- Frequently Asked Questions:
- Is GOautodial the same as VicidialNOW?
- Is GOautodial related to the Vicidial group?
- Is GOautodial free?
- What phones will work with GOautodial?
- Does GOautodial work with trunks other than SIP?
- What T1/E1/Analog telephony cards do you recommend?
- What about hardware? How do I know if a particular NIC or motherboard is compatible?
- Help! All my RAM is being eaten up! What do I do?
- How do I update my system?
- I am getting one-way or no audio on my calls. Why is that?
- What web browsers do you recommend?
- Why am I getting "choppy" calls? Why are most of my calls of poor quality? Are you inside a tunnel?
- I get "Sorry, there are no available sessions". What do I need to do?
- Agent XXX is currently in use or improper logout. Contact Administrator.
- Sorry, your phone login and password are not active in this
Frequently Asked Questions:¶
Is GOautodial the same as VicidialNOW?¶
Yes. We renamed the project to "GOautodial" since the word "Vicidial" is a registered trademark. The name change was necessary since GOautodial evolved from being more than just a Vicidial distribution. It's now a complete open source dialer system.
Is GOautodial related to the Vicidial group?¶
No! GOautodial is in no way related to the Vicidial group.
Is GOautodial free?¶
What phones will work with GOautodial?¶
Most any SIP compatible phone from companies like Aastra, Polycom, Linksys, SNOM, Cisco, and others will work, you want to make sure it is fully SIP compliant. You can also use a regular analog phone if you have a card with an FXS port on it or you can use an ATA (analog telephone adapter) to bridge between SIP and the analog phone. As long as it works with Asterisk, it will work with GOautodial.
Does GOautodial work with trunks other than SIP?¶
Yes. GOautodial also works with IAX, Analog and E1/T1 lines. It utilizes trunks being used by Asterisk. H.323 should also work but we haven't fully tested it and its also not installed and configured by default.
What T1/E1/Analog telephony cards do you recommend?¶
GOautodial is tested with Sangoma and Digium. It has out of the box support for the two. The important thing to remember is that as long as it works with Asterisk, it will work with GOautodial.
What about hardware? How do I know if a particular NIC or motherboard is compatible?¶
GOautodial is built on CentOS which is itself based on Red Hat Enterprise Linux. Current version of GOautodial use CentOS 5 as it's base.
Red Hat has a hardware compatibility list (HCL) for versions 3, 4 and 5 here: https://hardware.redhat.com/
Help! All my RAM is being eaten up! What do I do?¶
Probably nothing. The short answer is that Linux (and other Unix like systems) use RAM differently than you may be used to in other operating systems. The long answer is outside the scope of this FAQ, but the following links have more information: http://www.linuxhowtos.org/System/Linux%20Memory%20Management.htm
How do I update my system?¶
Just run the following command:
yum update -y
This will download and install the latest system updates. Updating the system regularly is recommended. Bug fixes and security patches are applied.
I am getting one-way or no audio on my calls. Why is that?¶
These problems are normally related to firewall/NAT issues. If your GOautodial server is behind a firewall, edit sip.conf:
;externip = 192.168.1.1
externip = 192.168.1.1
Where 192.168.1.1 is your public IP address. Reload Asterisk after the changes.
asterisk -rx "reload"
What web browsers do you recommend?¶
Mozilla Firefox and Google Chrome are highly recommended.
Why am I getting "choppy" calls? Why are most of my calls of poor quality? Are you inside a tunnel?¶
There are a lot of factors affecting the quality of calls. They are mainly:
- Asterisk codec being used by the server
- Agent workstation
- Bandwidth consumption
- Overloaded workstation
- Softphone (try to other softphones like zoiper, xlite and eyebeam)
- Poor quality headset (USB headsets are highly recommended)
If you have limited bandwidth, the codec used by your GOautodial server (to your SIP gateway) should either be GSM or G729. These are bandwidth efficient codecs.
You might also need to check the agents workstations to see if they're not overloaded. Meaning they're just running the necessary applications for dialing (Firefox, softphone, notepads or etc). Softphones eat CPU resources. If the workstation is overloaded then call quality can suffer.
Lastly, check your bandwidth consumption. You might be eating all your bandwidth. Make sure your internet connectivity is just being used for dialing purposes. Browsing social media sites like Facebook, Google+, Youtube and others will eat up most of your bandwidth.
I get "Sorry, there are no available sessions". What do I need to do?¶
Make sure in the admin section under admin->conferences->Vicidial Conferences your server ip is defined/displayed. If you have re-configured your network settings, make sure the new IP address is applied to your Vicidial configurations. Run the following command:
Agent XXX is currently in use or improper logout. Contact Administrator.¶
Follow the steps below:
1. Go to Telephony.
2. Choose and click USERS.
3. Select the Agent that having problem.
4. Click GREEN button (Info user AgentXXX).
5. Click FORCE LOGOUT.
Sorry, your phone login and password are not active in this¶
Check the following settings. Navigate to Telephony > Users > Choose the user > check if your phone login password matches with the Phone extension login password. Phone extension login password can be found by navigating to Admin Settings > Phones