[Serusers] SER + Asterisk Voicemail -- Seeking Advise

Java Rockx javarockx at yahoo.com
Sun Sep 26 04:29:50 CEST 2004


Great idea! Asterisk stores user mailboxs under
/var/spool/asterisk/voicemail/DOMAIN.COM/

Each user gets their own directory. So user 1000 will have a directory 1000. I
bet I can symlink /var/spool/asterisk/voicemail/DOMAIN.COM/1000 to
/var/spool/asterisk/voicemail/DOMAIN.COM/4075551234

I'll give it a test Monday since I'm preparing for hurricane Jeanne this
weekend.

Thanks,
Paul

--- Joseph Cheung <Joseph.Cheung2 at TELUS.COM> wrote:

> Don't know much about the asterisk voicemail capability. However, does the
> voicemail has "aliases" capability in terms it could take multiple phone
> number for the same mailbox?
> 
> -----Original Message-----
> From: serusers-bounces at iptel.org [mailto:serusers-bounces at lists.iptel.org] On
> Behalf Of Java Rockx
> Sent: September 25, 2004 9:53 PM
> To: ser users
> Subject: [Serusers] SER + Asterisk Voicemail -- Seeking Advise 
> 
> Hello All.
> 
> I finally have my lookup("aliases") working thanks to Zeus Ng. Now that users
> can have aliases on my ser proxy I have a question regarding voicemail. I'm
> hoping someone can give me an idea of how best to address this issue.
> 
> I use ser for all SIP stuff and Asterisk for voicemail only. I have ser and
> asterisk working nicely together.
> 
> A typical scenerio would be like this. I have a ser user named
> 1000 at mycompany.com with a PSTN alias 4075551234. In addition this user has an
> Asterisk mailbox configured as 1000 at mycompany.com
> 
> When someone dials sip:1000 at mycompany.com and there is no answer they will
> get
> sent to voicemail, which then Asterisk will say "The user at extension
> one-zero-zero-zero is unavailable. Please leave your message after the
> tone..."
> 
> But what happens when a caller dials 4075551234 at mycompany.com and gets routed
> to voicemail? 4075551234 doesn't exist in asterisk. If I use
> lookup("aliases")
> in my ser.cfg routing plan can I revert back to the original
> sip:1000 at mycompany.com before sending the caller off to the asterisk
> voicemail?
> If so, how?
> 
> Regards,
> Paul
> 
> 
> 		
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