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Email notification of VoiceMail

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bubbaphone

Technical User
Aug 4, 2003
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Is there a step-by-step guide to configuring Email notification of Voice Mail on the Asterisk? I have looked at dozens of threads on many web sites, and none seem to have a complete guide to setting it up..i.e which files must be edited, and with what entries.
Some mention files, but don't tell you what goes in it.
I'm about ready to pitch this thing out the window...

 
You don't need any specific configuration unless you use a smarthost to relay the e-mail. Is this what you are planning? If the server can resolve DNS names then it should be able to resolve MX records and send e-mails to any domain. Have a look in /etc/resolv.conf to see if you have any name server setup in their. Something like this should be here:

nameserver 10.10.1.1

You also need a valid configuration for each mailbox in /etc/asterisk/voicemail.conf. Each mailbox should have any entry something like this:

100=>1234,Office,blobby@blob.com,,attach=no|saycid=no|envelope=no|delete=no



Lee Mason
Optimal Projects Ltd
 
ok I have the same issue. I have the following in my resolv.conf file

nameserver 192.168.1.20
nameserver 192.168.1.7


and my voicemail.conf email line is as follows:

1124 => 1234,Brian Simpson,brian.simpson@docomointertouch.com,,|attach=yes|tz=eastern|saycid=no|dialout=fromvm|review=yes|operator=yes|envelope=no|sayduration=yes|saydurationm=1

If i point change my email to an internal email server it works fine. This email address is hosted by another email server and I never get teh notification that I have a voicemail there.

Any ideas?

Brian




To error is human.....if the machine doesnt work, then KICK IT !!!!!!!!!!!!!
 
Mine finally decided to work...After many frustrating days of changing and testing. It ended up being a DNS server issue.
I must say that the documentation on this product from Digium is very poor in lots of areas, and their web site FAQ was absolutely NO help.

Any idea why the "include=default" statement does not work in the dial plan? I have to manually build around the VM , Internal Numbers, and Conf Rooms in the dial plan. I finally built a bogus SIP trunk and routed the internal numbers to it so I could keep my dial plan simple as possible.
 
Thanks for all of your help. I found that the firewall was blocking port 25 from the asterisk server. I had the system admin implement new rules for the server to this port and everything is working fine.

Thanks Once Again

Brian

To error is human.....if the machine doesnt work, then KICK IT !!!!!!!!!!!!!
 
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