We are experiencing cutting in and out and users cannot hear
each other using meetme conference. This behavior is noticed thru most of
the call. We are using Asterisk 1.0.7 with a Digium TE410p for the zap PRI
lines on a Dell 1850. We noticed the following in the output from the CLI
Unable
What causes this?
Dec 13 15:16:06 NOTICE[2660]:
chan_iax2.c:1561 iax2_destroy: Avoiding IAX destroy deadlock
Something occurs and I get a flood of
these then the box quits taking calls and asterisk wont die.
-Jonathan
The contents of this email message and any attachments are
i have two cards md3200 buy they dont work is
possible connect two single phone lines with 2 cards x100 clone ??
Visita www.tutopia.com
y comienza a navegar ms rpido en Internet. Tutopia es Internet
para todos.
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On Tue, Dec 13, 2005 at 05:47:47PM -0200, Dov Bigio wrote:
Actually I did it manually (tar -xvzf)... but I am not sure which files I
have to delete manually.. is there an explanation somehere? I couldn't find
it on Google...
If you do a tar -tvzf of the same tar archive, those will be the
Hi,
I've got problems with asterik 1.2 and a card TE411P.
Firstly I often got blanks (1 sec ) during a conversation between a
sipphone and a PSTN line (incoming calls from TE411P).
Secondly, I baught this card because it seems to resolv echo problems
even if the environnement is dusty. But i
you try the xten sip phone?, i have similar troubles with other sip phones!
www.xten.com
http://www.xten.com/index.php?menu=download
- Original Message -
From: michael delvoye [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
No, I try xlight but I change my computers (DELL) and I've got a
problem to configure the micro (My voice looks like a robot???)
but I 'll try tomorrow with an other PC.
Your pb was similar ? (blanks during the conversation + echo ?)
What sort of headset do you use with xten ? I must equip a
Derek Conniffe wrote:
Anyone any ideas? (apart from go buy other hardware :).
I've yet to get mISDN + chan_misdn (or even mISDN + chan_capi-cm) to
work. My advice is to stick with CAPI+chan_capi-cm. :) No matter how I
compiled chan_misdn, Asterisk would always segfault on startup (on both
Well its official, libpri 1.21 will not work with asterisk 1.0.9 even if
already using zaptel 1.2.1.
I tried to match up the patch in http://bugs.digium.com/view.php?id=5266 to the
1.0.9 version of chan_zap.c and it is too different
for me to make the changes.
I will have to look hard at
I fell right flat on the floor when I pulled it out of the mailbox a few
minutes ago.
The cover picture (and story) are about Mark Spencer and Asterisk. . .
B.
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To
Dear all !
Do somebody have any idea on the way to add empty
line in a SIP NOTIFY, throught the sip_notify.conf
The problem is each line mus begin with "=" which
is then replaced by a ":" at run time.
But it would be great if I can add an empty line in
order asterisk no to be limited to
Hi, All.
We recently installed Asterisk 1.2.1 through the Debian package/CVS.
The CLI, however, seems to be missing some of the commands I'm familiar
with in older versions of Asterisk, namely the SIP and IAX2 commands, as
well as extensions reload.
I've viewed the changelog, and frankly I'm
We currently are running Asterisk 1.0.7 (though we hope to upgrade to
1.2.1 shortly). Most users are complaining that digits are not being
recognized properly when asked for a long string, such as CC number or
telephone number. We've played with the DTMF settings in our config
files, we've played
Recommend you upgrade to Asterisk 1.2.1 and chan_capi-cm-0.6.1
Have no problems here with a AVM C4 and 2 lines (4 channels in P2P mode)
plus a line in P2MP (MSN) mode.
Mike
- Original Message -
From: stéphane plichon [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Are the sip and iax modules actually loading? Try starting asterisk with
asterisk -cgvv and see if it hangs on anything. If it hangs on a
particular module, you can set it to 'noload' in modules.conf
On Tue, 13 Dec 2005 16:57:08 -0500, Leah Newmark [EMAIL PROTECTED]
wrote:
Hi,
Hi:
I am testing IAXmodem+HylaFAX setup on asterisk. I have two * boxes.
They can send faxes to each other, it works perfect. But if I register
The two boxes into a provider I can't send a fax to each other
(my temporal provider is goiax.com).
IAXmodem seems to transmit something, but it
Miguel Soto wrote:
Hi:
I am testing IAXmodem+HylaFAX setup on asterisk. I have two * boxes.
They can send faxes to each other, it works perfect. But if I register
The two boxes into a provider I can't send a fax to each other
(my temporal provider is goiax.com).
IAXmodem seems to transmit
About the 4801, Kristian said:
- No FXS ports - the Soekris doesn't have the means to
provide ringing voltage for the card.
Doesn't it use the 5V rail of a standard molex connector to generate ring
voltage? Or does it use the 12V rail. If it's the former, I think you could
probably use power
Hey all - I'm sure this has been done before, but I'm curious about how well
it works.. Typically we have all our servers setup for dual fast/gig
ethernet failover... I.e. bond0 slaves eth0 and eth1 and fails over between
the two. This together with dual p/s and raid1'd(at least) drives
List users,
I've traced the writing of the leg files to two functions in channel.c:
ast_write()
ast_read()
They both contain similar code, so I'm going to limit my analysis to one
of them. If I'm misunderstanding anything or am flat out wrong, please
don't hesitate to correct me. Your input
On FC2, I execute a shell script as a user: asterisk (i.e. non-root) from
System() and it executes, but the commands that the system executes will not
nice 19. Instead, I get permission denied and the commands execute anyway,
but as nice 0. Now, it was my understanding that non-root users could
Matt,
I have a similar issue to the 'Skips and Pops' with the On Hold
music on my Ast 1.2.1 box. I've tried moving stuff to a RAM Disk, yet I
still get reports from agents that callers report that the 'music on
hold sounds horrible'.
It has squeaks and pops.. kind of like digital
I can say that I've implemented it on several Asterisk servers using the
802.3ad mode and it works very well. The failover is quick, and there are
none of the issues mentioned here. I'm not particularly concerned about
running at GigE speeds as the level of traffic in/out of these boxes is
I found www.digiumchina.com
How many asterisk companions in china?
-邮件原件-
发件人: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 代表 Chris Bagnall
发送时间: 2005年12月14日 7:17
收件人: 'Asterisk Users Mailing List - Non-Commercial Discussion'
主题: RE: [Asterisk-Users] Small / embedded system
Nice response.
I just wanted to let you know that we do provide a tool to debug echo.
We send a unit impulse and record the Finite Impulse Response (FIR) so it
can be plotted and analyzed. The code that does this is the release at
ftp.sangoma.com/linux/custom/2.3.4. Instructions on using
I think both asterisk and Mark Deserve it :)
Congratulations Mark :)
We want you on TIME next :)
Rehan
On 12/14/05, Brian Capouch [EMAIL PROTECTED] wrote:
I fell right flat on the floor when I pulled it out of the mailbox a fewminutes ago.The cover picture (and story) are about Mark Spencer
On Tue, 2005-12-13 at 19:15 -0500, Watkins, Bradley wrote:
I can say that I've implemented it on several Asterisk servers using the
802.3ad mode and it works very well. The failover is quick, and there are
none of the issues mentioned here. I'm not particularly concerned about
running at
From: Griffith,
MichaelSent: Tue 12/13/2005 7:50 PMTo:
asterisk-users@lists.digium.comSubject: Asterisk 1.2 SIP register
problems
Hello All,
I am using Asterisk 1.2. I have it
registering to a SIP carrier for DID's and to 2 NEC PBX's. The PBX uses
UDP port 5060 for SIP trunking and 5070
I was wondering if anyone has used asterisk in a real estate
development project. I know someone that is developing a ~400
home project and thought asterisk might be a possible
alternative to the phone company and a way to offer more
service to buyers.
How about deploying asterisk to
Rich Adamson wrote:
.
Last, the bonding of two nics at the server level _requires_ the associated
switch interface to support the exact same bonding algorithm. Historically,
that has been a problem for many switch vendors.
Not so sure I understand, but if you mean, 'the algorithm to select a
On Wed, 2005-12-14 at 08:42 +0800, hoowa sun wrote:
I found www.digiumchina.com
How many asterisk companions in china?
I dont know about them specifically, I havent ever dealt with them,
however last week on asterisk-biz someone was saying that this company
bought hardware and bounced some
We are working with a few Developers, but asterisk is only one part of the
solutionbut we are using it for the telephony side of things, combined
with Channel banks etc...etc..etc..
The Biggest Bugbear is billing.
We are also rolling out and maintaining a GEPON structure...so everything
Anybody got already to make Vizufon CIP-4500 working with Asterisk through
SIP?
I got to register by Asterisk send a Notify back and receive a Bad
Request
Isamar
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On Sun, 2005-12-11 at 16:18 +0200, Warren Burstein wrote:
I'm running asterisk 1.0.9 with TDM400B's for both internal and external
lines.
I put in the macro that dials outside lines an AbsoluteTimeout(36000),
never expecting it to happen. But it does, a few times a month.
I've noticed
Doug Lytle wrote:
I agree with Eric on this one. On my Polycom IP501s, I had to change
the digit map to allow for # and * matching. For testing, remove the
# and try again.
Remove it from the phone's dial plan or all together? Also, my phone
has a local dial plan that is set to this:
hey chris,
The only issue you'll run into is that with all temp stuff like construction
trailers ect they like to cut there lines A LOT with all there nice
machines.
Carlos Alcantar
Race Technologies, Inc.
101 Haskins Way
South San Francisco, CA 94080
P: 650.246.8900
F: 650.246.8901
E: [EMAIL
I fell right flat on the floor when I pulled it out of the mailbox a
few
minutes ago.
The cover picture (and story) are about Mark Spencer and Asterisk. . .
B.
That's cool. I have not received the latest issue yet but look forward
to it.
Was there some sort of full disclosure
I'm curious about something minor, and haven't seen anything covering it
yet:
If you want to match an 11 digit US phone number (and know that ONLY proper
numbers are being passed to this portion of your dialplan), which is more
speed memory efficient:
exten = _1.,1,VERBOSE(1|${EXTEN})
Very true. How about using WiFi and DEC phones?
Thanks,
Steve
hey chris,
The only issue you'll run into is that with all temp stuff like
construction
trailers ect they like to cut there lines A LOT with all there nice
machines.
Carlos Alcantar
Race Technologies, Inc.
101 Haskins
Don't quite understand what you mean by multi sms ports?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Friday, December 09, 2005 12:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] New
It supports US power or EU power system.
Don’t worry about it.
It has been tested before and it should be
able to be used world wide without a problem
Sam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rusty Dekema
Sent: Friday, December 09, 2005
11:01 PM
been there done that
The biggest problems that we have had to deal with is stability. Stability
with respect to system stability and also stability of the user. System
stability is extremely difficult to deal with when you have poor power,
power outages, guys pressing the buttons next to the
On Dec 13, 2005, at 8:25 AM, Michael George wrote:
On Fri, Dec 09, 2005 at 03:23:31PM -0500, Tom Rymes wrote:
On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi:
i added these two lines to my general context ,but
nothing happened the same result the sound came in one
way for 3 seconds
Colin,
Nice summary, what gateway are you using and with what carrier.
Thanks,
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Colin Anderson
Sent: Tuesday, December 13, 2005 10:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Thanks to everyone who took all of the extra digium hardware off my
hands. Now I have some other misc. stuff which I have been using with an
asterisk deployment but is no longer needed:
Cisco 7960 http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5842276412
Cisco 7960
SIP Subscriptions are cached in memory. I'd like to find out if there's a way
to have them cached in a file, say astdb ('ie database show') rather than in
RAM. If your Asterisk process is stopped and restarted, you lose all the SIP
subscriptions. Registrations on the other hand are still in the
Rogers (Canada) + Ateus VoiceBlue 4 port - total bitch to set up but works.
Don't know if I'd recommend it though. I'm sure there's something else
better out there. Rogers is also difficult to deal with but then what
carrier isn't. But in the end everything works. There's other tricks you
have to
On Wed, December 14, 2005 3:33, Robert La Ferla said:
Doug Lytle wrote:
I agree with Eric on this one. On my Polycom IP501s, I had to change
the digit map to allow for # and * matching. For testing, remove the
# and try again.
Remove it from the phone's dial plan or all together? Also, my
Douglas Garstang wrote:
I can't understand why it was implemented this way (lack of design maybe?).
Yep, that's it. Asterisk was designed by a bunch of fools who never
even gave the first thought to what they were coding up.
Yore kinda quick to knock over the china, pardner.
B.
Hi!
I was hoping someone might answer a few questions.
For a SIP user it is possible to configure
something called regcontext and regexten. My
understanding is that when the use registers with
asterisk it will automatically add an extension
for the user in the context specified by
regcontext?
How
Kristof Hardy wrote:
The problem has been consistent from 1.0 through CVS to 1.2, and
across different machines and distributions. Does anyone have any
suggestions on how I can deal with this? I have had echo cancellation
happening, but half-duplex speech is not acceptable.
You're not using
stéphane plichon wrote:
stéphane plichon wrote:
Hi all,
currently i running * 1.0.9 with chan_capi 0.3.5
my first problem is:
in incoming call, when BCHAN is full in contr1 incoming call on contr2
are not answered with error :
chan_capi.c:1953 in capi_handle_msg: received a call waiting
hi
list,
does anyone know how
to configure asterisk to be able sending
and receiving
SMS over my
own SMS gateway?
it is connected via
a serial (V24) cable on the asterisk server.
i know that i have
to use COM1 and our tel.number;
(ServiceCenterAddress) but don't know where toconfigure
On Wed, 14 Dec 2005, stéphane plichon wrote:
stéphane plichon wrote:
stéphane plichon wrote:
Hi all,
currently i running * 1.0.9 with chan_capi 0.3.5
my first problem is:
in incoming call, when BCHAN is full in contr1 incoming call on contr2
are not answered with error :
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