From my experience prune does not take effect without reload. And after reload ALL your phones are unreachable for 2 minutes!
Imagine you have several thousands devices unreachable for 2 minutes. How much calls will fail during that time? Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Tuesday, March 02, 2010 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] rtcachefriends & qualify & sip reload On Tue, 2010-03-02 at 15:59 +0100, jonas kellens wrote: > On Tue, 2010-03-02 at 11:32 +0000, Ishfaq Malik wrote: > > If you are changing RealTime config in your DB you need to do a sip > > prune realtime either directly from asterisk cli or using AMI. You > > really do not need to do a SIP reload when changing the config of > > one sip extension. > I notice that after a "sip prune realtime all" I also loose all of my > realtime sip peers. Same result actually as with "sip reload". > > I close the softphone of gerrie2 (becomes unspecified) > > asterisk*CLI> sip show peers > Name/username Host Dyn Nat ACL Port Status > Realtime > gerrie005/gerrie005 192.168.1.106 D N 5060 OK > (4 ms) Cached RT > gerrie002/gerrie002 (Unspecified) D N 0 > UNKNOWN Cached RT > gerrie001/gerrie001 192.168.1.105 D N 5060 OK > (11 ms) Cached RT > > I prune the realtime peers to no longer have gerrie002 in cache : > > asterisk*CLI> sip prune realtime all > 3 peers pruned. > 2 users pruned. > [Mar 2 15:42:19] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: > Peer 'gerrie001' is now UNREACHABLE! Last qualify: 91 > > The realtime peers are all gone : > > asterisk*CLI> sip show peers > Name/username Host Dyn Nat ACL Port Status > Realtime > > Internal call fails : > > [Mar 2 15:46:38] WARNING[558]: app_dial.c:1272 dial_exec_full: Unable > to create channel of type 'SIP' (cause 20 - Unknown) > [Mar 2 15:46:38] == Everyone is busy/congested at this time > (1:0/0/1) > [Mar 2 15:46:38] == Auto fallthrough, channel > 'SIP/gerrie001-09f631e0' status is 'CHANUNAVAIL' > > I re-register 2 softphones (gerrie001 & gerrie005) : > > asterisk*CLI> sip show peers > Name/username Host Dyn Nat ACL Port Status > Realtime > gerrie002/gerrie002 (Unspecified) D N 0 > UNREACHABLE Cached RT > gerrie001/gerrie001 192.168.1.105 D N 5060 OK > (11 ms) Cached RT > gerrie005/gerrie005 192.168.1.106 D N 5060 OK > (7 ms) Cached RT > > The SIP-peer 'gerrie002' is still in the cache ! Don't know where this > is coming from ?? > > I prune again : > > asterisk*CLI> sip prune realtime all > 3 peers pruned. > 1 users pruned. > [Mar 2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: > Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 [Mar 2 > 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: > Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 [Mar 2 > 15:52:01] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: > Peer 'gerrie001' is now UNREACHABLE! Last qualify: 11 > > And again no more peers until I re-register : > > asterisk*CLI> sip show peers > Name/username Host Dyn Nat ACL Port Status > Realtime > > > This realtime thing isn't really working out here... What exactly do I > need to do to clear the cache and thus the old SIP-peers so they can > no longer be used ?? > Do not prune all peers, only the peer you wish to reload or eliminate! Do "sip prune realtime peer peername". That way you do not lose all the other registrations. I really do not see this as a problem as the phones will usually re register quickly or if the user dials any number. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _____________________________________________________________________ -- 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
