The Digium® X100M FXO module and X400M FXO module are daughter cards that allow Digium® analog cards to terminate analog telephone lines (POTS).
This document is intended to be a brief description of toublehsooting steps that Digium’s customers can take to resolve Caller ID issues on their Asterisk Server.
Requirements
At least one analog line.
A regular analog phone with Caller ID display feature
Fully configured Asterisk server with the latest version of DAHDI and Asterisk
Troubleshooting Steps
A Caller ID issue could be caused by several factors. This section explores the possibility that the issue could be located at the telco or by noise on the lines.
Connect an analog phone to the demarcation point. In telephony, the demarcation point is the point at which the telephone company network ends and connects with the wiring at the customer premises.
Place a call — using another line or mobile phone — from the PSTN to the analog phone connected at the demarcation point and check if the phone displays Caller ID. If the Caller ID information is not being shown by the phone, please contact your telco and ask them how to enable this feature.
Otherwise, please disconnect the phone and plug it to the line that is directly connected or closest to your Asterisk server and repeat the test call. Continue reading How to troubleshoot CallerID issues on a FXO
I have taken a scenario of Elastix PBX install on two geographical location connecting over OpenVpn
and working as a VPN server and client. As we want to use OPENVPN for data and voice connectivity for both offices.
All our Internet traffic should go out through DSL Routers and only Voice and data traffic 192.168.1.x and 192.168.200.x should go over VPN Tunnel for KU to IN and IN to KU offices. Continue reading Elastix OpenVPN Configuration
missing whois program in fail2ban email alerts . you are not able to receive IP information .
#yum search whois
gnome-nettool.x86_64 : A GNOME interface for various networking tools
jwhois.x86_64 : Internet whois/nicname client.
perl-Net-Whois.noarch : Get and parse “whois” domain data from InterNIC
perl-Net-Whois-IP.noarch : Perl extension for looking up the whois information
: for ip addresses
Download the GoAutoDial CE 3.0 release from http://goautodial.com/download/ – Burn to CD using program like Nero on Windows or K3B on Linux and configure your server to boot from CD.
Installation
Before starting, please make sure machine is CONNECTED TO A NETWORK.
Boot machine from the GoAutoDial CD and hit Enter to get started.
The automated installer takes care of everything so you just need to wait for around 15 minutes depending on your hardware for the whole installation process to finish
Enter your desired root password
Halfway through the package installation
Installation Complete! Remove the installation CD then press ENTER to reboot.
After the reboot, you need to run an update. Type “yum update” on your console (see picture below). After the update completes, reboot your server.
By default DHCP is enabled. The IP of your server depends on what is assigned by your DHCP server.
GoAutoDial does not have a desktop manager installed so you need to access it via network from your workstation or login physically to the server.
How to Configure Static IP
1. via SSH or Putty type setup then Press Enter
2. Select Network Configuration
3. Edit Device
4. select eth0
5. click the SPACE Bar in your keyboard to unchecked the DHCP option after that input your Static IP
6. Edit DNS configuration
7.Then input DNS (e.g 8.8.8.8 / 208.67.222.222)
After saving the Network configuration Restart the Network services by running this command service network restart via SSH
How to change Time zone
1. via SSH or Putty type setup then Press Enter
2. select Timezone Configuration press enter
3. select Timezone then OK
4. after configuring your Timezone restart mysqld and httpd by typing /etc/init.d/mysqld restart and service httpd restart
5. after restarting mysqld and httpd type date to see Date and Time in your Server
Open your GoAutodial CE portal via a browser by putting the IP address on the address bar.
Login : admin
Password : goautodial|
How to setup an Outbound Campaign
1. Login to your portal as an administrator and Select TELEPHONY and CAMPAIGNS on the left side panel.
2. Click on ADD NEW CAMPAIGN and select the CAMPAIGN TYPE (Outbound).
3. Check the tickbox to manually edit the Campaign ID and Campaign Name
4. Browse to your Lead File directory, Select the Specific Country/Code then Select a Duplicate phone checker between Check for duplicates by phone in List ID and Check for duplicates by phone in all Campaign List then click Upload Leads once done.
5. Select the correct fields for your leads e.g. Phone Number, First Name, Last Name…
6. Lead status will be posted after you successfully uploaded your lead file, Click Next to proceed.
7. Select a preferred settings for your Campaign. For the Dial Method you can choose Manual, Auto_Dial or Predictive and for the Auto Dial Levelyou can choose Slow, Normal, Max or High. “Carrier to use for this Campaign” allows you to choose between Manually created carrier and an existing JustGoVoIP account for your campaign. Please note that only active carriers will appear/ will be shown. Click “Finish” once done.
Softphone Configurations
For this guide we can use the following SIP account:
Username:
8001
Password:
goautodial
Realm/Domain:
(Server IP)
Please visit the links below for 2 of the most popular freely downloadable softphones,
2. Input the agent credentials below then click LOGIN.
Agent Login:
agent001
Agent Password:
goautodial
After clicking on SUBMIT your softphone/extension will ring, ANSWER the call and you will hear “You are currently the only person in this conference” DO NOT HANG UP.
Note: You must input the correct credentials for the campaign name to show up. Click Refresh Campaign List to double check
3. Click Resume Button to start receiving calls if your campaign is using a Predictive dialing or Auto Dial method of dialing
4. To do Manual Dial click the Manual Dial Button on your agent interface
NOTE: Dial Code
For USA and Canada Route, use 1 Dial Code
For UK, use 44 Dial code
For AUS, use 61 Dial Code
Then input the Phone Number and Click Dial Now
The highlighted “LIVE CALL” will be the key/basis to know if the call was successfully connected
5. Click Hangup Button on the agent portal to hangup the call and select a Call Disposition
NOTE: When you are done with your shift, click on the “Logout” button in the upper right corner of the
agent screen.
Source link :- http://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev3
Disclaimer
The author of this document does not warrant or assume any legal liability or responsibility for the accuracy, completeness, or usefulness of any information, product, or process disclosed. Any consequences or results achieved directly or indirectly by this document or information are entirely your responsibility.
About This Document
The author of this document does not warrant or assume any legal liability or responsibility for the accuracy, completeness, or usefulness of any information, product, or process disclosed. Any consequences or results achieved directly or indirectly by this document or information are entirely your responsibility.
I found the below guide helpfull to start with GOautodial
Requirements
Download the GoAutoDial CE 2.1 final release from http://goautodial.com/download/ – Burn to CD using program like Nero on Windows or K3B on Linux and configure your server to boot from CD.
Installation
Boot machine from the GoAutoDial CD and hit Enter to get started.
The automated installer takes care of everything so you just need to wait for around 15 minutes depending on your hardware for the whole installation process to finish
Starting package installation
Halfway through the package installation
Package Installation almost finished
Running post-install scripts
Installation Complete! Press Enter and remove the installation CD.
System Configurations
Goautodial does not have a desktop manager installed so you need to access it via network from your workstation, the server’s default IP Address is192.168.1.2
Open http://192.168.1.2/ using Firefox (We highly recommend using Firefox web browser)
Default Gateway is not configured by default so you have to manually configure it, you can also change the IP Address from the same menu.
Click on Systems/Network > Configuration > eth0 config
Set the default gateway and click Go
Update Database IP
If you changed the IP address of the server you also need to update the IP addresses entries in the database:
Click on Systems/Network > Configuration > Update Database IP
Carrier Configuration
Before you can start dialing, you need to configure your carrier/trunk, if you do not have your own carrier yet you can sign up for an account athttp://goautodial.com/voip-store/
Click in VICIDIAL Admin > Admin > Carriers > Add A New Carrier
Activate Newly created Trunk by setting Active = Y and click Submit
How To Load Leads
1. If this is the first time you will load leads, you need to create a List first, go to Lists > Add a New List and make sure you select the campaign you will use, for this guide you can use the default campaign TESTCAMP
See below for a sample .csv format list/leads file:
2. Go to Lists > Load New Leads > Browse the list file in .csv format > type the Listid you created in the List ID Override > type 1 in the Phone Code Override if you are calling US numbers > Select Custom Layout > Submit
3. Select the appropriate fields via the drop down menu then click OK TO PROCESS
If the loading of leads is successful you will see something similar to this:
Go to Campaigns > TESTCAMP
You can see that the leads are successfully loaded and there are currently 5 leads in the dial hopper.
If the dial hopper is not being populated try to set the Local Call Time to 24hours:
Campaigns > TESTCAMP > Detail view> Set the Local Call Time to 24 Hours > Submit
Softphone Configurations
Below is a link for 2 of the most popular freely downloadable softphones, for this guide we can use the following SIP account:
Username: 8001
Password: goautodial
Realm/Domain: 192.168.1.2
Phone Login: 8001
Phone Password: goautodial
User Login: agent001
User Password: goautodial
Campaign: TESTCAMP
After clicking on SUBMIT your phone/extension will ring, ANSWER the call and you will hear “You are currently the only person in this conference” DO NOT HANG UP.
3. If the campaign is configured as Manual/Preview dialing you will see the agent screen below, Click DIAL NEXT NUMBER > DIAL LEAD , If the campaign is configured for Predictive Dialing Mode just click the RESUME button to start receiving calls.
If the call is successful you will see the LIVE CALL indicator turn to GREEN, If you are in Predictive Dialing Mode you will hear a short beep sound every time a call comes in.
4. After each call click HANGUPreverse phone lookup and select a CALL DISPOSITION.
noip provides free Dynamic DNS services ( http://www.noip.com/remote-access) as DynDns was providing but its allow only three hostnames as free.
This guide will walk you through the installation and setup of the Dynamic Update Client (DUC) on a computer running Linux. If you are using Ubuntu or Debian Linux please check our support site for guides on their specific setup.
I was trying to configure my Elastix voicemail to send voicemail on Email. but the email was not being delivered to the my email address. I had configured properly my remote SMTP under mail configuration. I was able to send email from my elastix mail system and emails was being delivered to my email address.
But Voice mail was not being delivered to users email address.
I started troubleshooting I checked the log
tail -f /var/log/maillog
I found below error message
Apr 16 01:29:26 localhost postfix/qmgr[1820]: 6B600C90A91: from=<asterisk@pockw.localdomain>, size=813, nrcpt=1 (queue active)
Apr 16 01:29:49 localhost postfix/smtp[1937]: 6B600C90A91: to=<gkhanin@outlook.com>, relay=gkhan.in[173.254.220.66]:25, delay=23, delays=0.02/0.01/22/0.33, dsn=5.0.0, status=bounced (host gkhan.in[173.254.220.66] said: 550-Verification failed for <asterisk@pockw.localdomain> 550-The mail server could not deliver mail to asterisk@pockw.localdomain. The account or domain may not exist, they may be blacklisted, or missing the proper dns entries. 550 Sender verify failed (in reply to RCPT TO command)) Apr 16 01:29:49 localhost postfix/cleanup[1934]: 7537CC90A92: message-id=<20160415222949.7537CC90A92@pockw.localdomain>
Apr 16 01:29:49 localhost postfix/qmgr[1820]: 7537CC90A92: from=<>, size=3184, nrcpt=1 (queue active)
Apr 16 01:29:49 localhost postfix/bounce[1939]: 6B600C90A91: sender non-delivery notification: 7537CC90A92 Continue reading Elastix Voicemail to Email not working