On Fri, Nov 12, 2010 at 7:52 AM, Brett Woollum <[email protected]> wrote: > More information: When I have "rtcachefriends = yes" in sip.conf, > everything seems fine. With "rtcachefriends = no" I see this behavior. > > I'd rather not cache. I'm aiming for as near real-time as possible. > > Any thoughts? > > Brett Woollum > [email protected] > > > ----- Original Message ----- > From: "Brett Woollum" <[email protected]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Sent: Friday, November 12, 2010 5:34:03 AM GMT -08:00 US/Canada Pacific > Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 > Realtime ODBC Tables > > Hi Brad, > > I did notice that bug in the bug tracker. That's different from the behavior > I am seeing. I don't get multiple values in the "Mailbox". I just upgraded > to 1.6.2.14 and it's still there. > > By the way, the quantity of SIP NOTIFY's generated is significant. It > appears to be way more that the number of peers I have (3) times a handful > of duplicates per peer. I've been doing a Wireshark capture, and it appears > as though any time there is a new message in the ODBC voicemail store for a > mailbox that has been subscribed to, Asterisk continually generates as many > of the messages as possible. At one point I noticed my CPU jump from 0% to > ~50% just by moving one message from an mailbox that hadn't been subscribed > to to a mailbox that was subscribed to by the 3 peers. It only came back to > ~0-1% by moving the message back to an unsubscribed user. > > When I set rtcachefriends = yes in sip.conf, I get the following for each > peer: > > ast01*CLI> sip show peer 412 > > > * Name : 412 > Realtime peer: Yes, cached > Secret : <Set> > MD5Secret : <Not set> > Remote Secret: <Not set> > Context : sipphones > Subscr.Cont. : blf_subscriptions > Language : en > AMA flags : Unknown > Transfer mode: open > CallingPres : Presentation Allowed, Not Screened > Callgroup : > Pickupgroup : > Mailbox : vm_...@default > VM Extension : asterisk > LastMsgsSent : 32767/65535 > Call limit : 0 > Dynamic : Yes > Callerid : "" <> > MaxCallBR : 384 kbps > Expire : 69 > Insecure : no > Nat : RFC3581 > ACL : No > T.38 support : No > T.38 EC mode : Unknown > T.38 MaxDtgrm: -1 > DirectMedia : Yes > PromiscRedir : No > User=Phone : No > Video Support: No > Text Support : No > Ign SDP ver : No > Trust RPID : No > Send RPID : No > Subscriptions: Yes > Overlap dial : Yes > Forward Loop : Yes > DTMFmode : rfc2833 > Timer T1 : 500 > Timer B : 32000 > ToHost : > Addr->IP : 10.20.1.225 Port 5064 > Defaddr->IP : 0.0.0.0 Port 5060 > Prim.Transp. : UDP > Allowed.Trsp : UDP > Def. Username: 412 > SIP Options : (none) > Codecs : 0x1004 (ulaw|g722) > Codec Order : (g722:20,ulaw:20) > Auto-Framing : No > 100 on REG : Yes > Status : Unmonitored > Useragent : Yealink SIP-T28P 2.50.0.52 > Reg. Contact : sip:[email protected]:5064 > Qualify Freq : 120000 ms > Sess-Timers : Accept > Sess-Refresh : uas > Sess-Expires : 1800 secs > Min-Sess : 90 secs > Parkinglot : > > This is Asterisk 1.6.2.14 using the ODBC store for voicemail and ODBC for > sip_peers. > > Brett Woollum > [email protected] > > > ----- Original Message ----- > From: "Bradley Watkins" <[email protected]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Sent: Friday, November 12, 2010 5:14:49 AM GMT -08:00 US/Canada Pacific > Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 > Realtime ODBC Tables > > > >>-----Original Message----- >>From: [email protected] >>[mailto:[email protected]] On Behalf Of >>Paul Belanger >>Sent: Friday, November 12, 2010 7:58 AM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: Re: [asterisk-users] Official Documentation for >>Asterisk 1.6 Realtime ODBC Tables >> >>On Fri, Nov 12, 2010 at 6:07 AM, Brett Woollum >><[email protected]> wrote: >>> I'm having an issue where Asterisk continuously sends out a >>GAZILLION >>> "SIP NOTIFY" messages when a user has a voice message in >>their INBOX. >>> This issue is only present when my SIP users and peers are >>configured >>> from my ODBC backend (MySQL). A static configuration of users in >>> sip.conf resolves this and everything works fine. >>> >>What version of 1.6? I _think_ this may have been a bug, that >>was fixed. >> >>Don't hold me to that. > > I agree with Paul, this sounds like a bugs that's been fixed. > > What does the 'Mailbox :' line look like when you do a 'sip show peers'? > > My guess is that there will be multiple entries of the same mailbox, and > that's why you're receiving a bunch of NOTIFY messages. > > - Brad > > -- > _____________________________________________________________________ > -- 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 > -- > _____________________________________________________________________ > -- 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 >
That's the problem, you've got rtcache friends turned off. If full realtime is that important, modify whatever scripts you have that make updates to your sip accounts to run "asterisk -rx 'sip prune realtime peer PEERNAME' " and then "asterisk -rx 'sip show peer PEERNAME load' " after it makes the update to the sip table. That clears Asterisk's cache for the modified sip peer and then loads the information from the database. Technically, I believe you might be able to get away with not clearing the cached info, but I've always played it safe. Cheers, Sherwood McGowan A LOOOOONG Time user of all things Asterisk Realtime -- _____________________________________________________________________ -- 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
