[asterisk-users] rtcachefriends & qualify & sip reload

[Mindaugas Kezys]

Sip reload

 

Regards,

Mindaugas Kezys

 

Kolmisoft UAB 

VoIP Billing Solutions

e-mail:  <mailto:info at kolmisoft.com> info at kolmisoft.com

URL:  <http://www.kolmisoft.com> http://www.kolmisoft.com

 

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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 +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 ??

Jonas. 

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