other examples online.

Code: Select all

[xxxxxxxxxx]
disallow=all
allow=alaw
allow=ulaw
type=peer
host=x.x.x.x   -IP
dtmfmode=rfc2833
canreinvite=no
qualify=4000

2. Create a dial plan entry. This is to let the vicidial know on what to do when a call comes in. Our exact config below. Search for other examples online.

exten => _981XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _981XXXXXXXXXX,2,Dial(${SIPTRUNKA}/${EXTEN:2},,tTor)
exten => _981XXXXXXXXXX,3,Hangup

Go to a sample campaign and set 98 for "Dial Prefix" and Omit Phone Code to no.

[ccetech20]

Thanks Olsen.

How do I work around the part where it says: registration string?

[ctc_olsen]

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.

Usually its blank but very it with your provider. let us know how this works out for you.

[ccetech20]

Scenario:
We have tollfree from a provider and 3 DID numbers. Our ISP has given us static IP address as well.

Now someone told me that with vicidial I can configure inbound numbers directly, that means I do not have to pay for airtime/channel to anyone else. Essentially, this would mean that the provider will forward the TollFree over to my static ip. He says all he needs is SIP address in following form anything@ipaddress.

[williamconley]

In case you did not know it: a Toll Free Number is also a DID. And to Vicidial, there is no discernable difference at all. The manager's manual covers setup of a DID, these instructions apply equally to 800NXXXXXX as they do to NXXNXXXXXX and of course 866NXXXXXX and 877NXXXXXX (ie: All Toll Frees just like all regular DIDs). Vicidial neither knows that there is a difference nor cares.

[ccetech20]

Finally!

Its all working now. Thanks all.

@William, manual and the heads up helped

[williamconley]

cool. now go make some $$ and then come back and help newbies.

[sana1990]

So this is what I follow from above post:

at the tollfree provider end I put username@static ip so that it can be forwarded.

At our end:
Make Campaign -- Done
Make DID number with DID Extension as tollfree number
exten => _TOLLFREENUMBER,1,AGI(agi-DID_route.agi)

Right?

[sana1990]

i have same problem , i bought a number from toll free forwarding but i don't have idea that how to configure the trunk setting
plz give some suggestion thank you

[williamconley]

Welcome to the Party!

toll free forwarding means you must forward the toll free number to a DID. from another provider. then you would provision that DID into your vicidial system from the other provider. But now you're paying for the call twice.

regardless of all that: To use Vicidial you would start with the Vicidial Manager's Manual available at EFLO.net. Download at least the free version and begin at page one. When you have completed ALL the exercises in the book, you'll probably not have any other questions! However, if you do have a problem, you can bring us the Manual Version, Manual Page, and Line number and your actual error, copied and pasted, and we'll walk you through it.

And since you're obviously a newbie and trying hard, here are your newbie suggestions:

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
___________________

Happy Hunting!

[sana1990]

sir i do every thing but when i login it show me...... Sorry, your phone login and password are not active in this system, please try again:
plz tell me why it is happen? thank you

[sana1990]

register string- 111@107.159.0.129
My carrier set up:
[mydivert]
type=friend
host=107.159.0.129
context=trunkinbound
allow=alaw
canreinvite = no
qualify = yes

dial plan:

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

[sana1990]

plz reply williamconley

[williamconley]

sir i do every thing but when i login it show me...... Sorry, your phone login and password are not active in this system, please try again:
plz tell me why it is happen? thank you

Your phone login is on the Admin->Phones->(choose a phone), field name "Agent Screen Login" and the password is "Login Password". Many people use the User login and pass where the phone login and pass belong. The phone must also be active, of course.

[sana1990]

Sorry, there are no available sessions
Login:agent001
Password:goautodial
Campaign:xxxxxxx

[sana1990]

it show when i login in with agent webclient

Sorry, there are no available sessions
Login:agent001
Password:goautodial
Campaign:xxxxxxx



my carrier setting


MODIFY A CARRIER RECORD: INET_CARRIER
Carrier ID: INET_CARRIER
Carrier Name: xxx
Carrier Description: xxxx
Registration String: xxxx
Template ID: xxxx
Account Entry: [INET_CARRIER]
disallow=all
allow=g729
allow=gsm
allow=ulaw
type=friend
host=216.52.221.140
dtmfmode=rfc2833
context=trunkinbound

Protocol: sip

Globals String:

Dialplan Entry: exten => _8777865099,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8777865099,2,Dial(${111@107.6.13.173}/${EXTEN:2},,tTor)
exten => _8777865099,3,Hangup HELP
Server IP: xxxxxxxx
Active: yes

[sana1990]

plz reply williamconley

[sana1990]

I RESOLVE IT BUT NOW SHOW ANOTHER ERROR LIKE
WHEN I LOGIN
There is a time synchronization problem with your system, please tell your system administrator


Go Back

[williamconley]

I RESOLVE IT BUT NOW SHOW ANOTHER ERROR LIKE
WHEN I LOGIN
There is a time synchronization problem with your system, please tell your system administrator


Go Back

when you have a new error, you should create a new post topic with that error (after searching for this error in the forums directly and/or via google).

did you change the IP of your server after you built it?

[sana1990]

yes but it show same error

[williamconley]

yes but it show same error

somehow i doubt that. your old error was a problem with a did, but now you have a time sync error. plus you did not answer my question. i also get the general impression that you have not searched the forums for another user with the same error so you could learn how to fix it. it's a fairly common error with some fairly common causes. which is why i asked the question i asked ... happy to help, but you gotta do your part or we're kinda stuck. (like asking the mechanic to fix your car ... and never bothering to bring it to the shop!)

[sana1990]

if i have static ip then i no need to configure carrier setting for inbound call plz reply williamconley

[williamconley]

if i have static ip then i no need to configure carrier setting for inbound call plz reply williamconley

carrier settings are not affected by the "status" vs "dynamic" nature of an ip address.

but you still have not answered whether you changed your IP address ... or do you have a different problem you are trying to address now?

[sana1990]

williamconley i said i done every thing but i want to know that if my voip provider provide me a static ip like aything@yourip
so then i will configure my tollfree number in DID entry in extension .conf
exten => your DID,1,AGI(agi-DID_route.agi)
then do i need to configure my carrier setting? plz reply

[williamconley]

williamconley i said i done every thing but i want to know that if my voip provider provide me a static ip like aything@yourip
so then i will configure my tollfree number in DID entry in extension .conf
exten => your DID,1,AGI(agi-DID_route.agi)
then do i need to configure my carrier setting? plz reply

1) voip providers do not provide a static ip to you (in fact, they provide nothing except a DID). Although they may provide you with THEIR IP, which you would use in the Admin->Carrier settings to properly configure the inbound carrier.
2) start at the beginning of the Vicidial Manager's Manual and do not skip any exercises. Of course, if you do skip something you'll end up here asking for support for something that's in the manual ... LOL (I do believe that configuring a DID is included in the manual so ...)
3) if you follow the manual to configure inbound DID routing, it will work. and it will not involve modifying any .conf files. If you complete the inbound DID exercise in the manual and you have a problem, bring us the Manual Version, Page Number, Line Number and a detailed description of the problem ... and we'll walk you through it.

[sana1990]

how to upload agent-UI-06112011.fix.tar.gz in go auto dial

[williamconley]

how to upload agent-UI-06112011.fix.tar.gz in go auto dial

you're just all over the place today aren't you. LOL

i've never patched a goautodial box (we generally just "fresh install" for a clean system).

but I will say that gardo is usually very careful about having documentation available for this sort of thing on his site (goautodial.org). I believe the site has a wiki on it with documentation for this very purpose.

oh: and you should really create a new thread instead of continually asking fresh, unrelated questions on an existing thread. just sayin'

[sana1990]

can i ask one Q.
about go auto dial
if my internet provide provide me static ip then i need to add carrier setting
and my tollfree provide me this type of service
Enter the SIP address you’ve setup in your SIP system into the Ring-To field.
The SIP address must match the address setup in your SIP system - userpart@host/domainpart.

For example:
anything@yourIPaddress
or
anything@yourdomain.com


Manage Phone Numbers>Phone Numbers – Call Action (New)>Call Forwarding>Create New Ring-To


Ring-to Notes:
• IP addresses can have fewer numbers (like 1.2.3.4) but always have four sets of numbers separated by three periods.
• Domains can be as varied as any email address and do not need to end in ‘com’. They can end in .uk or .net or .org or anything that the customer specifies.
• Ensure no spaces: user name @domain.com

[williamconley]

you would just use your IP address instead of a domain name.

it does not matter whether it is static or dynamic, as long as you change it in their system whenever you get a new IP.

you would likely use something like this:

77777@111.111.111.111

where 77777 is a DID you set up in your system (under "inbound->DIDs") and 111.111.111.111 is replaced with your IP address. Domain is not necessary for inbound calls to Vicidial. Although I've never tested this exactly.

[sana1990]

my tollfree number is 18777865099 how to set in in did plz told me

[sana1990]

and plz explain it (under "inbound->DIDs")

[williamconley]

that is what the manual is for. have you downloaded and worked all the way through the Vicidial Manager's Manual? this would be a "carrier" for inbound calls and that carrier would be providing a DID.

you'll need to create a DID for 18777865099 and/or one for 8777865099, depending on which one the carrier sends. but you can always find that out by watching the asterisk CLI during an inbound phone call to see what values are being sent to you.

if you at least get SOME activity on the asterisk cli during an inbound call, it will eventually work.

to get the asterisk cli:

Code: Select all

asterisk -R

if you do not see any activity during an inbound call attempt, you can turn on debugging with

Code: Select all

sip set debug

or

Code: Select all

sip set debug on

depending on your asterisk version.

[sana1990]

sir as i say to you. i have bought a number from tollfreeforwarding.com plz tell me how to setup carrier with these detail and how my call in route plz sir

i ready to pay

[williamconley]

at this point, i would recommend you pay someone to configure it for you. we only charge $25. i'm sure others have similarly low fees for this.

i say this because you just asked the "base" question which is from several questions ago in this thread. unfortunately, we do not ever deal with that particular provider, so i can't just give you instructions.

if you were to provide a sample from that carrier's "suggested asterisk settings", I may be able to help. but otherwise we seem to be running in circles.

but here we go:

First phase is to get the call to arrive at your vicidial server. do they have a web interface that will allow you to "point" the Toll Free Number to your Vicidial server in some fashion?