How to configure Reliance Jio SIP trunk on asterisk
Reliance Jio provide SIP E1 trunks with DIDs . For old PBX they give you SIP to E1 converter which give you RJ45 connector to connect your E1 port.
Jio put a fiber cable and terminate on ONT from there they provide a network CAT 6 cable to put in to your network switch or directly to your IP PBX network interface not E1 interface .
Reliance jio provide SIP E1 trunk on there own IP address which will be configured on your asterisk server or Grandstream UCM or any IP IPBX .
Goautodial 3.0 Getting Started Guide Requirements 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 … Read more
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.
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