[Asterisk-Users] Asterisk MWI and Realtime
I'm testing my asterisk system and the realtime backend. My Asterisk build is rather aged, 03/18/2005 CVS. I have successfully moved Sip peers and Voicemail boxes to the realtime database backend and this works very well except for MWI. I don't seem to be able to get MWI to work when I store the voicemail information in a database backend, from a flat file it does work fine. I'm using rtcachefriends=yes for my sip users, per the WIKI, I'm presuming asterisk can't see these mailboxes, and therefore can't poll them to send the alerts when necessary. Is there anything that can be done to make this work properly, short of going back to a flat file for voicemail.conf? Regards Michael Baird ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MWI and Realtime
Michael Baird wrote: I'm testing my asterisk system and the realtime backend. My Asterisk build is rather aged, 03/18/2005 CVS. I have successfully moved Sip peers and Voicemail boxes to the realtime database backend and this works very well except for MWI. I don't seem to be able to get MWI to work when I store the voicemail information in a database backend, from a flat file it does work fine. I'm using rtcachefriends=yes for my sip users, per the WIKI, I'm presuming asterisk can't see these mailboxes, and therefore can't poll them to send the alerts when necessary. Is there anything that can be done to make this work properly, short of going back to a flat file for voicemail.conf? Regards Michael Baird We have been using RealTime and Voicemail for quite some time and have no problems getting MWI's. If your source is that old, it might be why the config option isn't working. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MWI and Realtime
On Fri, 2005-08-05 at 14:06 -0500, Matthew Boehm wrote: Michael Baird wrote: I'm testing my asterisk system and the realtime backend. My Asterisk build is rather aged, 03/18/2005 CVS. I have successfully moved Sip peers and Voicemail boxes to the realtime database backend and this works very well except for MWI. I don't seem to be able to get MWI to work when I store the voicemail information in a database backend, from a flat file it does work fine. I'm using rtcachefriends=yes for my sip users, per the WIKI, I'm presuming asterisk can't see these mailboxes, and therefore can't poll them to send the alerts when necessary. Is there anything that can be done to make this work properly, short of going back to a flat file for voicemail.conf? Regards Michael Baird We have been using RealTime and Voicemail for quite some time and have no problems getting MWI's. If your source is that old, it might be why the config option isn't working. -Matthew Well, ok, the WIKI said anything newer then 03/16/2005 should work with the rtcachefriends=yes option. It is definitely working, since sip show peers also shows, my dynamicly configured sip clients. What would a stutter tone signal look like in the debug log? I have verbose logging turned on there. I do get the Sip Notify when a message is left, just no stutter tone when picking up the phone. I will also refresh my build as well, hopefully no big changes. Regards Michael Baird ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MWI and Realtime
turned on there. I do get the Sip Notify when a message is left, just no stutter tone when picking up the phone. I will also refresh my build as well, hopefully no big changes. The stutter tone will be a phone/ATA specific issue not related to Asterisk. For instance, I know the Linksys PAP2-NA's give you the option of stutter tone, or quick ring, etc..when there are voicemails. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk MWI and Realtime
He's right. My company uses Linksys PAP2s, and yes they do have stutter tone and short ring options in the box. We also use MWI, and have had no issues with it at all --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Matthew Boehm -Sent: Friday, August 05, 2005 4:43 PM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] Asterisk MWI and Realtime - - turned on there. I do get the Sip Notify when a message is -left, just - no stutter tone when picking up the phone. I will also refresh my - build as well, hopefully no big changes. - - The stutter tone will be a phone/ATA specific issue not -related to Asterisk. - - For instance, I know the Linksys PAP2-NA's give you the -option of stutter tone, or quick ring, etc..when there are voicemails. - --Matthew - -___ -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MWI and Realtime
Ok, thanks, I tried it on my device again (Zoom X5v), and it is not working via the flat file either. I'll inform Zoom engineering and see what they say, it could have been broken in a firmware update. So I can explain it properly, am I correct in thinking that asterisk only sends the Sip Notify command to the ATA, then the ATA actually provides the stutter tone/short rings, to notify the user they have voicemail waiting. I have notified zoom in the past about their device not recognizing the Sip Notify, asterisk notes it as being unsupported. I have several other ATA's, I will try them to verify asterisk is working properly, but with your input, I think the ATA is the issue now. Regards Michael Baird turned on there. I do get the Sip Notify when a message is left, just no stutter tone when picking up the phone. I will also refresh my build as well, hopefully no big changes. The stutter tone will be a phone/ATA specific issue not related to Asterisk. For instance, I know the Linksys PAP2-NA's give you the option of stutter tone, or quick ring, etc..when there are voicemails. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
