Re: [asterisk-users] rtcachefriends & qualify & sip reload
Hi We run production servers for various customers all using realtime with web interfaces so they can change their own config whenever they want. Prune works fine for us and we never do sip reloads (1.4.17) Ish Mindaugas Kezys wrote: > 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 +, 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 HostDyn Nat ACL Port Status >> Realtime >> gerrie005/gerrie005192.168.1.106D N 5060 OK >> (4 ms) Cached RT >> gerrie002/gerrie002(Unspecified)D N 0 >> UNKNOWNCached RT >> gerrie001/gerrie001192.168.1.105D 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 HostDyn 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 HostDyn Nat ACL Port Status >> Realtime >> gerrie002/gerrie002(Unspecified)D N 0 >> UNREACHABLE Cached RT >> gerrie001/gerrie001192.168.1.105D N 5060 OK >> (11 ms) Cached RT >> gerrie005/gerrie005192.168.1.106D 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 HostDyn 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 "si
Re: [asterisk-users] rtcachefriends & qualify & sip reload
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 +, 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 HostDyn Nat ACL Port Status > Realtime > gerrie005/gerrie005192.168.1.106D N 5060 OK > (4 ms) Cached RT > gerrie002/gerrie002(Unspecified)D N 0 > UNKNOWNCached RT > gerrie001/gerrie001192.168.1.105D 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 HostDyn 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 HostDyn Nat ACL Port Status > Realtime > gerrie002/gerrie002(Unspecified)D N 0 > UNREACHABLE Cached RT > gerrie001/gerrie001192.168.1.105D N 5060 OK > (11 ms) Cached RT > gerrie005/gerrie005192.168.1.106D 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 HostDyn 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
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 +, 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 HostDyn Nat ACL Port Status > Realtime > gerrie005/gerrie005192.168.1.106D N 5060 OK > (4 ms) Cached RT > gerrie002/gerrie002(Unspecified)D N 0 > UNKNOWNCached RT > gerrie001/gerrie001192.168.1.105D 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 HostDyn 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 HostDyn Nat ACL Port Status > Realtime > gerrie002/gerrie002(Unspecified)D N 0 > UNREACHABLE Cached RT > gerrie001/gerrie001192.168.1.105D N 5060 OK > (11 ms) Cached RT > gerrie005/gerrie005192.168.1.106D 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 HostDyn 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 signature.asc Description: This is a digitally signed message part -- _ -- 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
Re: [asterisk-users] rtcachefriends & qualify & sip reload
jonas kellens wrote: > On Tue, 2010-03-02 at 11:32 +, 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 HostDyn Nat ACL Port > Status Realtime > gerrie005/gerrie005192.168.1.106D N 5060 OK > (4 ms) Cached RT > gerrie002/gerrie002(Unspecified)D N 0 > UNKNOWNCached RT > gerrie001/gerrie001192.168.1.105D 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 you're doing the wrong thing! If you want to get rid of just gerrie002 you need to do sip prune realtime gerrie002 That will clear only gerrie002 from the realtime cache and leave the others alone. > > The realtime peers are all gone : > > asterisk*CLI> sip show peers > Name/username HostDyn 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 HostDyn Nat ACL Port > Status Realtime > gerrie002/gerrie002(Unspecified)D N 0 > UNREACHABLE Cached RT > gerrie001/gerrie001192.168.1.105D N 5060 OK > (11 ms) Cached RT > gerrie005/gerrie005192.168.1.106D 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 HostDyn 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 ?? > > Jonas. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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
Re: [asterisk-users] rtcachefriends & qualify & sip reload
Sip reload Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: <mailto:i...@kolmisoft.com> i...@kolmisoft.com URL: <http://www.kolmisoft.com> http://www.kolmisoft.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Tuesday, March 02, 2010 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] rtcachefriends & qualify & sip reload On Tue, 2010-03-02 at 11:32 +, 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 HostDyn Nat ACL Port Status Realtime gerrie005/gerrie005192.168.1.106D N 5060 OK (4 ms) Cached RT gerrie002/gerrie002(Unspecified)D N 0UNKNOWN Cached RT gerrie001/gerrie001192.168.1.105D 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 HostDyn 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 HostDyn Nat ACL Port Status Realtime gerrie002/gerrie002(Unspecified)D N 0UNREACHABLE Cached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (11 ms) Cached RT gerrie005/gerrie005192.168.1.106D 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 HostDyn 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 ?? Jonas. -- _ -- 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
Re: [asterisk-users] rtcachefriends & qualify & sip reload
On Tue, 2010-03-02 at 11:32 +, 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 HostDyn Nat ACL Port Status Realtime gerrie005/gerrie005192.168.1.106D N 5060 OK (4 ms) Cached RT gerrie002/gerrie002(Unspecified)D N 0 UNKNOWNCached RT gerrie001/gerrie001192.168.1.105D 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 HostDyn 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 HostDyn Nat ACL Port Status Realtime gerrie002/gerrie002(Unspecified)D N 0 UNREACHABLE Cached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (11 ms) Cached RT gerrie005/gerrie005192.168.1.106D 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 HostDyn 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 ?? Jonas. -- _ -- 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
Re: [asterisk-users] rtcachefriends & qualify & sip reload
jonas kellens wrote: > On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: >> In my experience, yes, that is normal behaviour. Generally any SIP phone >> will try to reconnect with the server within 2 mins anyway. >> > In the Zoiper softphone, it is set to 3600 seconds... I don't want my > customers have to do a lot of configuration on their softphone. > Can I force the SIP clients to re-register every 5 minutes (a setting > in sip.conf ?) ?? Will this cause a lot of overhead ? If you get the AMI part working you will no longer need to do SIP reloads and this becomes academic. >> 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'm using a php-webGUI to change the sip_buddies table. Is their an > easy php class that facilitates working with AMI (as I have no > experience with AMI) ? Hi, Have a look at this page http://www.voip-info.org/wiki/view/Asterisk+manager+API Further down the page it has links to using the AMI with different programming languages including example classes. In the end I ended up writing my own class as I was only doing 3 or 4 things with the AMI and didn't need the rest. All you are doing is opening a socket on the asterisk machine and writing to and reading from it so it's not exactly rocket science. I went through exactly the problems you are having myself a fair few months ago. It took me a day or 2 to get it all sorted out in my test environment so it's not too difficult to implement. > > > Greetingz, > Jonas. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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
Re: [asterisk-users] rtcachefriends & qualify & sip reload
On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: > In my experience, yes, that is normal behaviour. Generally any SIP phone > will try to reconnect with the server within 2 mins anyway. In the Zoiper softphone, it is set to 3600 seconds... I don't want my customers have to do a lot of configuration on their softphone. Can I force the SIP clients to re-register every 5 minutes (a setting in sip.conf ?) ?? Will this cause a lot of overhead ? > 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'm using a php-webGUI to change the sip_buddies table. Is their an easy php class that facilitates working with AMI (as I have no experience with AMI) ? Greetingz, Jonas. -- _ -- 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
Re: [asterisk-users] rtcachefriends & qualify & sip reload
jonas kellens wrote: > I'd like to add to my thread that realtime SIP peers do not seem to be > surviving a "sip reload". > > step 1 : 2 realtime SIP peers are registered to Asterisk, they can > make a phone call to each other. > step 2 : I do a 'sip reload' > step 3 : the 2 realtime SIP peers are no longer able to phone to each > other > > [Mar 2 11:32:41] WARNING[32668]: app_dial.c:1272 dial_exec_full: > Unable to create channel of type 'SIP' (cause 20 - Unknown) > [Mar 2 11:32:41] == Everyone is busy/congested at this time (1:0/0/1) > [Mar 2 11:32:41] == Auto fallthrough, channel > 'SIP/gerrie001-09ed70d0' status is 'CHANUNAVAIL' > > I look at the mysql-table 'sip_buddies' and the values for 'ipaddr' > and 'port' are still filled in and correct. > > When executing 'sip show peers', the realtime peers also have disappeared. > At first there was : > Name/username HostDyn Nat ACL Port > Status Realtime > gerrie002/gerrie002192.168.1.104D N 5060 OK > (10 ms) Cached RT > gerrie001/gerrie001192.168.1.105D N 5060 OK > (30 ms) Cached RT > > Now there is : > Name/username HostDyn Nat ACL Port > Status Realtime > gerrie002/gerrie002192.168.1.104 D N 5060 > UNREACHABLE Cached RT > > Using Zoiper softphone, the SIP-accounts still show status 'registered'. > > Re-registering is the only thing that helps : > Name/username HostDyn Nat ACL Port > Status Realtime > gerrie001/gerrie001192.168.1.105D N 5060 OK > (9 ms) Cached RT > gerrie002(Unspecified)D N 0 > UNREACHABLE Cached RT > > And for account 2 : > Name/username HostDyn Nat ACL Port > Status Realtime > gerrie002/gerrie002192.168.1.104D N 5060 OK > (6 ms) Cached RT > gerrie001/gerrie001192.168.1.105D N 5060 OK > (9 ms) Cached RT > > In the mysql-DB, the field 'regseconds' turns from zero to some large > integer... > > I can reproduce the above very easy by just initiating 'sip reload'... > > Is this behaviour normal ?? > > Jonas. Hi In my experience, yes, that is normal behaviour. Generally any SIP phone will try to reconnect with the server within 2 mins anyway. 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. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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
Re: [asterisk-users] rtcachefriends & qualify
The problems we have with Asterisk Realtime: 1. After reload all registrations are void. 2. Without reload prune does not take effect. Test it in your scenario also. Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: i...@kolmisoft.com URL: http://www.kolmisoft.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Tuesday, March 02, 2010 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] rtcachefriends & qualify Thank you for your answer, Nic. It seems that by putting rtcachefriend=yes, the qualify works as expected and even changes made to my realtime MySQL-DB take affect immediately without the need of a reload (I changed the username and name). However the old username and name are still valuable and using this old SIP-user, one can still make outgoing calls. Receiving calls is no longer possible : WARNING[32439]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) Adding 'rtautoclear=yes' to sip.conf makes no difference. Changes to SIP-account are taken immediately, but the old SIP-credentials are still valid. (even after an unregister and re-register) Only after a "sip reload" I get the notice : [Mar 2 10:41:03] NOTICE[32498]: chan_sip.c:15889 handle_request_register: Registration from '"Gerrie"' failed for '192.168.1.105' - No matching peer found So a "sip reload" is always necessary to clear the cache ?? Jonas. On Mon, 2010-03-01 at 14:31 +, Nic Colledge wrote: Hi, I think so, maybe someone can help clarify this for me also. I have: rtcachefriends=yes rtautoclear=yes in sip.conf and was under the impression that this caches the settings from the database until a user unregisters. When they unregister the data is removed from the cache (rtautoclear). For me this was a nice compromise. This is from memory but I’m pretty sure I got this from the documentation online, if someone can confirm what I’m saying that would be sweet. Thanks. Nic. -- _ -- 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
Re: [asterisk-users] rtcachefriends & qualify & sip reload
I'd like to add to my thread that realtime SIP peers do not seem to be surviving a "sip reload". step 1 : 2 realtime SIP peers are registered to Asterisk, they can make a phone call to each other. step 2 : I do a 'sip reload' step 3 : the 2 realtime SIP peers are no longer able to phone to each other [Mar 2 11:32:41] WARNING[32668]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Mar 2 11:32:41] == Everyone is busy/congested at this time (1:0/0/1) [Mar 2 11:32:41] == Auto fallthrough, channel 'SIP/gerrie001-09ed70d0' status is 'CHANUNAVAIL' I look at the mysql-table 'sip_buddies' and the values for 'ipaddr' and 'port' are still filled in and correct. When executing 'sip show peers', the realtime peers also have disappeared. At first there was : Name/username HostDyn Nat ACL Port Status Realtime gerrie002/gerrie002192.168.1.104D N 5060 OK (10 ms) Cached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (30 ms) Cached RT Now there is : Name/username HostDyn Nat ACL Port Status Realtime gerrie002/gerrie002192.168.1.104 D N 5060 UNREACHABLE Cached RT Using Zoiper softphone, the SIP-accounts still show status 'registered'. Re-registering is the only thing that helps : Name/username HostDyn Nat ACL Port Status Realtime gerrie001/gerrie001192.168.1.105D N 5060 OK (9 ms) Cached RT gerrie002(Unspecified)D N 0 UNREACHABLE Cached RT And for account 2 : Name/username HostDyn Nat ACL Port Status Realtime gerrie002/gerrie002192.168.1.104D N 5060 OK (6 ms) Cached RT gerrie001/gerrie001192.168.1.105D N 5060 OK (9 ms) Cached RT In the mysql-DB, the field 'regseconds' turns from zero to some large integer... I can reproduce the above very easy by just initiating 'sip reload'... Is this behaviour normal ?? Jonas. -- _ -- 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
Re: [asterisk-users] rtcachefriends & qualify
Thank you for your answer, Nic. It seems that by putting rtcachefriend=yes, the qualify works as expected and even changes made to my realtime MySQL-DB take affect immediately without the need of a reload (I changed the username and name). However the old username and name are still valuable and using this old SIP-user, one can still make outgoing calls. Receiving calls is no longer possible : WARNING[32439]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) Adding 'rtautoclear=yes' to sip.conf makes no difference. Changes to SIP-account are taken immediately, but the old SIP-credentials are still valid. (even after an unregister and re-register) Only after a "sip reload" I get the notice : [Mar 2 10:41:03] NOTICE[32498]: chan_sip.c:15889 handle_request_register: Registration from '"Gerrie"' failed for '192.168.1.105' - No matching peer found So a "sip reload" is always necessary to clear the cache ?? Jonas. On Mon, 2010-03-01 at 14:31 +, Nic Colledge wrote: > Hi, > > > > I think so, maybe someone can help clarify this for me also. I have: > > rtcachefriends=yes > > rtautoclear=yes > > in sip.conf and was under the impression that this caches the settings > from the database until a user unregisters. When they unregister the > data is removed from the cache (rtautoclear). For me this was a nice > compromise. > > > > This is from memory but I’m pretty sure I got this from the > documentation online, if someone can confirm what I’m saying that > would be sweet. > > > > Thanks. > > Nic. -- _ -- 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
Re: [asterisk-users] rtcachefriends & qualify
Hi, I think so, maybe someone can help clarify this for me also. I have: rtcachefriends=yes rtautoclear=yes in sip.conf and was under the impression that this caches the settings from the database until a user unregisters. When they unregister the data is removed from the cache (rtautoclear). For me this was a nice compromise. This is from memory but I’m pretty sure I got this from the documentation online, if someone can confirm what I’m saying that would be sweet. Thanks. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: 01 March 2010 14:06 To: Asterisk Mailing Subject: [asterisk-users] rtcachefriends & qualify [Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'gerrie' Am I correct that when I turn on rtcachefriends in sip.conf, database-changes in my MySQL-DB will not be reflected untill a reload ?? Am I correct that when I turn off qualify in my realtime sip-database, I could be confronted with NAT-problems for SIP-peers that are behind a NAT-router ? Is this the choice I need to take ? Greetingz, Jonas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rtcachefriends & qualify
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'gerrie' Am I correct that when I turn on rtcachefriends in sip.conf, database-changes in my MySQL-DB will not be reflected untill a reload ?? Am I correct that when I turn off qualify in my realtime sip-database, I could be confronted with NAT-problems for SIP-peers that are behind a NAT-router ? Is this the choice I need to take ? Greetingz, Jonas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
