-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP realtime
|working ?
|Importance: High
|
|So your Polycom 501's will eventually re-subscribe and BLF
|will eventually
|start working again after a reboot using your patch? How long
|will that
|take? Is the time to re
]
Sent: Monday, March 27, 2006 6:05 AM
To: asterisk-users@lists.digium.com
Subject: FW: [Asterisk-Users] Re: Fw: anybody has SIP
realtime working ?
Actually, I have tested this here with an Aastra 9133i and an
[EMAIL PROTECTED] server, and the 9133i will re-subscribe on its
own after
AK == Andrew Kohlsmith [EMAIL PROTECTED] writes:
AK There is no mechanism in place for the DB to tell Asterisk that a
AK row changed and that the cache is invalid. If you are using the
AK cache in Asterisk you must manually clear out the peer entry to
AK get the new value, or simply wait for the
On 3/24/06, mustardman29 [EMAIL PROTECTED] wrote:
So your Polycom 501's will eventually re-subscribe and BLF will eventually
start working again after a reboot using your patch? How long will that
take? Is the time to re-subscribe something you can set on the phone?
That would be quite
On Friday 24 March 2006 06:17, Benny Amorsen wrote:
In a slightly more ideal world, asterisk could be told: reload sip
peer whatever, and would only update the changed values while
retaining MWI and qualify information etc.
Isn't that *exactly* what sip prune realtime peer x does?
*CLI sip
AK == Andrew Kohlsmith [EMAIL PROTECTED] writes:
AK If you have a specific problem, let's hear it.
If I add or remove a pickupgroup or call group from a phone in the
database, I need to sip reload.
/Benny
___
--Bandwidth and Colocation provided by
23 mar 2006 kl. 16.24 skrev Benny Amorsen:
AK == Andrew Kohlsmith [EMAIL PROTECTED]
writes:
AK If you have a specific problem, let's hear it.
If I add or remove a pickupgroup or call group from a phone in the
database, I need to sip reload.
The reason we call them dynamic peers and users
If I add or remove a pickupgroup or call group from a phone in the
database, I need to sip reload.
No you don't... sip prune realtime exten works like a charm. The single
phone re-registers and the new info is in the system. No reloading
required :)
--
Aaron Daniel
Computer Systems
: Olle E Johansson [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 23, 2006 7:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Fw: anybody has SIP
realtime working ?
23 mar 2006 kl. 16.24 skrev Benny Amorsen:
AK == Andrew Kohlsmith [EMAIL
: Thursday, March 23, 2006 12:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP
realtime working ?
Sorry for the stupid question because I don't really
understand the gory
details of this as much as I'd like to.
Do
OEJ == Olle E Johansson [EMAIL PROTECTED] writes:
OEJ Unless you turn on caching, but that's another story. SIP reload
OEJ will be needed to clear them out from memory and load them again
OEJ from database next time they communicate with Asterisk.
Not much good when MWI and BLF require caching.
AD == Aaron Daniel [EMAIL PROTECTED] writes:
AD No you don't... sip prune realtime exten works like a charm. The
AD single phone re-registers and the new info is in the system. No
AD reloading required :)
Thank you, that will be very useful.
/Benny
-
From: mustardman29 [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 23, 2006 12:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP
realtime working ?
Sorry for the stupid question because I don't really
: Thursday, March 23, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP
realtime working ?
I _think_ you can do a 'sip reload' instead of a 'reload' and
keep your BLF...
But as for a server boot don't be crazy
: Thursday, March 23, 2006 2:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Fw: anybody has SIP
realtime working ?
On 3/23/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I _think_ you can do a 'sip reload' instead of a 'reload'
and keep your
]
Sent: Thursday, March 23, 2006 4:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP
realtime working ?
Thanks BJ,
I tried your patch and it worked fine for me so thank you so
much for the
effort. It is very much
On 3/23/06, mustardman29 [EMAIL PROTECTED] wrote:
Thanks BJ,
I tried your patch and it worked fine for me so thank you so much for the
effort. It is very much appreciated. Especially since I am not capable of
coding myself.
Unless I can get a total solution so that it just works no matter
On 3/23/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I don't know why the situation is different, but we've been using Polycom
phones with BLF, and it's ok. I'm using Asterisk 1.2.5, and a 'reload' will
clear sip subscriptions and BLF, but a 'sip reload' does not.
It's different because
Message-
From: mustardman29 [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 23, 2006 4:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: Fw: anybody has SIP
realtime working
?
Thanks BJ,
I tried your patch
Discussion
Subject: Re: [Asterisk-Users] Re: Fw: anybody has SIP
realtime working ?
On 3/23/06, mustardman29 [EMAIL PROTECTED] wrote:
Thanks BJ,
I tried your patch and it worked fine for me so thank you
so much for
the effort. It is very much appreciated. Especially since
I am
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