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@iptel.org [mailto:serusers-bounces@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@mycompany.com with a PSTN alias 4075551234. In addition this user has an Asterisk mailbox configured as 1000@mycompany.com
When someone dials sip:1000@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@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@mycompany.com before sending the caller off to the asterisk voicemail? If so, how?
Regards, Paul
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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@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@iptel.org [mailto:serusers-bounces@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@mycompany.com with a PSTN alias 4075551234. In addition this user has an Asterisk mailbox configured as 1000@mycompany.com
When someone dials sip:1000@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@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@mycompany.com before sending the caller off to the asterisk voicemail? If so, how?
Regards, Paul
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Java Rockx -
Joseph Cheung