Hello,
I'm using an AMI Originate command to send a fax. The fax is sent by
a script, and I'd like my script to send the fax, wait until it has
succeeded or failed, then exit with an appropriate error code (it is
driven by a mail system, so the exit code will tell the mail system
whether to
Hi again,
I figured out the problem.
My dialplans were as follows..
8XXX are my asterisk number subnet..
other 3 numbers are my local numbers that works on an ericsson which is
connected by e1.
[local]
exten = _8XXX,1,Dial(SIP/${EXTEN})
exten = 1234,1,Dial(SIP/1234)
exten = 2345,1,Dial(SIP/2345)
Hi all,
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hello:
I can set a environments to test the two pri cards. the patlooptest is ok.
the result has no problem. how do i use patgen and pattest with two pri
cards and the setting of zaptel/sys.conf? in the
http://docs.tzafrir.org.il/man/pattest.8.html, there are no setting files
and cablling for
Hi all,
I need to specify two groups of peers who are on two sub networks, the
case is as follows:
two groups of users (that are supposed to use the X-lite) group1 and group2,
each group is on a sub network net1, and net2, respectively, each group has
its own dial plan defined in the
It appears I opened some flood gates when I offered my 'alternative'
version.
So, rather than send hundreds of e-mails out - here's the link :
http://www.dv-ip.com//downloads/files/misc/weather.txt
Any questions - just 'yell'.
Andrew Thomas
Technical Services Manager
a...@datavox.co.uk
DataVox
On 22/7/09 7:24 PM, Scott Gifford wrote:
Hello,
I'm using an AMI Originate command to send a fax. The fax is sent by
a script, and I'd like my script to send the fax, wait until it has
succeeded or failed, then exit with an appropriate error code (it is
driven by a mail system, so the exit
'host=dynamic' is your problem - as this allows any IP address to register as
that friend - assuming they know the password/username combination.
Why not simply have group 1 as 'secret=pass123' and group2 as 'secret=pass456'?
Just don't tell group 1 uses the password for group 2 - and
Scott Gifford schrieb:
I'm using an AMI Originate command to send a fax. The fax is sent by
a script, and I'd like my script to send the fax, wait until it has
succeeded or failed, then exit with an appropriate error code (it is
driven by a mail system, so the exit code will tell the mail
Hello,
I lookin' for a call in number from UK or USA. Can somebody offers me
a peering for this or specify any sip provider that offers this thing?
Thank you very much,
Jonson.
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On Wednesday, July 22, 2009, Catalin S. wrote:
I lookin' for a call in number from UK or USA. Can somebody offers
me a peering for this or specify any sip provider that offers this
thing?
There are several providers who offer UK or US regional geographical
numbers for little or no cost if you
Curious - Why?
What is the peer switch and why does it have this requirement?
John Novack
hadi motamedi wrote:
Dear All
Can you please let us know how we can modify our Asterisk
extensions.conf file so it interprets the subscriber dialed digits
in one-by-one digit manner . At its current
John Novack wrote:
Can you please let us know how we can modify our Asterisk
extensions.conf file so it interprets the subscriber dialed digits
in one-by-one digit manner . At its current configuration , it
interprets them in an whole packet . I mean , say the subscriber dials
as 665
Hello !
Are there any plans at Digium to include also german voice prompts ?
Thanks
regards
Hans
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Imagine that you have this code:
exten = _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an
argument: queuename
You should submit this as a bug. It may or may not get fixed, but it
definitely won't until someone reports it or takes it upon themselves to fix
it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny
Carlos Ruiz Diaz wrote:
@Steve: I considered the hardware separation between servers but when I
exposed the idea it was immediately discarded because it is mandatory to
have all in a box.
Well, I'll start the migration then.
Thank you.
I doubt this helps anyone, but today I built the
That is exactly what happens to me.
Still looking for a solution.
On Wed, Jul 22, 2009 at 9:44 AM, Thomas Kenyon dig...@sanguinarius.co.ukwrote:
Carlos Ruiz Diaz wrote:
@Steve: I considered the hardware separation between servers but when I
exposed the idea it was immediately discarded
hello all...I have been trying to get a handle on CallerPres with an
analog handset. I have usecallingpres=yes in my chan_dahdi.conf file
and when I dial *67 on my analog handset I see Disabling Caller*ID on
DAHDI/4-1 but when the call is then forwarded to my outbound SIP
provider the
Carlos Ruiz Diaz wrote:
That is exactly what happens to me.
Still looking for a solution.
Well, it's a step forward from what I was getting before.
Have you tried with different USB adapters and handsets?
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Yes, I tried with:
Dell Computer Corp. Wireless 355 Bluetooth, built-in
Encore, USB adapter.
Always with:
Nokia N80
Kernel: 2.6.27.21-0.1-pae.
On Wed, Jul 22, 2009 at 10:46 AM, Thomas Kenyon
dig...@sanguinarius.co.ukwrote:
Carlos Ruiz Diaz wrote:
That is exactly what happens to me.
I finally found a reason TO run Asterisk as root.
By default, ext[23] file systems reserve 5% of the filesystem for root.
Thus, you may get some warning when everything non-root starts failing
and give you a chance to free up some space before Asterisk is affected.
--
Thanks in advance,
Benny Amorsen schrieb:
Imagine that you have this code:
exten = _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT
Ketema Harris schrieb:
hello all...I have been trying to get a handle on CallerPres with an
analog handset. I have usecallingpres=yes in my chan_dahdi.conf file
and when I dial *67 on my analog handset I see Disabling Caller*ID on
DAHDI/4-1 but when the call is then forwarded to my outbound
On Wed, Jul 22, 2009 at 11:31 AM, Steve
Edwardsasterisk@sedwards.com wrote:
I finally found a reason TO run Asterisk as root.
By default, ext[23] file systems reserve 5% of the filesystem for root.
Hehe, sounds like a reason to standardize on ReiserFS
tune2fs -m 0 [device]
:) not anymore ;p
David Backeberg wrote:
On Wed, Jul 22, 2009 at 11:31 AM, Steve
Edwardsasterisk@sedwards.com wrote:
I finally found a reason TO run Asterisk as root.
By default, ext[23] file systems reserve 5% of the filesystem for root.
Hehe, sounds like
Hi,
I've got a general question about analog gateways (Xorcom, Audiocodes,
Patton, ...) .
Is it usual for analog gateways to detect when an analog phone is plugged in
or out ?
If positive, would it be then useful to send qualify queries for each
connect phone (I'm implying here that an analog
2009/7/22 Steve Edwards asterisk@sedwards.com
I finally found a reason TO run Asterisk as root.
By default, ext[23] file systems reserve 5% of the filesystem for root.
Do you imply this default can (and should) be changed ?
Is it the same for other filesystems ?
Thus, you may get some
2009/7/21 James Green james.gr...@mjog.com
Hi,
We currently fire multiple HTTP requests (via multi-curl) to the AJAM
interface in order to place calls. We are finding Asterisk has it's limits
however, and I've found astmanproxy recommended for helping maintain the
connections. This would
On Wed, Jul 22, 2009 at 10:31 AM, Steve
Edwardsasterisk@sedwards.com wrote:
I finally found a reason TO run Asterisk as root.
By default, ext[23] file systems reserve 5% of the filesystem for root.
Thus, you may get some warning when everything non-root starts failing
and give you a
Olivier wrote:
Hi,
I've got a general question about analog gateways (Xorcom, Audiocodes,
Patton, ...) .
Is it usual for analog gateways to detect when an analog phone is
plugged in or out ?
If positive, would it be then useful to send qualify queries for
each connect phone (I'm
On Wed, 22 Jul 2009, Olivier wrote:
2009/7/22 Steve Edwards asterisk@sedwards.com
I finally found a reason TO run Asterisk as root.
By default, ext[23] file systems reserve 5% of the filesystem for root.
Do you imply this default can (and should) be changed ?
Is it the same for other
On Tue, Jul 21, 2009 at 10:15 AM, James Green james.gr...@mjog.com wrote:
Hi,
We currently fire multiple HTTP requests (via multi-curl) to the AJAM
interface in order to place calls. We are finding Asterisk has it's limits
however, and I've found astmanproxy recommended for helping maintain
Jeff LaCoursiere schrieb:
2009/7/22 Steve Edwards asterisk@sedwards.com
I finally found a reason TO run Asterisk as root.
By default, ext[23] file systems reserve 5% of the filesystem for root.
So to rephrase it:
One GOOD reason to run asterisk as root is that you get to take
On Wed, 22 Jul 2009, Jonathan Moore wrote:
On Wed, Jul 22, 2009 at 10:31 AM, Steve
Edwardsasterisk@sedwards.com wrote:
I finally found a reason TO run Asterisk as root.
By default, ext[23] file systems reserve 5% of the filesystem for root.
Thus, you may get some warning when
Yeah, and GOTO's are a good reason not to use COBOL. But they both still
LIVE!! (Wah ha ha ha!!!)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: Wednesday, July 22, 2009 12:25 PM
To:
On Wednesday 22 July 2009 03:44:22 peace keeper wrote:
currentlly users in network1 can register as peer 2003 which is supposed to
be allowed just for users from network2 , although this registration is
supposed to be failed, any suggestions plz!!
hope I made the scenario clear , any help
On Wednesday 22 July 2009 07:20:41 Johann Steinwendtner wrote:
Are there any plans at Digium to include also german voice prompts ?
There are no plans currently, but we do accept translation contributions from
community members, wanting to ensure that prompts make sense for various
languages.
On Wednesday 22 July 2009 08:30:03 Danny Nicholas wrote:
You should submit this as a bug. It may or may not get fixed, but it
definitely won't until someone reports it or takes it upon themselves to
fix it.
Don't bother. It's not fixable.
--
Tilghman Teryl
with Peter, Cottontail,
Greeting everyone,
I'm trying to connect an old PBX to a Asterisk box with a 4 BRI card.
The idea is for the PBX to follow asterisk's dialplan rules such as
calling through VoIP when possible, ISDN when needed, etc, and all
incoming calls being redirected to the PBX.
The odd part is that
2009/7/22 Steve Underwood ste...@coppice.org
Olivier wrote:
Hi,
I've got a general question about analog gateways (Xorcom, Audiocodes,
Patton, ...) .
Is it usual for analog gateways to detect when an analog phone is
plugged in or out ?
If positive, would it be then useful to send
I see your name enough to know this must be a true statement; Can you
elaborate a little on why?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Wednesday, July 22, 2009 12:35 PM
To:
On Wednesday 22 July 2009 11:07:32 Steve Underwood wrote:
Olivier wrote:
Hi,
I've got a general question about analog gateways (Xorcom, Audiocodes,
Patton, ...) .
Is it usual for analog gateways to detect when an analog phone is
plugged in or out ?
If positive, would it be then
Hi,
I've created a tiny dialplan to test the return of a call on transfers,
like this: (I had to use the DEVSTATE backport here)
[phones]
exten = _12XX,1,Dial(SIP/${EXTEN},6,tT)
exten = _12XX,n,GotoIf($[ x${BLINDTRANSFER} = x ]?noBT)
exten = _12XX,n,Set(DIALRET=${CUT(BLINDTRANSFER,-,1)});
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: Wednesday, July 22, 2009 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] A reason TO run
on Wednesday 07/22/2009 Gordon Henderson(gordon+aster...@drogon.net) wrote
On Wed, 22 Jul 2009, Jonathan Moore wrote:
On Wed, Jul 22, 2009 at 10:31 AM, Steve
Edwardsasterisk@sedwards.com wrote:
I finally found a reason TO run Asterisk as root.
By default, ext[23] file
On Wed, Jul 22, 2009 at 8:20 AM, Johann
Steinwendtnersteinwendt...@gmx.net wrote:
Hello !
Are there any plans at Digium to include also german voice prompts ?
I cannot speak on behalf of Digium, but I suspect that if somebody:
* made cogent and sensible German translations of the English
I think siax -from cydia- could also be an alternative as they stated to
use natively 3g. I only test WIFI.
SIAX on WIFI works
SIAX on WIFI works great so far. I don't have a router that i can secure my
network with so I haven't tested it over 3G yet. I plan on doing that
soon. Putting
While I can't be sure this is correct, I'd assume there are 2 pieces
to executing a line of code, the first one does all the expansion and
variable replacement, and the second one actually executes the line.
From the behavior I'd have to guess that INC() is handled by first
part and not the
On Wednesday 22 July 2009 13:56:39 Ira wrote:
Danny Nicholas wrote:
Tilghman Lesher wrote:
On Wednesday 22 July 2009 08:30:03 Danny Nicholas wrote:
You should submit this as a bug. It may or may not get fixed, but it
definitely won't until someone reports it or takes it upon
On Wed, Jul 22, 2009 at 04:33:28PM +0800, Chris YM wrote:
hello:
I can set a environments to test the two pri cards. the patlooptest is ok.
the result has no problem. how do i use patgen and pattest with two pri
cards and the setting of zaptel/sys.conf? in the
Hi guys,
I have a bunch grandstream phones using ulaw and my users are
complaining they are jittery when I use canreinvite=yes. The data
connection is an ADSL link dedicated for phone traffic. At any given
time, I have at most 2 calls in parallel.
I'm not a huge fan of asterisk being in media
Here's what Philipp Kempgen wrote on this topic back in January. Nice
summary, I believe.
Klaus Darilion schrieb:
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#
German
lists a plenty of sound files for German.
Can someone recommend one for Asterisk 1.4 (any
On Wed, Jul 22, 2009 at 12:40:23PM -0500, Tilghman Lesher wrote:
Yes, but on the other side of the switch, a station can detect battery. This
is what the Digium analog cards do in order to report whether a line is in
red alarm status or not. Whether the analog gateways do this or not is
jbmaxsize=80 is way overkill. If your jitter is really close to 80ms then you
have some serious issues on your link and it's not suitable for VoIP at all.
Try jbmaxsize=40.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Kelvin Chan
On Wed, Jul 22, 2009 at 2:41 PM, Geoff Lanege...@gjctech.co.uk wrote:
On Wednesday, July 22, 2009, Catalin S. wrote:
I lookin' for a call in number from UK or USA. Can somebody offers
me a peering for this or specify any sip provider that offers this
thing?
There are several providers who
Hello sorry for earlier message, I push send before write something.
Anyway I tried that sites and also lowratevoip.com.
All gives me the follwing message:
Sorry – at this moment there are no VoIP-In numbers available for
your country (yet). We will inform you as soon as there are (new)
numbers
On Wed, 2009-07-22 at 16:52 +, Jeff LaCoursiere wrote:
So to rephrase it:
One GOOD reason to run asterisk as root is that you get to take advantage
of the default filesystem overflow space reserved for root.
It might be A reason, but it certainly isn't a GOOD one.
A GOOD system
does Asterisk suppoet CSTA protocol for CTI applications?___
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Yes. I am able to match the *67 and appropriately set the
SetCallerPres when SIP phones make calls because the *67 is passed
through and can be matched.
However on my analog handset its as if the *67 is processed and
discarded. Here is my chan_dahdi.conf and a snippet of console output:
If the users' understanding of jitter is technically correct, and they're
complaining about quality issues (due to jitter or packet loss), then
lowering the jitter buffer isn't going to help.
An ADSL link, depending on the sync rate, can have 40+ msec of latency
between the DSL modem and DSLAM.
Can I assume that your project has stalled?
PaulH
logan wrote:
Thanks Paul. Your help is much appreciated here.
I don't really understand this question - Asterisk can make calls over
phone lines. And it does it well.
Surely, Asterisk does that well, but Asterisk needs to have
Hi,
I have come across an odd problem.
Basically I am transferring a call to an agent. The agent is logged in
and set as paused.
In order to find which agent to call I am using a fastagi script to just
set a variable.
When it falls through the agi script and dials the agent (using the
On Wed, Jul 22, 2009 at 4:03 PM, gergis.rasmygergis.ra...@gmail.com wrote:
does Asterisk suppoet CSTA protocol for CTI applications
I'd never heard of it, so I googled it.
http://en.wikipedia.org/wiki/Computer-supported_telecommunications_applications
So, ummm, I can't think of a good synonym
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