risk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michaël
Gaudette
Sent: April 2, 2017 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Issue with Asterisk 13, multiple CDR per queue
and arbitrary upper limit
elatively small?
Regards,
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: April 1, 2017 6:56 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Issue with Aste
On Fri, Mar 31, 2017, at 10:55 PM, Michaël Gaudette wrote:
>
>
> Hi,
>
>
>
> I`ve recently upgraded a server from 1.8 to Asterisk 13. While
> everything
> is under control, I have one issue with the way CDRs are kept for queues.
> And I don`t mean “I don`t like it”. I mean it crashes the serv
Hi,
I`ve recently upgraded a server from 1.8 to Asterisk 13. While everything
is under control, I have one issue with the way CDRs are kept for queues.
And I don`t mean I don`t like it. I mean it crashes the server.
I realize there are multiple CDRs per queue call one per ring/per phon
Hi List
With asterisk 1.4 intermittently I have an issue when i call manually using
a hard phone Snom (sip extension) a number X I found another client with
another number Y
I don’t know if I have any things wrong in my configuration .i have this
issue 1/100
Any help will be appreciated.
> In that case it suggests it is some setting you have applied to the
> phones that is causing it.
I just called Aastra tech support. I'm always VERY impressed that the
first person who picks up the phone is very technical. He said that they've
had reports of this issue. The problem goes away
> In that case it suggests it is some setting you have applied to the
> phones that is causing it. Can you post the local.cfg & server.cfg
> files from the phone (removing the passwords from there first)?
Sure: local.cfg is checksums, server information, and:
contrast level: 3
ringer volume:
On 05/05/11 13:41, Richard Kenner wrote:
Asterisk does indeed send an Options before the OK but my 57i doesn't
seem to mind.
That's odd. It does for me.
Perhaps you need to upgrade firmware on the Aastra phone?
The problem occured when I DID upgrade it! Precisely to the one
you mentioned.
> Asterisk does indeed send an Options before the OK but my 57i doesn't
> seem to mind.
That's odd. It does for me.
> Perhaps you need to upgrade firmware on the Aastra phone?
The problem occured when I DID upgrade it! Precisely to the one
you mentioned.
> Or turning off qualify for this pee
On 05/05/11 04:37, Richard Kenner wrote:
I recently tried to update my Aastra 57i to version 3.2 and ran into
a problem. It won't properly register and says "contact mismatch".
I added "sip contact matching: 2" to aastra.cfg, but that didn't help.
When I look at the SIP trace, but I see is the
> Is asterisk replying differently when firmware 3.2 is used ?
No, but the phone cares with 3.2 and not with 2.6.
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2011/5/5 Richard Kenner
> I recently tried to update my Aastra 57i to version 3.2 and ran into
> a problem. It won't properly register and says "contact mismatch".
> I added "sip contact matching: 2" to aastra.cfg, but that didn't help.
>
> When I look at the SIP trace, but I see is the Aastra s
I recently tried to update my Aastra 57i to version 3.2 and ran into
a problem. It won't properly register and says "contact mismatch".
I added "sip contact matching: 2" to aastra.cfg, but that didn't help.
When I look at the SIP trace, but I see is the Aastra sending a
REGISTER and Asterisk rep
Hi,
Here is my confing:
[out]
Exten => _X.,1,Noop()
Exten => _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1))
Exten => _X.,3,Playback(tt-monkeys)
Exten => _X.,4,Playback(tt-monkeys)
Exten => _X.,5,Playback(tt-monkeys)
Exten => h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ)
[do_dtmf_cc-take-
Hi,
Here is my confing:
[out]
Exten => _X.,1,Noop()
Exten => _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1))
Exten => _X.,3,Playback(tt-monkeys)
Exten => _X.,4,Playback(tt-monkeys)
Exten => _X.,5,Playback(tt-monkeys)
Exten => h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ)
[do_dtmf_cc-take
Hi!
> It is causing an issue for me. One SIP UA works fine - ring, forward, etc.
> While the other does not.
Make the UAs listen on different ports (for example 5060 and 5062) and
see if that solves your problem - if you can't make them have different
IPs, that is.
Also be sure to fully under
Hey;
I never thought of that.
It is causing an issue for me. One SIP UA works fine - ring, forward,
etc. While the other does not.
I am a little clueless here - where would I start with this?
Thanks
Glen
On 11/1/2010 19:15, Philipp von Klitzing wrote:
> Hi!
>
>> [Nov 1 19:55:49] WARNING
Steve;
You are so right - it was the end of the day, I was tired and pissy.
Let me try this again:
Version:
ns211156*CLI> core show version
Asterisk 1.4.31 built by root @ some_server.foo.net on a x86_64 running
Linux on 2010-06-10 14:32:34 UTC
Name and version of endpoints involved:
Sip Sett
Hi!
> [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
> <6839>, digest has <3169>
You most likely have two SIP UAs that use the same IP, of which the 6839
account is listed last in sip.conf while 3169 is trying to auth
(unsuccessfully).
Philipp
--
___
On 1 November 2010 21:11, Silver Thorne wrote:
> Hey;
>
> Anyone see this before:
>
> [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
> <6839>, digest has <3169>
>
> G
>
>
> `
Is it causing a problem for you?
--
_
On Mon, 1 Nov 2010, Silver Thorne wrote:
> Anyone see this before:
>
> [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
> <6839>, digest has <3169>
You may have better luck with a more descriptive subject. Lots of users
have an issue or two with Asterisk.
Some details will al
Hey;
Anyone see this before:
[Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
<6839>, digest has <3169>
G
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New to Asterisk? Join u
Dear folks,
I'm using * realtime with no problems on most of the systems i've setup but
rarely i confront this problem that the asterisk doesn't load from database
when the systems rebooted and i have to reload it manually or restart it, but
it would work fine afterward, no problem how many ti
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