<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Arial, Helvetica, sans-serif; font-size: 12pt; color: #000000'>Hi Sherwood,<br><br>Thanks for the reply. That's interesting to me. What is the point of rtcachefriends = no if it causes weird things like this to happen? <br><br>As mentioned, I'd like to stay real-time and fully database driven for everything. Not only does it make life easier in terms of changing settings on the system (without reloads!), but it will make scaling the system to more Asterisk servers much easier. Is there a way for Asterisk to automatically look up the sip user or peer's information from the ODBC backend every time and work properly? It seems to be doing that with rtcachefriends = no, with the exception of the MWI subsystem. How can I retain the database driven behavior of rtcachefriends = yes, but still keep the MWI working?<br><br>Also, the BLF subscriptions and subsequent NOTIFY's are working fine. A capture of the wire by the phone shows the only issue as being the NOTIFY's for MWI. <br><br>Thanks!<br><br><div><span style="font-family: arial,helvetica,sans-serif;">Brett Woollum</span><br style="font-family: arial,helvetica,sans-serif;"><span style="font-family: arial,helvetica,sans-serif;">Brett@Woollum.com</span><br></div><br><br>----- Original Message -----<br>From: "Sherwood McGowan" <sherwood.mcgowan@gmail.com><br>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br>Sent: Friday, November 12, 2010 7:36:22 AM GMT -08:00 US/Canada Pacific<br>Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables<br><br>On Fri, Nov 12, 2010 at 7:52 AM, Brett Woollum <brett@woollum.com> wrote:<br>> More information: When I have "rtcachefriends = yes" in sip.conf,<br>> everything seems fine. With "rtcachefriends = no" I see this behavior.<br>><br>> I'd rather not cache. I'm aiming for as near real-time as possible.<br>><br>> Any thoughts?<br>><br>> Brett Woollum<br>> Brett@Woollum.com<br>><br>><br>> ----- Original Message -----<br>> From: "Brett Woollum" <brett@woollum.com><br>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br>> <asterisk-users@lists.digium.com><br>> Sent: Friday, November 12, 2010 5:34:03 AM GMT -08:00 US/Canada Pacific<br>> Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6<br>> Realtime ODBC Tables<br>><br>> Hi Brad,<br>><br>> I did notice that bug in the bug tracker. That's different from the behavior<br>> I am seeing. I don't get multiple values in the "Mailbox". I just upgraded<br>> to 1.6.2.14 and it's still there.<br>><br>> By the way, the quantity of SIP NOTIFY's generated is significant. It<br>> appears to be way more that the number of peers I have (3) times a handful<br>> of duplicates per peer. I've been doing a Wireshark capture, and it appears<br>> as though any time there is a new message in the ODBC voicemail store for a<br>> mailbox that has been subscribed to, Asterisk continually generates as many<br>> of the messages as possible. At one point I noticed my CPU jump from 0% to<br>> ~50% just by moving one message from an mailbox that hadn't been subscribed<br>> to to a mailbox that was subscribed to by the 3 peers. It only came back to<br>> ~0-1% by moving the message back to an unsubscribed user.<br>><br>> When I set rtcachefriends = yes in sip.conf, I get the following for each<br>> peer:<br>><br>> ast01*CLI> sip show peer 412<br>><br>><br>> * Name : 412<br>> Realtime peer: Yes, cached<br>> Secret : <Set><br>> MD5Secret : <Not set><br>> Remote Secret: <Not set><br>> Context : sipphones<br>> Subscr.Cont. : blf_subscriptions<br>> Language : en<br>> AMA flags : Unknown<br>> Transfer mode: open<br>> CallingPres : Presentation Allowed, Not Screened<br>> Callgroup :<br>> Pickupgroup :<br>> Mailbox : vm_bob@default<br>> VM Extension : asterisk<br>> LastMsgsSent : 32767/65535<br>> Call limit : 0<br>> Dynamic : Yes<br>> Callerid : "" <><br>> MaxCallBR : 384 kbps<br>> Expire : 69<br>> Insecure : no<br>> Nat : RFC3581<br>> ACL : No<br>> T.38 support : No<br>> T.38 EC mode : Unknown<br>> T.38 MaxDtgrm: -1<br>> DirectMedia : Yes<br>> PromiscRedir : No<br>> User=Phone : No<br>> Video Support: No<br>> Text Support : No<br>> Ign SDP ver : No<br>> Trust RPID : No<br>> Send RPID : No<br>> Subscriptions: Yes<br>> Overlap dial : Yes<br>> Forward Loop : Yes<br>> DTMFmode : rfc2833<br>> Timer T1 : 500<br>> Timer B : 32000<br>> ToHost :<br>> Addr->IP : 10.20.1.225 Port 5064<br>> Defaddr->IP : 0.0.0.0 Port 5060<br>> Prim.Transp. : UDP<br>> Allowed.Trsp : UDP<br>> Def. Username: 412<br>> SIP Options : (none)<br>> Codecs : 0x1004 (ulaw|g722)<br>> Codec Order : (g722:20,ulaw:20)<br>> Auto-Framing : No<br>> 100 on REG : Yes<br>> Status : Unmonitored<br>> Useragent : Yealink SIP-T28P 2.50.0.52<br>> Reg. Contact : sip:412@10.20.1.225:5064<br>> Qualify Freq : 120000 ms<br>> Sess-Timers : Accept<br>> Sess-Refresh : uas<br>> Sess-Expires : 1800 secs<br>> Min-Sess : 90 secs<br>> Parkinglot :<br>><br>> This is Asterisk 1.6.2.14 using the ODBC store for voicemail and ODBC for<br>> sip_peers.<br>><br>> Brett Woollum<br>> Brett@Woollum.com<br>><br>><br>> ----- Original Message -----<br>> From: "Bradley Watkins" <Bradley.Watkins@compuware.com><br>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br>> <asterisk-users@lists.digium.com><br>> Sent: Friday, November 12, 2010 5:14:49 AM GMT -08:00 US/Canada Pacific<br>> Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6<br>> Realtime ODBC Tables<br>><br>><br>><br>>>-----Original Message-----<br>>>From: asterisk-users-bounces@lists.digium.com<br>>>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of<br>>>Paul Belanger<br>>>Sent: Friday, November 12, 2010 7:58 AM<br>>>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>>>Subject: Re: [asterisk-users] Official Documentation for<br>>>Asterisk 1.6 Realtime ODBC Tables<br>>><br>>>On Fri, Nov 12, 2010 at 6:07 AM, Brett Woollum<br>>><brett@woollum.com> wrote:<br>>>> I'm having an issue where Asterisk continuously sends out a<br>>>GAZILLION<br>>>> "SIP NOTIFY" messages when a user has a voice message in<br>>>their INBOX.<br>>>> This issue is only present when my SIP users and peers are<br>>>configured<br>>>> from my ODBC backend (MySQL). A static configuration of users in<br>>>> sip.conf resolves this and everything works fine.<br>>>><br>>>What version of 1.6? I _think_ this may have been a bug, that<br>>>was fixed.<br>>><br>>>Don't hold me to that.<br>><br>> I agree with Paul, this sounds like a bugs that's been fixed.<br>><br>> What does the 'Mailbox :' line look like when you do a 'sip show peers'?<br>><br>> My guess is that there will be multiple entries of the same mailbox, and<br>> that's why you're receiving a bunch of NOTIFY messages.<br>><br>> - Brad<br>><br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>><br>> -- _____________________________________________________________________ --<br>> Bandwidth and Colocation Provided by http://www.api-digital.com -- New to<br>> Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or<br>> update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>><br><br>That's the problem, you've got rtcache friends turned off. If full<br>realtime is that important, modify whatever scripts you have that make<br>updates to your sip accounts to run "asterisk -rx 'sip prune realtime<br>peer PEERNAME' " and then "asterisk -rx 'sip show peer PEERNAME load'<br>" after it makes the update to the sip table. That clears Asterisk's<br>cache for the modified sip peer and then loads the information from<br>the database. Technically, I believe you might be able to get away<br>with not clearing the cached info, but I've always played it safe.<br><br>Cheers,<br>Sherwood McGowan<br>A LOOOOONG Time user of all things Asterisk Realtime<br><br>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> http://www.asterisk.org/hello<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users</div></body></html>