> I use realtime on 1.4 and 1.6 servers but always with rtcachefriends=yes in
> sip.conf
I already use that and it doesnt seem to re-register when a call comes in.
Only when the registration period expires, or the peer dials out.
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On Tue, 2010-09-21 at 19:04 -0400, Dan Journo wrote:
> I checked the bug reports and all I could find was similar issues with the
> Asterisk 1.6 which (according to the reports) have been resolved.
> I couldnt find anyone talking about 1.4 so I created a new issue and someone
> closed the case an
I checked the bug reports and all I could find was similar issues with the
Asterisk 1.6 which (according to the reports) have been resolved.
I couldnt find anyone talking about 1.4 so I created a new issue and someone
closed the case and added this note:-
> This does not appear to be a bug, but
On Mon, Sep 20, 2010 at 11:59:16AM -0400, Dan Journo wrote:
>Can we not do pastebin any more?
No, it's just one user with an excessively paranoid and chatty
mailfilter.
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> My question is "if you are using realtime, why are you doing a sip reload?"
I said previously:-
> Let's say I add a new provider to my service and therefore have to add
> another "register=>" command into sip.conf, I'd have to issue a "sip reload"
> which would kill off all the realtime sip p
users@lists.digium.com;
Subject: Suspicious URL:Re: [asterisk-users] Bug with Realtime?
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: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Monday, September 20, 2010 10:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?
> Check the SIP debug and see
> Check the SIP debug and see what is going on.
> Leif.
Hi,
I checked the SIP debug.
As soon as I issue the RELOAD command, no SIP data gets transferred to the
phone.
Asterisk output: http://pastebin.com/FB675N16
Any ideas how I can do a SIP reload without losing the Sip Phones registration?
> Check the SIP debug and see what is going on. Alternatively you could turn
> off
the qualify option with qualify=no.
I'll take a look at the sip debug, but qualify needs to stay on, so thats not
an option.
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On 10-09-16 09:43 AM, Dan Journo wrote:
>> That's not a bug. Only when the phone registers or performs some sort of
>> action
>> (such as placing a call, etc...) does Asterisk query the database. If your
>> phones have a short re-registration time this becomes less of a problem.
>
> How do you exp
On Thursday 16 September 2010 11:23:37 Dan Journo wrote:
> Is there any development work being done on the realtime addon? Theres been
> no updates since April.
Realtime is integrated into the core; it is not an addon. Perhaps you're
referring to the mysql realtime driver? The driver modules ten
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime
Is there any development work being done on the realtime addon? Theres been no
updates since April.
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> Have you checked the Issue Tracker
Not yet. I wanted to see if it's just me before searching through/raising a bug
report.
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> But it doesnt explain why the phones that are hard coded in the sip.conf
file don't lose registration.
On a reload, it re-reads the sip.conf config file and sees the users in
there, so it doesn't flush them. It doesn't pull down the whole SIP table
on a reload, it only loads a realtime peer con
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime
> Noted - but if OP does a "reload" once a day, 120 seconds (2 minutes) out of
> 1 day (14400 seconds) is 99.17% uptime; "close enough" to 99.999 percent in
> most folks books. What percentage of businesses use their phones 24/7?
Even if its once a month, it's still too much in my book. No wonder
do a 'reload'", do an "extensions reload" or whatever it is specific
to your changes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-bo...
Sent: Thursday, September 16, 2010 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Di
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Thursday, September 16, 2010 9:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 9:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime
Danny Nicholas wrote:
If your clients can't take 2 minutes of "downtime" on a phone, they
don't need to be on VOIP.
If VOIP ( and Asterisk ) ever really expect to be "the future of
Telephony " this ( attitude ) has to change
90 percent availability is unacceptable, even 95 percent,
> As someone else said, the answer is
"don't do a 'reload'", do an "extensions reload" or whatever it is specific
to your changes.
You are correct. I'm just being lazy. But I'm just worried that some time in
the future, I'll have to reload the sip config, and therefore flush out all the
realtime
> A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone.
Finally an answer that seemed more realistic. But it doesnt explain why the
phones that are hard coded in the sip.conf file don't lose registration.
Any ideas?
Thanks
Dan
--
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime
sage-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?
> That's
> That's not a bug. Only when the phone registers or performs some sort of
> action
> (such as placing a call, etc...) does Asterisk query the database. If your
> phones have a short re-registration time this becomes less of a problem.
How do you explain that as soon as I issue a "reload" comma
You can do 'extensions reload' or 'ael reload' if you don't want to lose
real-time sip registrations. I only reload what is needed to be reloaded
instead of reloading everything.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-15 4:28 PM, "Leif Madsen" wrote:
On 10-09-15 03:41 PM, Dan Jour
On 10-09-15 03:41 PM, Dan Journo wrote:
> I think ive found a bug but need someone to double check.
>
> Whenever I issue a "reload" in Asterisk, any realtime extensions stop
> receiving calls.
>
> I have to reboot the sip phones in order to get them to re-register.
>
> Can anyone see if they have a
On 09/15/2010 09:41 PM, Dan Journo wrote:
Hi,
I think ive found a bug but need someone to double check.
Whenever I issue a "reload" in Asterisk, any realtime extensions stop
receiving calls.
I have to reboot the sip phones in order to get them to re-register.
Can anyone see if they have a
> By reload you mean "sip reload" or just any reload in general?
Reload in general.
It might be an issue only with the Polycom sip phones. Not been able to test
any others. I'll try a software phone tomorrow.
--
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, September 15, 2010 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Bug with Realtime?
Hi,
I think ive found a
Hi,
I think ive found a bug but need someone to double check.
Whenever I issue a "reload" in Asterisk, any realtime extensions stop receiving
calls.
I have to reboot the sip phones in order to get them to re-register.
Can anyone see if they have a similar problem?
Asterisk 1.4.32
Mysql realti
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