vicidial.org

Hi,

I've spent 3 days trying to get GoAutoDial to work but with no success.

Phones are connecting, we are connected to our VOIP provider, but whenever we place a call, we get an error.

Below is the errors we are getting.

I should note that we are in Australia and I have tried calling several types of numbers with no luck.

Please Help!!!!!!!!!!!!

Our account entry is

[goautodial]
disallow=all
allow=ulaw
type=friend
secret=xxxxxx
username=xxxxxx
host=ourhost
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
fromdomain=ourhost

Our dial plan entry is

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:2}@goautodial,,tTor)
exten => _91XXXXXXXXXX,3,Hangup

Here are the errors......


[Feb 13 15:50:16] -- Got SIP response 502 "Bad Gateway" back from 111.111.111.11
[Feb 13 15:50:16] -- SIP/goautodial-00000005 is circuit-busy
[Feb 13 15:50:16] == Everyone is busy/congested at this time (1:0/1/0)

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.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

Hi williamconley,

I've been through the FREE managers manual too many times and still can't get it working.

I've also looked through the forums but couldn't see anyone with this same problem.

My config is


GoAutoDial 2.2.1 from .iso | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

| Vicidial X.X.X Build XXXX | Asterisk X.X.X |

Perhaps you should consider getting a SIP Provider other than goautodial.

We have a SIP provider already.

I'm not quite sure what you mean?

this says you're still dialing goautodial ...?

exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:2}@goautodial,,tTor)

Have you successfully REGISTERED at your provider?

Yes we have successfully registered with our provider and have also registered a soft phone.

The problem occurs when we try to dial out.

seems like you'd have to try SIP debug and/or discuss what is wrong with your provider. there are many methods of authentication from externip and sendrpid through various user/pass and even auth "types". Did they give you a "sample context" for your sip.conf?

We do have a sample, but the strange this is, we have an Elastix PBX server running with our SIP provider and it works perfectly.

Is there any difference between GoAutoDial and ViciBox?

We really need to get this sorted out ASAP because our call centre is not functional otherwise.

GoAutoDial and Vicibox are both "installers" for Vicidial and all its supporting software. The main two differences are the OS and the "pretty screen" in GoAutoDial. Worthy of note is the lack of support by The Vicidial Group for CentOS (so Gardo is your main support line from a "Distro", or guys like me ...).

If your Elastix PBX (which is yet another way to say "FreePBX on Asterisk") works ... you should look at your sip.conf and "included" sip.conf files to find the setup. Also remember that if you have pointed specific ports to your "Elastix" box, and those ports are required for SIP ... you have a single conduit with two wires to run. Investigate using another port for SIP with this provider.

Of course ... if you have multiple IP addresses from your ISP and multiple network cards in your servers .. you COULD put both of them on external IP addresses and resolve the entire thing.

What country are you trying to call? The dialplan you have below is generally for US and Canada. Posting the Asterisk CLI from the beginning and end of your call will be helpful.

Our account entry is

[goautodial]
disallow=all
allow=ulaw
type=friend
secret=xxxxxx
username=xxxxxx
host=ourhost
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
fromdomain=ourhost

Our dial plan entry is

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:2}@goautodial,,tTor)
exten => _91XXXXXXXXXX,3,Hangup

i will assume that your are calling in the us..


SIPtrunk=SIP/goautodial ( set this first to your global string before the dial entry)


then try this dial plan dude

exten =>_91X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten =>_91X.,2,Dial(${SIPtrunk}/${EXTEN:1},,Tor)
exten =>_91X.,3,Hangup

so If I reinstall goautodial with FXO/FXS A 200 R card, it will do automatically the configuration of /etc/zapata.conf and /etc/asterisk/zaptel.conf and in /etc/asterisk/extensions.conf ??
Please advice

Haryer wrote:so If I reinstall goautodial with FXO/FXS A 200 R card, it will do automatically the configuration of /etc/zapata.conf and /etc/asterisk/zaptel.conf and in /etc/asterisk/extensions.conf ??
Please advice

1)

Both GoAutoDial and Vicibox Redux have INSTALL MANUALS that cover installation. Read them, they will help to understand hardware installation as well. (GoAutoDial.com & Vicibox.com)

2) Please list a MODEL NUMBER when asking a question like this, with the name of the manufacturer. Often this will make a difference

3) You should have started your own post! This is not relevent to the discussion in progress (but welcome to the party! when you respond, feel free to do it in your own post! Matt doesn't charge extra for starting your own )

Hi everyone,

I forgot to check back at the forum for a reply.

We are in Australia and we are trying to call phone numbers in Australia.

The general format of a phone number here is

for a landline 02 9555 5555

for a mobile phone 0411 111 111

I've changed the diaplan slightly but still no calls going out.

The current dialplan is

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial($(ourprovider)/${EXTEN:2},,tTor)
exten => _91XXXXXXXXXX,3,Hangup

The error message that now pops up is

Call Rejected: CHANUNAVAIL
Cause: 66 - Channel type not implemented.

Any help would be appreciated.

Thanks

1) versions: you have likely installed goautodial2.0ce, but you have not posted your vicidial version with build. These are required for the free help. GoAutoDial does not have a "2.2.1" that's likely your vicidial version, but missing the "build" (not your fault, Gardo kinda put it on like that and it's been confusing people ever since). The "Vicidial Version with build" looks like this and should be on almost every administrator web page: Vicidial X.X.X Build XXXX. So your signature should contain:
GoAutoDial 2.0CE from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Asterisk version is available by entering "asterisk -R" at the command line.

2) show asterisk command line output from an attempted call. this will help narrow down your issue.

3) detail the exact values of your campaign dial prefix and a single lead's "dial code" and "phone_number" (change the last four digits to XXXX for privacy).

4) do you know the exact string your carrier is expecting to allow you to successfully dial? (asterisk cli from a freepbx box showing a successful "dial" command with the following string sent to your provider will show this)

the Dial command in the "Dial Plan" entry is designed to "strip off" your "campaign dial prefix" and send the dial code + phone number to your voip provider. give us this information, and we can construct a successful dial command for you.

hi change the dialplan
instead of using the {} brackets you used ()

###original####
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial($(ourprovider)/${EXTEN:2},,tTor)
exten => _91XXXXXXXXXX,3,Hangup

####change it to #####
exten => _91X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91X.,2,Dial(${TESTTRUNK}/${EXTEN:2},,tTor)
exten => _91X.,3,Hangup

and in gloabals string use TESTTRUNK=SIP/ourprovider name used in account entry
and then make a test call by using 91 as prefix

Thank Striker

I made those changes and while I feel I'm getting closer, I received a different error this time.

This time it said

Call Rejected: CONGESTION
Cause: 27 - Destination out of order.

Any ideas?

check

paste the cli result here


1.whether your carrier is registered
in cli type sip show peers , ur carrier should show OK
and type sip show registry it should show registered

2. check wether u have the balance in your account
3. check what codec does the carrier supports and if they says g729 then check wether u have g729 codec in ur system by typing show translation in cli

Cause No. 27: destination out of order. This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term "not functioning correctly" indicates that a signaling message was unable to be delivered to the remote user: e.g., a physical layer or data link layer failure at the remote user, user equipment off-line, etc.

do you have permission to dial the country you're calling? (did you include a country code? do you have a sample dial string from your provider to know how to format your string?)

The carrier is definitely registered.

I checked -> sip show registry

We are in Australia and we are trying to call Australia -> so local calls.

We're doing this in Elastix and I know there were some issues at first, but our Elastix server has been running perfectly for years.

This is the output when we try to make a call with GoAutoDial

If we can use the settings from Elastix, which settings do we use and where do we put them?


2] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 4 00:03:12] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-8b7f,2", "8600051|F") in new stack
[Mar 4 00:03:12] > Channel Local/8600051@default-8b7f,1 was answered.
[Mar 4 00:03:12] -- Executing [91xxxxxxxxxx@default:1] AGI("Local/8600051@default-8b7f,1", "agi://127.0.0.1:4577/call_log") in new stack
[Mar 4 00:03:12] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Mar 4 00:03:12] -- Executing [91xxxxxxxxxx@default:2] Dial("Local/8600051@ default-8b7f,1", "SIP/ourcarrier/xxxxxxxxxx||tTor") in new stack
[Mar 4 00:03:12] -- Called ourcarrier/xxxxxxxxxx
[Mar 4 00:03:12] -- Got SIP response 502 "Bad Gateway" back from xxx.xxx.xxx.xxx
[Mar 4 00:03:12] -- SIP/ourcarrier-0000000e is circuit-busy
[Mar 4 00:03:12] == Everyone is busy/congested at this time (1:0/1/0)
[Mar 4 00:03:12] -- Executing [91xxxxxxxxxx@default:3] Hangup("Local/8600051@default-8b7f,1", "") in new stack
[Mar 4 00:03:12] == Spawn extension (default, 91xxxxxxxxxx, 3) exited non-zero on 'Local/8600051@default-8b7f,1'
[Mar 4 00:03:12] -- Executing [h@default:1] DeadAGI("Local/8600051@default-8b7f,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----27-----CONGESTION----------") in new stack
[Mar 4 00:03:12] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----27-----CONGESTION---------- completed, returning 0
[Mar 4 00:03:12] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-8b7f,2'
[Mar 4 00:03:12] -- Executing [h@default:1] DeadAGI("Local/8600051@default-8b7f,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Mar 4 00:03:12] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----0--------------- completed, returning 0
[Mar 4 00:03:16] == Manager 'sendcron' logged off from 127.0.0.1

and this time you got a completely different code back:

Got SIP response 502 "Bad Gateway" back from xxx.xxx.xxx.xxx

have you considered discussing this with your carrier?

if you want anyone to extract your elastix info ... we have to have something to extract it FROM ... perhaps you could show a CLI sample of a working call to the same number from elastix? (just the dial command, because the freepbx dialplan will generate 100 lines of code where vicidial generates 5 LOL)

What exactly should I copy and paste?

some basic check to ensure you have the right configuration.
1.using a softphone to register your carrier to ensure your configuration with softphone works. Using the same dial string to dial (removing the 91 in front of it)
2. check with your elastix to see what is the correct configuration to your carrier, compare it with your vicidial configure. You would find the difference
3. If your vicidial on local net, make sure you set the default gateway for your vicidial. This gateway can be your router local ip address. If your vicidial has public fixed ip address, set your gateway to your internet vendor's gateway ip address.
4. register a softphone with your vicidial and make sure it logged in without problem, then manually dial out using the string exactly the same as your dialing string with 91x.
5. Check you asterisk cli to see what output you would get. You can find out the error there.
6. the global string for the trunk and the dialing string should be translated correctly showing up in the asterisk cli
You should resolve the issue of not connecting with your sip vendor after these steps. Good luck

Just read on another forum about an upgrade our service provider is having. It seems this could be the problem.

below is the response

====================

[i]Hi,

We had a upgrade on the weekend, and it's likely changed the accepted signalling.

Please either:

* Remove P-Asserted-Identity or Remote-Party-Id headers from your SIP Signalling

OR

* See below example.

---

INVITE sip:1300887899@sip10.mynetfone.com.au:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.150:5060;branch=z9hG4bK-d8754z-591caa70b8520754-1---d8754z-;rport=61973;received=xxx.xxx.xxx.xxx
Max-Forwards: 70
Contact: <sip:09123456@xxx.xxx.xxx.xxx:5060>
To: <sip:1300887899@sip10.mynetfone.com.au:5060>
From: "0280088000"<sip: 09123456@sip10.mynetfone.com.au:5060>;tag=e53ba10d
Call-ID: MTY4YTMxOTE2MDk4ZTAzMThkYmFlY2QwY2MxZTE4MDE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent:
Content-Length: 332
Remote-Party-ID: "0280088000"<sip: 0280088000@sip10.mynetfone.com.au:5060>;party=calling

^ Please ensure that your remote party ID or P asserted identity string is exactly the same as the "from" string – The above example would fail because it is not the same.

Typically the P Asserted Identity would be your internal extension number OR your DID.

Just as example.

==================

Ok, now how to we fix this?

OMG - Believe it or not, we finally got it going...............

There are a few issues we need help with though

1) It takes a long time (at least 20-30 seconds) to show up the agent screen once logged in - the phone takes up to 20 seconds to start ringing

2) We want to use our Linksys phones to make outbound calls, we added the phone ip and computer ip in the phones section and the phones do ring after a while. The problem is that when you click on "hang Up" to end the call, it hangs up on the phone and you need to login again.

Any ideas?