I'm bone stock except for the following global(s). rtcachefriends=yes limitonpeers=yes allowsubscribe=yes notifyringing=yes notifyhold=yes notifybusy=yes My Sip Table looks like (Sorry, Hard to read, Note the 207@103 mailbox setting. 90 103_14 "Nick" <321XXXXXXX> 103_14 SECRETPASS 103_internal no rfc2833 dynamic port,invite 207@103 force_rport,comedia yes 900 1300 friend all ulaw 0 yes 0 207 0 And my voicemail table (Hard to read as well) 98 207 103 207 PASS Nick central no yes no no no no 1 no no yes no no yes 2015-08-17 15:22:09 yes no 4 25 Forcing a reload from the DB with the Prune>Load method loads the mailbox about 20% of the time Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 1/0 Call limit : 0 Max forwards : 0 Dynamic : Yes Callerid : "Nick" <321XXXXXXX> Nick Olsen Network Operations (855) FLSPEED x106
---------------------------------------- From: "Stefan Tichy" <asteri...@pi4tel.de> Sent: Sunday, September 20, 2015 9:28 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime Voicemail MWI On Wed, Sep 16, 2015 at 04:44:45PM -0400, Nick Olsen wrote: > I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, > These are loaded into asterisk without the mailbox info. Leading to > "Received SIP subscribe for peer without mailbox" notices. And non-working > MWI. > > Occasionally, It just works. But only on a peer or two at a time. And > it'll stop working after a few minutes. Here it seems to be the other way round. Occasionally I see that peers have lost there mailbox setting and don't get notify messages with voicemail information. It is Asterisk 13.5.0 "sip prune realtime peer ..." "sip show peer ... load" After this the setting is restored, but until now I have no idea why this happens. The database field mailbox remains unchanged. Could you post the Realtime SIP Settings? -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users