[asterisk-users] rtcachefriends & qualify & sip reload

Ishfaq Malik ish at pack-net.co.uk
Wed Mar 3 03:45:48 CST 2010


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: info at kolmisoft.com
> URL: http://www.kolmisoft.com
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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
>
>
>   

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062



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