On 09/16/2010 07:58 PM, Paul Belanger wrote:
Please do not send me direct email, post them to the list for others
to help. Your backtrace is optimized (value optimized out). You
need to reinstall asterisk with DONT_OPTIMIZE enabled, described in
doc/backtrace.txt.
Hello,
I have
Dear all,
We're experiencing some (what appear to be) deadlocks using
asterisk-1.4.35 (the problem also occurred on 1.4.24). Some of the
symptoms are that new SIP calls cannot be established and when running
sip show channels from the CLI, the CLI stops responding to any
further commands. The
On Thu, Sep 16, 2010 at 10:03:09AM -0400, Jerry Geis wrote:
Jerry Geis wrote:
Somewhere on your system you have a modprobe install command that's
running when the module is loaded. Most likely it was installed on your
system by
Hi all,
i have the following setup
PSTN - routing server (asterisk 1.6.2.11) - IAX - callcenter asterisk
1.6.2.9 - SIP - agent
Does work quit fine - then agent does have the abibility to transfer a call
to a third party - the agent can initiate the transfer over a web interface
- it does
Hi,
I'm experiencing an issue with asterisk 1.6.2.10 12rc1,
i'm not sure if it's to be expected or not, so here it is:
When transferring call (blind-transfer) using asterisk feature key,
things are working OK, however when using ZoIPer's transfer key
(which is implemented with a Refer-To SIP
Hi
Does Sangoma 8-port card A108 support PCIe version 2.0 ?
The cards is here
And we want to use 3 such cards in this motherboard because it has 3 PCIe
slots of version 2.0
http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm
Is this a good idea ? Do you have any
2010/9/17 Benoit maver...@maverick.eu.org
Hi,
I'm experiencing an issue with asterisk 1.6.2.10 12rc1,
i'm not sure if it's to be expected or not, so here it is:
When transferring call (blind-transfer) using asterisk feature key,
things are working OK, however when using ZoIPer's transfer
2010/9/17 Wolfgang Pichler wpich...@yosd.at
Hi all,
i have the following setup
PSTN - routing server (asterisk 1.6.2.11) - IAX - callcenter asterisk
1.6.2.9 - SIP - agent
Does work quit fine - then agent does have the abibility to transfer a call
to a third party - the agent can
Hi
Does Sangoma 8-port card A108 support PCIe version 2.0 ?
The card is here
http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html
And we want to use 3 such cards in this motherboard because it has 3 PCIe
slots of version 2.0
Hi,
How to create dialplan restriction for particular extensions..
--
Thank you with regards,
Gopalakrishnan A.N,
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Hi all,
to be able to transfer calls I have set call-limit to 2 for all of my peers.
Now how can I determine if a peer is in busy state using the first line if I
don't want to route a second call to it?
Thanks in advance,
Oliver
--
Anita Hall wrote:
Hi
Does Sangoma 8-port card A108 support PCIe version 2.0 ?
Ask Sangoma They are very helpful
The card is here
http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html
And we want to use 3 such cards in this motherboard because it
i suppose that depends on the number of eggs and baskets you have... but i'm
guessing not many of either since you're considering using a desktop board
for this...
but, email sangoma support, they will tell you.
On 17 September 2010 12:47, John Novack jnov...@stromberg-carlson.orgwrote:
2010/9/17 Olivier oza_4...@yahoo.fr
2010/9/17 Wolfgang Pichler wpich...@yosd.at
Hi all,
i have the following setup
PSTN - routing server (asterisk 1.6.2.11) - IAX - callcenter asterisk
1.6.2.9 - SIP - agent
Does work quit fine - then agent does have the abibility to transfer a
call
I'd forgot about doing it that way (I use that for $).
Thanks for the memory jog :)
Cheers
Andy
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 16 September 2010 13:51
To: Asterisk Users
Any hints please?
I would appreciate your valuabl help
Thanks
2010/9/16 Danny Dias ing.diasda...@gmail.com
Hello my friends,
I would like to understand the output from pri intense debug span X, the
Telco always says that their side is OK, but i always receive alarms,
loosing connection,
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.
--
_
-- Bandwidth
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan
A.N
Sent: Friday, September 17, 2010 6:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call restriction for particular
Hi All,
We are running to a weird problem, we're using asterisk 1.2 as a production
server (I'm wiling to move very soon to more recent version) and our problem
is when somebody try to join a conference he's told that he's the only one
in the conference but in fact there is some 3 or 5 or whatever
Geraint Lee wrote:
i suppose that depends on the number of eggs and baskets you have...
but i'm guessing not many of either since you're considering using a
desktop board for this...
24 T1 ports, if my math is correct.
Lots of eggs for any PC, desktop or not!
Lots of circuits/channels to go
With some carriers the initial Audio (2-4 secs) seems to get cut off when using
a Auto Attendant or Conf Meetme.
Is there any known remedies for that. Just want to know if others have seen
that esp. with Level 3.
If Auto Attendant says - Welcome to ABC bank
Caller only hears Bank
--
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval
Karihaloo
Sent: Friday, September 17, 2010 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Initial Audio Cut off
With
Hi,
I have a Netmod ISDN BRI router and from the router I have connected the
analog port in Asterisk via FXO card. Two analog lines I have connected to
asterisk machine. When both the lines are established, after 31 minutes the
call is automatically disconnected.
While checking the log it shows
On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
warning: exec file is newer than core file.
Jonas,
I encourage you to read the output. Did you run gdb with a core file
dumped from the old build? You need to generate a new core dump with
the new executable.
While this is too many eggs in one basket, but can be useful if you have
too many E(T)1s say equivalent to a STM1 (OC3) or more. In that case, it
would be too many boxes at 8ports / box.
Somewhere in the mailing list, Sangoma devs said that they do 32E(T)1 per
box on the labs quite frequently,
Hi list
i setup successfull asterisk version 1.4 + opensips,
Opensips is the Registrar Server, Asterisk is the IVR server
the call flow
IP phone ---INVITE 1001 opensips - ASterisk INVITE
5001---opensips --- Busy|cancel|404..---asterisk---wait 10s to bye ---IP
phone (5000)
my case
On 09/17/2010 05:29 PM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellensjonas.kell...@telenet.be
wrote:
warning: exec file is newer than core file.
Jonas,
I encourage you to read the output. Did you run gdb with a core file
dumped from the old build? You
On Fri, Sep 17, 2010 at 11:51 AM, Jonas Kellens
jonas.kell...@telenet.be wrote:
On 09/17/2010 05:29 PM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellensjonas.kell...@telenet.be
wrote:
warning: exec file is newer than core file.
Jonas,
I encourage you to read the output.
All-
Recently an Asterisk server we host was hacked and used to route some
unauthorized calls. We have since improved our
security measures, including installation of fail2ban.
The interesting thing is the way in which this was discovered. The
unauthorized calls were occurring intermittently
On Fri, Sep 17, 2010 at 9:24 AM, khalid touati khalidtou...@gmail.com wrote:
in the dialplan, that would be a big help if you guys can help diagnose the
issue.
A debug log of the actually problem will be more helpful.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
Is there the ability in the Asterisk 1.8 CEL logging to log the SIP
endpoint IP as weell as the medie enpoint's ID's?
Thanks
Bryant
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Friday, September 17, 2010 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Not able to join
Hi All,
We have a production system running 1.4.25.1 and yesterday we upgraded it to
1.4.36. Basically we use this system to generate scheduled calls via .call
files.
Sample .call file used:
Channel: local/11...@context-out
WaitTime: 30
CallerId: 3
Extension: 2
Context:
I'm trying to use a couple of old Western Electric type 500 phones (desk
model, rotary dial). These phones work fine, as tested with telco lines
(they dial, receiver/transmitter works fine, etc).
I'm running Asterisk 1.6.2.11.
I can't get them to dial through Asterisk. They are connected to a
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joel Maslak
Sent: Friday, September 17, 2010 12:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Rotary phone on Asterisk
I'm trying
I have had where the Phone provider (this was a PRI) cut long distance
service to a box that was compromised till we called them to assure
them that the security holes where fixed.
On Fri, Sep 17, 2010 at 1:10 PM, Jeff Brower jbro...@signalogic.com wrote:
All-
Recently an Asterisk server we
Just put in:
Answer()
Wait(1.5)
On Fri, Sep 17, 2010 at 11:00 AM, Ujjval Karihaloo
ujj...@simplesignal.com wrote:
With some carriers the initial Audio (2-4 secs) seems to get cut off when
using a Auto Attendant or Conf Meetme.
Is there any known remedies for that. Just want to know if others
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, September 17, 2010 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Initial Audio Cut off
Just
On 09/17/2010 06:00 PM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 11:51 AM, Jonas Kellens
jonas.kell...@telenet.be wrote:
On 09/17/2010 05:29 PM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellensjonas.kell...@telenet.be
wrote:
warning: exec file is
I recently came across this email that I wrote in May 2008 ..
http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html
It's such a shame that Digium manhandled the project away from the community
only to then bury it and not allow it to proceed. I really wonder when I look
I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys
SPA942 SIP phone and outgoing SIP and IAX routes.
When I dial local PSTN numbers from the SPA942 using the FXO channels I
observe a 5-7 second delay between when the PSTN number answers the call
and when
Is DAHDI the Analog /PRI card..or something.. We never use it..
Call is delivered over SIP from the carrier...and plays the standard WAV file
in Asterisk...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday,
We have already tried that...but still there is say 1.5 sec delay but the
actual Audio first 2-4 secs still get cut off..
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
-Original
Have you tried something like this
exten x = 1,Answer()
exten x = n,Wait(2)
exten x= n,(whatever you are doing now)
Thanks,
Lyle J. McKarns
---
Network Engineering Team
n|m Nexus Management
4 Industrial Parkway
Suite 101
Brunswick, Maine 04011
Tel (USA)
In 1.4 I used alsa.conf and Dial(Console/Dsp)
In 1.8 this is not working (as I had it) . I know there is a new
chan_console
I'd like to try both.
What is the correct Dial() for ALSA direct?
What is the correct Dial() for chan_console?
I thought if chan_alsa was loaded it would default to old
Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Joel
Maslak
*Sent:* Friday, September 17, 2010 12:29 PM
*To:* Asterisk Users
Hi Guys,
Paul
you meant a debug file while the problem is happening, actually the thing is
i cannot even reproduce this issue, I'll keep trying though, but is there a
way to debug just Meetme app output?
On Fri, Sep 17, 2010 at 1:04 PM, Danny Nicholas da...@debsinc.com wrote:
-Original
On Friday 17 September 2010 12:51:16 Dean Collins wrote:
I recently came across this email that I wrote in May 2008 ..
http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html
It's such a shame that Digium manhandled the project away from the
community only to then bury it
My understanding was that pulse dialing from a channel bank was iffy, but
not pulse reception, so long as the channel bank properly reports on/off
hook state - that there is no real pulse detection in the channel bank,
simply on/off hook status (looking at some of my documentation, real D-2,
D-3,
I am getting several hundred registration attempts on my aserterisk per
minute. I have fail2ban installed but it's not stopping the attempts. Any
suggestions. Whatever they are using is changing the userid on each
attempt.
Latest IP: 209.172.57.219
Thanks,
Dave
--
I wrote a script to help with these here:
http://www.teamforrest.com/blog/171/asterisk-no-matching-peer-found-block
To each their own... there's 1000 ways of combatting this.
---fred
http://qxork.com
On Sep 17, 2010, at 5:18 PM, dave george wrote:
I am getting several hundred
What is the difference between this and the other option suggested below?
Just put in:
Answer()
Wait(1.5)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle McKarns
Sent: Friday, September 17, 2010 12:40
Wow when did that happen?
How come here is no reviews/traffic
Cheers,
Dean
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: Friday, 17 September 2010 4:03 PM
To: Asterisk
It means that fail2ban is not configured correctly on your machine. For me
it works fine, and in fact lately these registration/hack attempts have gone
up significantly, thanks to cloud computing I guess.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-17 5:28 PM, dave george
On Friday 17 September 2010 16:53:58 Dean Collins wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: Friday, 17 September 2010 4:03 PM
To: Asterisk Users Mailing List -
Hi All,
Is it possible to specify more than 1 localnet? I know this is an odd
question. I have a customer that has multiple sites linked by VPN.
Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24
We want to allow some access to the public IP address at the main site. For
this to
Any thoughts on why the lack of traffic?
Cheers,
Dean
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: Friday, 17 September 2010 6:37 PM
To: Asterisk Users Mailing List -
This week I was experiencing attacks sip log into my accounts were more than
1,000 requests for records Sip accounts in less than an hour THROUGH deny the
ip of my router access list in cisco and my asterisk server to go through the
iptables drop ip attacker is a way for an account with
On Friday 17 September 2010 22:52:02 Dean Collins wrote:
Tilghman Lesher wrote:
On Friday 17 September 2010 16:53:58 Dean Collins wrote:
Tilghman Lesher wrote:
On Friday 17 September 2010 12:51:16 Dean Collins wrote:
I recently came across this email that I wrote in May 2008
59 matches
Mail list logo
