Vicibox set up but no VOIP provider

[alin.iacob]

Hello,

I installed a vicibox server (latest, for the concerning matter I don't think the version matters) and I am now stuck in finding a Voip provider.

I don't really have the big picture of the whole configuration. Until now, my call center was set-up on my clients platform, and we had to pay a monthly fee.
I now want my own platform and I could use some help in defining the steps and finalizing the platform configuration.

I tried with callcentrevoip com, but when trying to use their SIP trunk, I get error : ""Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)". No real support from them.

My call centre is based in Eastern Europe.

Can you please help/guide me in setting up the call center platform? I only need outbound campaign and planning to call in Italy.

Regards,
Alin.

[striker]

can you post your sip trunk settings entered under Carrier page.


also post the dialplan used to dialout, and the number u r dialling.
also post the below command outputs

#asterisk -rx "sip show peers"
#asterisk -rx "sip show registry"

[alin.iacob]

AccountEntry:

[callcentrevoip]
type=peer
username=1576
secret={my_secret}
host=67.215.225.96
dtmfmode=rfc2833
context=default
insecure=very
disallow=all
allow=ulaw
canreinvite=no
qualify=1000

DialPlan
exten => _3361XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _3361XXXXXXXXX,2,Dial(${CALLCENTREVOIP}/${EXTEN:2},60,tTo)
exten => _3361XXXXXXXXX,3,Hangup

exten => _331XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _331XXXXXXXXXX,2,Dial(${CALLCENTREVOIP}/${EXTEN:2},60,tTo)
exten => _331XXXXXXXXXX,3,Hangup

Trying to dial 3312127539393 (some random NY number)

sip show peers
Name/username Host Dyn Forcerport ACL Port Status
callcentrevoip/1576 67.215.225.96 N 5060 UNREACHABLE
gs102/gs102 (Unspecified) D N 0 UNKNOWN
test01/test01 (Unspecified) D N 0 UNKNOWN
test02/test02 192.168.124.1 D N 20580 OK (8 ms)
4 sip peers [Monitored: 1 online, 3 offline Unmonitored: 0 online, 0 offline]
(I am using XLite with test02)

sip show registry
Host dnsmgr Username Refresh State Reg.Time
67.215.225.96:5060 N 1576 105 Registered Mon, 17 Mar 2014 18:14:42
1 SIP registrations.


when calling , the result is:

== Using SIP RTP CoS mark 5
[Mar 17 18:16:27] -- Executing [3312127539393@default:1] AGI("SIP/test02-00000001", "agi://127.0.0.1:4577/call_log") in new stack
[Mar 17 18:16:46] -- <SIP/test02-00000001>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Mar 17 18:16:46] -- Executing [3312127539393@default:2] Dial("SIP/test02-00000001", "SIP/callcentrevoip/12127539393,60,tTo") in new stack
[Mar 17 18:16:46] WARNING[5294]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Mar 17 18:16:46] == Everyone is busy/congested at this time (1:0/0/1)
[Mar 17 18:16:50] -- Executing [3312127539393@default:3] Hangup("SIP/test02-00000001", "") in new stack
[Mar 17 18:16:50] == Spawn extension (default, 3312127539393, 3) exited non-zero on 'SIP/test02-00000001'
[Mar 17 18:16:50] -- Executing [h@default:1] AGI("SIP/test02-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Mar 17 18:16:50] -- <SIP/test02-00000001>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

thank you for helping me.

[striker]

sip show peers
Name/username Host Dyn Forcerport ACL Port Status
callcentrevoip/1576 67.215.225.96 N 5060 UNREACHABLE


it shows unreachable.
make sure you have internet in your server and able to ping 67.215.225.96
open ports 5060 and 10000 to 20000 in your firewall,if its blocked

[williamconley]

1) what country are you dialing into (knowing where YOU are is an implication, of course, but specifying is always a good idea).

2) it is ALWAYS required that you post (as a bare minimum) your vicidial version with build.

3) Unable to create channel is often related to the "qualify" setting. It is also often related to a completely missing sip connection. The reason is that "qualify=yes" in the carrier->account settings will cause the asterisk server to send a packet to the carrier and expect a response. The timing of the response matters. If it takes too long (1 second max is the default), asterisk will "disqualify" the carrier in question. This behavior is designed to allow better connections to survive and poor connections to be disqualified. Of course, if they are your ONLY connection, that does not help much. Most of the time, however, the failure is due to a complete lack of response (firewall issues, externip= setting in the sip.conf file, many factors could be involved). In this case, in addition to being disqualified, which causes asterisk to refuse to attempt a call, even if you convince asterisk to attempt (qualify=no will do this) there may still be no response to the call attempt just like there was no response to the qualify attempt.

FYI: We ordinarily only charge $25 to set up a standard carrier. And we do provide carrier names to clients who make such a request. We do this because (a) it's easy and we do it all the time and (b) this is a basic setup task that is often a dealbreaker when trying to set up Vicidial. Once past this hump, things tend to get easier.

Also be sure you have downloaded and read the entire FREE version of the Vicidial manager's Manual from EFLO.net. Very Handy.

[alin.iacob]

sip show peers
Name/username Host Dyn Forcerport ACL Port Status
callcentrevoip/1576 67.215.225.96 N 5060 UNREACHABLE


it shows unreachable.
make sure you have internet in your server and able to ping 67.215.225.96
open ports 5060 and 10000 to 20000 in your firewall,if its blocked

striker, the ping to address 67.215.225.96 works. in that list, i think the relevant peer is the last one (which shows OK). the first one that is shown as Unreachable is probably one of my tests.
I might be wrong with this, but keep in mind that my IP is the one shown with OK in the list. What should the column of hosts show?

[alin.iacob]

thank you for your time William.

Version:
Asterisk 1.8.23.0-vici built by abuild @ build11 on a x86_64 running Linux on 2013-08-07 00:55:50 UTC (installed a ViciBox5.x86_64-5.0.3 preload).

I changed the qualify=no and now I get the following response:
[Mar 19 16:15:09] == Using SIP RTP CoS mark 5
[Mar 19 16:15:09] -- Executing [3312127539393@default:1] AGI("SIP/test02-00000003", "agi://127.0.0.1:4577/call_log") in new stack
[Mar 19 16:15:10] -- <SIP/test02-00000003>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Mar 19 16:15:10] -- Executing [3312127539393@default:2] Dial("SIP/test02-00000003", "SIP/callcentrevoip/12127539393,60,tTo") in new stack
[Mar 19 16:15:10] == Using SIP RTP CoS mark 5
[Mar 19 16:15:10] -- Called SIP/callcentrevoip/12127539393
[Mar 19 16:15:10] WARNING[2908]: chan_sip.c:21126 handle_response_invite: Received response: "Forbidden" from '"test02" <sip:02@79.114.83.199>;tag=as697c114c'
[Mar 19 16:15:10] == Everyone is busy/congested at this time (1:0/0/1)
[Mar 19 16:15:10] -- Executing [3312127539393@default:3] Hangup("SIP/test02-00000003", "") in new stack
[Mar 19 16:15:10] == Spawn extension (default, 3312127539393, 3) exited non-zero on 'SIP/test02-00000003'
[Mar 19 16:15:10] -- Executing [h@default:1] AGI("SIP/test02-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CHANUNAVAIL----------") in new stack
[Mar 19 16:15:11] -- <SIP/test02-00000003>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


About the firewall: I turned it off.
I have briefly read the Manager Manual.
Do you think you are able to solve my problem? What extra information do you need?

[williamconley]

You've already solved your problem.

WARNING[2908]: chan_sip.c:21126 handle_response_invite: Received response: "Forbidden" from '"test02"

I'd say that is quite clear. The carrier has responded "bite me, you can't dial here! go away!".
Nothing *I* can do to resolve that, only your carrier can do so. You may wish to have a look at sip debug output and see if there is a more detailed text response (sometimes there is ... such as "Payment Required"). But in the end you will likely have to contact your carrier and ask "Why?"

[alin.iacob]

OK... what other carrier provider can I work with, that also integrates with Vicidial?

[williamconley]

Any VOIP Provider will do. There are lists all over the place. I am prohibited from sharing Vendor names unless you're a client. But there are many mentioned here on the forum as well.

We also have one for domestic US (dialer deck or conversational deck) that we provide free of charge even to non-clients (and we install this one for free as well). But you'd have to be asking that question at the office, not here on the Freebie section

[gardo]

You can try our VoIP services at http://justgovoip.com. We have some configuration examples in the Tutorials section. And we can configure the Vicidial carrier settings (for JustGoVoIP trunk) for you for free if you load up $80 in your VoIP account.

If you're using GOautodial CE 3.0, you can easily add carriers via our "Add Carrier Wizard". You can check it here: http://goautodial.org/projects/goautodi ... Credits_30. Almost everything is done via point and click. No need to mess around with the configuration files.

[williamconley]

You can try our VoIP services at http://justgovoip.com. We have some configuration examples in the Tutorials section. And we can configure the Vicidial carrier settings (for JustGoVoIP trunk) for you for free if you load up $80 in your VoIP account.

If you're using GOautodial CE 3.0, you can easily add carriers via our "Add Carrier Wizard". You can check it here: http://goautodial.org/projects/goautodi ... Credits_30. Almost everything is done via point and click. No need to mess around with the configuration files.

From what I hear, it does seem to work with many carriers. Have not heard complaints. But I will say that when it fails it's not a big deal because you still have the regular setup screens available.