GOautodial Open Source Omni-channel Contact Center Suite

So, I've battled with all that I can and have gotten a little further.

I've managed to stop the agent being logged out, and have the Bria phone pointing at the GoAutodial server address. I can log in, and get the call from GoAutodial informing me that I'm the only one in the conference.

I'm having the most trouble resolving the carrier or phone settings.

We were provided with 40 SIP trunks through a third party supplier. This supplier has informed me that the lines (through myphones.com) aren't compatible with Asterisk. Something to do with the Altos platform?

Has anyone hit this issue, and is there a method to overcome it?

I tried setting up the carrier with my web portal login details, and had an email from the supplier asking me to stop it as the system sees this as an attack. I have tried setting up one of the individual phone logins as a carrier, and this has gone further, but not allowed me to dial out.

For the each of the phones lines, I have the following:
Phone number - 0xxxxxxxxxx ->UK by the way
Status (Port) - Active (1)
Port Password - **
Service PIN - **

Is this enough credentials?

For each of the Bria settings (prior to GoAutodial server IP change) I have:

User ID: landline phone number
Domain: myphones.com
Password: corresponding for phone number as above
Display name:
Authorization name: landline phone number

Register with domain and receive calls: ticked
Proxy: proxy-soft.myphones.net:4144

Dial plan: #1\a\a.T;match=1;prestrip=2;

Can any of this be used for a workaround or do we need to change provider?

Hey Daniel,

It's good to hear that you got your GOautodial working. Based from your description, it seems that your carrier doesn't support dialer traffic. You need to have a SIP/VoIP provider that supports dialer traffic (check our's at http://justgovoip.com). Normally, you just need 1 SIP trunk (a SIP trunk that supports dialer traffic) that has multiple lines.

Can you tell us more what you are trying to achieve so we can help you better?

Hi Damian,

Thank you for the above. Unfortunately, I've not actually got up and running yet. Still having issues getting the carrier settings correct. I think you are correct on the lack of support from our SIP provider, and my main confusion with this may have come from the fact that we have 40 SIP tunks, not just one.

Ideally, I was trying to achieve the full potential of the GoAutodial system. Now, I'm wondering if there is a way for the system to work without assigning phones to agents. Is there a way to use the lead distribution/campaign management side of the software without phone connectivity? If the system can provide the agents with leads from a campaign, and the agent can change the status of the lead through the web browser, then the agent can manually 'click to dial' with Firefox launching the number in Bria as a separate entity.

Any of this plausible?

Hey Daniel,

It's possible to use the system just like a regular CRM. If you need the agents changing the status of the leads, they need to have access level 7 and up so they can login to the GOadmin application.

We are actually working on bringing CRM functionality on the next release. So agents can directly modify leads or any information (as long as they have permission to) related to the leads/campaigns.

Hi Damian,

Thank you for the above.

I've created a group at 'Supervisor level' (changed to Level 7). I created a user for this group, however cannot log them in on the agent side without a phone login. Equally, they won't log in at all on the admin side. I'm probably missing something obvious.

Going back to the earlier messages, I had a meeting with our SIP provider. They have looked at the situation and gave me a test trunk to have a go with. The trunk has no login credentials, and they've filled the carrier settings as best they can. I'm not getting further than the agent login. The Bria phone calls, the voice states 'I'm sorry, that's not a valid extension'.

Asterisk readout below:

go*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
166262 [SIP TRUNK SERVER IP HIDDEN] N 5060 OK (15 ms)
5000/5000 [GOAUTODIAL SERVER IP HIDDEN] D N 44760 OK (5 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
[May 5 08:27:13] NOTICE⁴³¹²: chan_sip.c:25829 handle_request_register: Registration from '"1851" <sip:1851@XX.XXX.XXX.XX:5060>' failed for 'XXX.XXX.XX.XXX:5075' - Wrong password
[May 5 08:27:33] NOTICE⁴³¹²: chan_sip.c:25829 handle_request_register: Registration from '"1349" <sip:1349@XX.XXX.XXX.XX:5060>' failed for 'XXX.XXX.XX.XXX:5072' - Wrong password
[May 5 08:27:35] NOTICE⁴³¹²: chan_sip.c:25829 handle_request_register: Registration from '"949" <sip:949@XX.XXX.XXX.XX:5060>' failed for 'XXX.XXX.XX.XXX:5086' - Wrong password
[May 5 08:27:45] NOTICE⁴³¹²: chan_sip.c:25829 handle_request_register: Registration from '"251" <sip:251@XX.XXX.XXX.XX:5060>' failed for 'XXX.XXX.XX.XXX:5082' - Wrong password
[May 5 08:28:01] Manager 'sendcron' logged on from 127.0.0.1
[May 5 08:28:01] Manager 'sendcron' logged off from 127.0.0.1
[May 5 08:28:01] Manager 'sendcron' logged on from 127.0.0.1
[May 5 08:28:01] Manager 'sendcron' logged off from 127.0.0.1
[May 5 08:28:06] Manager 'sendcron' logged on from 127.0.0.1
[May 5 08:28:06] Manager 'sendcron' logged off from 127.0.0.1
[May 5 08:28:17] Manager 'sendcron' logged on from 127.0.0.1
[May 5 08:28:17] Using SIP RTP CoS mark 5
[May 5 08:28:19] > Channel SIP/5000-00000003 was answered.
[May 5 08:28:19] Starting SIP/5000-00000003 at default,,1 failed so falling back to exten 's'
[May 5 08:28:19] Starting SIP/5000-00000003 at default,s,1 still failed so falling back to context 'default'
[May 5 08:28:19] -- Sent into invalid extension 's' in context 'default' on SIP/5000-00000003
[May 5 08:28:19] -- Executing [i@default:1] Playback("SIP/5000-00000003", "invalid") in new stack
[May 5 08:28:19] -- <SIP/5000-00000003> Playing 'invalid.gsm' (language 'en')
[May 5 08:28:19] Manager 'sendcron' logged off from 127.0.0.1
[May 5 08:28:21] NOTICE[30620]: res_rtp_asterisk.c:2361 ast_rtp_read: Unknown RTP codec 126 received from 'XX.XXX.XXX.XX:50142'
[May 5 08:28:23] -- Executing [i@default:2] Hangup("SIP/5000-00000003", "") in new stack
[May 5 08:28:23] Spawn extension (default, i, 2) exited non-zero on 'SIP/5000-00000003'
[May 5 08:28:23] -- Executing [h@default:1] AGI in new stack
[May 5 08:28:23] -- <SIP/5000-00000003>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[May 5 08:28:33] NOTICE⁴³¹²: chan_sip.c:25829 handle_request_register: Registration from '"352" <sip:352@XX.XXX.XXX.XX:5060>' failed for 'XXX.XXX.XX.XXX:5087' - Wrong password
[May 5 08:28:37] NOTICE⁴³¹²: chan_sip.c:25829 handle_request_register: Registration from '"849" <sip:849@XX.XXX.XXX.XX:5060>' failed for 'XXX.XXX.XX.XXX:5075' - Wrong password
[May 5 08:29:01] Manager 'sendcron' logged on from 127.0.0.1
[May 5 08:29:01] Manager 'sendcron' logged off from 127.0.0.1
[May 5 08:29:01] Manager 'sendcron' logged on from 127.0.0.1
[May 5 08:29:01] Manager 'sendcron' logged off from 127.0.0.1

Any assistance much appreciated.

Thanks!

By the way, current settings are as follows:

ACCOUNT ENTRY:

[166262]
disallow=all
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
host=[SIP SERVER IP]
allow=alaw
allow=ulaw
allow=g729

GLOBALS STRING

SIP166262 = SIP/166262

DIAL PLAN (UK ONLY)

exten => _44.,1,AGI
exten => _44.,2,Dial(SIP/${EXTEN:2}@166262,,tTo)
exten => _44.,3,Hangup

If agents are logging in on the agent application, a phone extension is really needed. You need to have the user at Level 8 to be able to login to the GOadmin.

Can you post the output of "screen -ls"?

Why not try our cloud packages? We provide 30 days free trial (on our Go Small package). You can check them out here: https://www.goautodial.com/pricing.php. Less headaches. You can be up and running in no time at all.

Hi Demian,

Thanks for getting back to me on this.

Level 7 was the issue on the agent login through admin portal. 8 works well.

I've done the output, it's as follows:

[root@go ~]# screen -ls
There are screens on:
25885.ASTVDremote (Detached)
4694.ASTupdate (Detached)
4712.ASTfastlog (Detached)
4703.ASTVDauto (Detached)
4700.ASTlisten (Detached)
3977.asterisk (Detached)
9051.ASTVDadapt (Detached)
4666.goautodial_d (Detached)
4697.ASTsend (Detached)
9 Sockets in /var/run/screen/S-root.

I would love to go for a cloud package, but want to tire this process first. Thank you for all of your help.

What's the exact issue you're having again?

By the way, looks like your server is getting a SIP brute force attack.

[May 5 08:27:13] NOTICE4312: chan_sip.c:25829 handle_request_register: Registration from '"1851" <sip:1851@XX.XXX.XXX.XX:5060>' failed for 'XXX.XXX.XX.XXX:5075' - Wrong password
[May 5 08:27:33] NOTICE4312: chan_sip.c:25829 handle_request_register: Registration from '"1349" <sip:1349@XX.XXX.XXX.XX:5060>' failed for 'XXX.XXX.XX.XXX:5072' - Wrong password
[May 5 08:27:35] NOTICE4312: chan_sip.c:25829 handle_request_register: Registration from '"949" <sip:949@XX.XXX.XXX.XX:5060>' failed for 'XXX.XXX.XX.XXX:5086' - Wrong password
[May 5 08:27:45] NOTICE4312: chan_sip.c:25829 handle_request_register: Registration from '"251" <sip:251@XX.XXX.XXX.XX:5060>' failed for 'XXX.XXX.XX.XXX:5082' - Wrong password

Currently, our SIP provider has issued a single test trunk, which can have a few extensions coming from it. The SIP trunk is IP registered, without username/password, and our static IP has been assigned access.

The current setup means that an agent will log in through the browser, the Bria phone (pointing to the GoAutodial server) will receive the call and the call states "I'm sorry, that is not a valid extension. Goodbye".

That's as far as I am at this time.

The TESTCAMP has leads, there is one user on the system with a registered phone at ext 5000. Bria is set up as ext 5000, pointing back to the GoAutodial server.

I feel like I'm missing a setting somewhere to do with the softphone, rather than the carrier at this stage.

As for the attack - would that be me? I can't imagine that's coming from outside of our office, we're not even up and running yet!

If you're GOautodial server is on the internet and port 5060 is not being filtered, chances are you are experiencing a SIP brute force attack (based on your Asterisk CLI logs).

It looks like you don't have a meetme conference available for your phone extension. This should be automatically created. Did you change your server's private IP address? Did you run "update_server_ip" after changing the IP address?

[May 5 08:28:19] > Channel SIP/5000-00000003 was answered.
[May 5 08:28:19] Starting SIP/5000-00000003 at default,,1 failed so falling back to exten 's'
[May 5 08:28:19] Starting SIP/5000-00000003 at default,s,1 still failed so falling back to context 'default'
[May 5 08:28:19] -- Sent into invalid extension 's' in context 'default' on SIP/5000-00000003
[May 5 08:28:19] -- Executing [i@default:1] Playback("SIP/5000-00000003", "invalid") in new stack
[May 5 08:28:19] -- <SIP/5000-00000003> Playing 'invalid.gsm' (language 'en')
</per>

Hi Demian,

Thank you for your help - we're finally up and running to capacity. It seemed that a post from Striker24x7 helped a lot:

Troubleshoot 4: Conference Sessions *********************************
Login to the vicidial/goautodial admin page and check whether the conference sessions are created under ADMIN-conferences, ADMIN-vicidial conferences
If it is empty you have create the sessions, instead of creating manually we can use the existing script to create in bulk , follow the below steps for the same

login to the server via ssh using putty and run below commands in linux shell

cd /usr/src
wget http://download.vicidial.com/vicidial/trunk/extras/first_server_install.sql
mysql -u crong -p 1234
mysql>use asterisk
mysql>\. /usr/src/first_server_install.sql
mysql>quit
/usr/share/astguiclient/ADMIN_update_server_ip.pl --old-server_ip=10.10.10.15
reboot

once rebooted and login as agent and check whether your issue resolved.

[[http://striker24x7.blogspot.co.uk/2014/03/sorry-there-are-no-available-sessions.html]]

Thank you for your support - keep up the great work!