Tzafrir Cohen ha scritto:
On Thu, Dec 15, 2005 at 01:45:24PM +0100, Simone Cittadini wrote:
screen -d -m asterisk -vvvcng works well for me, but I'd prefer to run
safe_asterisk in production
Any reason you need to run asterisk in a console?
asterisk -r allows you to view the current
Hi:
I saw a hardware in callshops that attached to
analoge line and begin counting from the time call is
answered to the time it hangup ,So is there
ant hardware or a software added to asterisk to solve
this answering issue?
--- Steve Underwood [EMAIL PROTECTED] wrote:
Andrew Kohlsmith
Hi all,
I wanted to execute one of mySQL query when the call is answered i
tried with the following code but it dones not seems to work.
$AGI-exec('Dial', $dialext);
my $dialstatus = $AGI-get_variable(DIALSTATUS);
hi,
how many mins a month do u have ?
We can give you @ 4 cents a min if u want retail on virtualphoneline.com
On 12/18/05, Tom Vile [EMAIL PROTECTED] wrote:
Looking for a good toll free DID provider.Any suggestions?All ready tried Sellvoip and Gafachi and the experience was not desirable.
*8 is coded in res_features.so .
What are the right extension to dial for pickup calls
between sip=sip or zap=sip ...
Harry
--- Rich Adamson [EMAIL PROTECTED] a écrit :
You might have to use *8#. At least I do with my
7960.
I added callgroup=1 and
Hi All,
Thank you all. As you all mentioned it wasnt so serious and was just a
simple authentication problem. Its been solved.
Regards.
From: Diyanat Ali [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To:
*8 is coded in res_features.so .
What are the right extension to dial for pickup calls
between sip=sip or zap=sip ...
Harry
--- Rich Adamson radamson at routers.com a écrit :
You might have to use *8#. At least I do with my
7960.
I added callgroup=1 and
Hi there,
Any one confronted a crash in asterisk when using mixmonitor app. When i'm
using the mixmonitor app on a briged call as soon as the called party hangs
up the call asterisk crashes and the process terminates with following error
message :
Segmentation fault.
Ouch .. error while
Hi all.
I have an asterisk box on gentoo , and when i try to play MOH, it get
too much volume. At a point that it could damage my ear system :)
If i normalize the music, decreasing the volume, it normalizes again and
play at a volume that i could not use.
What could it be wrong?. In other
I have the same problem here.
It happend after I upgraded my server with Mandriva 2006.
What kernel are you using ?
Hey, I´m trying to *modprobe ztdummy, *but when i make modprobe, return
one
error.
I use kernel 2.4 and have UHCI USB Controller allowed in my kernel.
This problem can be,
On Sun, Dec 18, 2005 at 02:42:21PM +0100, Insider KT wrote:
I have the same problem here.
It happend after I upgraded my server with Mandriva 2006.
which has kernel 2.6 . ztdummy there does not depend on USB.
What kernel are you using ?
Hey, I´m trying to *modprobe ztdummy, *but when i
Hi,
Let's say an office has 20 people with 20 extensions and they want to enter
a code on their phone when they leave for lunch and a voice will tel lthe
caller like:
The person you are calling is out of the office and will return at 1 pm. Is
this something that is possible?
Many thanks,
Gabriel Sartor wrote:
Hey, I´m trying to *modprobe ztdummy, *but when i make modprobe,
return one error.
I use kernel 2.4 and have UHCI USB Controller allowed in my kernel.
http://bugs.digium.com/view.php?id=5236
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to
Christian wrote:
Hi,
Let's say an office has 20 people with 20 extensions and they want to
enter a code on their phone when they leave for lunch and a voice will
tel lthe caller like:
The person you are calling is out of the office and will return at 1
pm. Is this something that is possible?
Hi,
Great, do you know where I can find info about this? Many thanks!
- Original Message -
From: Doug Lytle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 18, 2005 4:02 PM
Subject: Re:
On Fri, 16 Dec 2005, Michael J. Tubby B.Sc (Hons) G8TIC wrote:
All,
I have the following set up:
Fedora Core 4 box (yum updated to current)
Asterisk 1.2.1 + Chan_Capi-cm-0.6.1
AVM C4 card
2 x ISDN2e lines bonded with switchboard number, fax number and 10 x DDI
numbers from British
Just the other day I tried connecting an Avaya
IP403 Office IP PBX to my asterisk.
The IP403 is currently used for all the phones at
our office and it is connected via it's own PRI to
the PSTN.
Now I have a Asterisk machine with three PRIs used
for our SIP services. To be able to utilize our
You should be able to edit prices from within the routes page.
However, you can't set different prices on different brands more
accurately than by using markup. That is one of the reasons that I've
branched / mostly rewritten the product. ASTPP, www.astpp.org, does
provide support for doing
Let's say an office has 20 people with 20 extensions and they want to
enter a code on their phone when they leave for lunch and a voice will
tel lthe caller like:
The person you are calling is out of the office and will return at 1 pm.
Is this something that is possible?
I'm tot shure if
Hi Tyler.
We're registering users with OpenSER, which also routes the calls to a series
of Asterisk systems. The really tricky part is allowing different phones
entering through different Asterisk systems to reach other. Currently, the
solution is to, upon registration from phones, issue a
On Sunday 18 December 2005 10:09, Christian wrote:
Great, do you know where I can find info about this? Many thanks!
There is nothing canned that does this. You need to break the problem down
into sections and implement each section. Elmar's already broken it down for
you.
If you have any
Hi Elmar and all others,
Will have a look and if I can't get it working I will post here!
many thanks!
- Original Message -
From: Elmar Haneke [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 18, 2005
Hello,
Polycom ip soundpoint support ACD login/logout .
Can we configure asterisk with polycom ACD support?
Regards
Harry
___
Nouveau : téléphonez moins cher avec Yahoo! Messenger !
Is there anything I can set or any scripts you guys
have where if it sees certain connections (my upstreams) are down, it attempts
to reconnect them say every minute or 5 minutes? If a provider reloads
something, the connection some times drops and I have to do a "sip reload" to
get it to
gate:/etc/asterisk/.sys# cat astdog.sh
#!/bin/sh
#
#
sleep 60
#
while [ 1 ] ; do
BEZI=`ps auxx|egrep 'asterisk -p'|egrep -v 'grep'|wc -l`;
if [ $BEZI = 0 ]; then `killall -9 mpg123`; `asterisk -p`; fi
sleep 10
done
gate:/etc/asterisk/.sys#
---
turby
Is there anything I can set or any
Hi all,
I just wiped my system and did a clean Asterisk 1.2.0 install with
Bristuff 0.3 Pre 1c. (It doesn't work with 1.2.1 yet!) :-(
Is it my server or is 1.2.0 considerably slower than 1.0.9 was?
It seems to me that all actions take noticably longer than before!
Also, despite setting
I just converted 5 7960's to the latest SIP firmware, used the Cisco example
configuration files, and nothing custom within Asterisk and my message
lights work fine.
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
You have to make sure that the uhciusb driver is not compiled in the
kernel but just loaded as a module, and during boot you could load it
using modprobe before you modprobe ztdummy.
On 12/18/05, Doug Lytle [EMAIL PROTECTED] wrote:
Gabriel Sartor wrote:
Hey, I´m trying to *modprobe ztdummy,
Hi There,
I can suggest you to check the dial status variable in dial plan and if its
NO_ANSWER guide the caller to voicemail with 'u' option, and if they leave
and get back on a fixed time you can take a look for day time night time
topic in asterisk documents.
HTH,
--
M. Shokuie Nia.
Dear pals,
As a matter of fact im serious to know where is the source of echo in a pure
VoIP connection, i think the most of echo problems come from hybrid circuits
which are not an issue in pure VoIP sessions.
Regards.
---
M. Shokuie Nia.
From: Luki [EMAIL PROTECTED]
Reply-To: Asterisk
Dear pals,
As a matter of fact im serious to know where is the source of echo in a pure
VoIP connection, i think the most of echo problems come from hybrid circuits
which are not an issue in pure VoIP sessions.
Regards.
---
M. Shokuie
From: Luki [EMAIL PROTECTED]
Reply-To: Asterisk Users
On Sunday 18 December 2005 14:15, Mohammad Shokuie wrote:
I can suggest you to check the dial status variable in dial plan and if its
NO_ANSWER guide the caller to voicemail with 'u' option, and if they leave
and get back on a fixed time you can take a look for day time night time
topic in
Hi there,
As a matter of fact its an awfull issue specially when you are using auto
announcement systems. As far as i know its possible to solve this problem on
analog boards with tone detection and VAD algorithems but dont think there
is anything out there you can use with asterisk and TDM
Ran into this myself. Portage has a depend for mpg123 but it
installs the one that's also in portage. The one in portage is
broken (read: it doesn't work well with Asterisk). I avoid portage
when it comes to anything asterisk related. I build from source.
The asterisk source has
On Sunday 18 December 2005 14:32, Mohammad Shokuie wrote:
As a matter of fact im serious to know where is the source of echo in a
pure VoIP connection, i think the most of echo problems come from hybrid
circuits which are not an issue in pure VoIP sessions.
Easy. Get better endpoints. In a
Hi:
So i think there is no possible way to terminate
minutes using analoge lines, Is that true?
--- Mohammad Shokuie [EMAIL PROTECTED] wrote:
Hi there,
As a matter of fact its an awfull issue specially
when you are using auto
announcement systems. As far as i know its possible
to solve
Its not a limitation. Its an architectural design which is based on pulse
code modulation (pcm) standards, which essentially says:
- 8,000 audio samples per second,
- each sample is an 8-bit value
- resulting in 64,000 bits/second (like g711 codec standard)
Thank you for your answer, but I
Hello guy´s
I´m trying to create a extension do modify the led colors of
a button on a FOP,
Via manager command, liked PHP, but I do not have a good
result, I have set de astdb family in op_astdb.conf, but never,
My Php script, and my extension.conf is bellow
Thanks for all
By
On Sun, December 18, 2005 20:05, Francesco Peeters (Asterisk) said:
Hi all,
I just wiped my system and did a clean Asterisk 1.2.0 install with
Bristuff 0.3 Pre 1c. (It doesn't work with 1.2.1 yet!) :-(
Is it my server or is 1.2.0 considerably slower than 1.0.9 was?
It seems to me that all
Hi Guys,
I'm trying to send and receive some faxes using iaxmodem and hylafax
through an hfc isdn board and a bristuffed asterisk.
All seems to work fine, but the faxes are sent randomly truncated
without any reported error.
Any idea or suggestion ? Anybody owns a working
Its not a limitation. Its an architectural design which is based on pulse
code modulation (pcm) standards, which essentially says:
- 8,000 audio samples per second,
- each sample is an 8-bit value
- resulting in 64,000 bits/second (like g711 codec standard)
Thank you for your
On Sunday 18 December 2005 18:01, Rich Adamson wrote:
Spec sheets are available for the TigerJet 320 pci chipset as well as
the Silcon Labs 3050, 3210 chip sets used on the TDM card. If you dig
through those I think you'll find that it would difficult if not impossible
to change the card's
[EMAIL PROTECTED] wrote:
Its not a limitation. Its an architectural design which is based on
pulse code modulation (pcm) standards, which essentially says:
- 8,000 audio samples per second,
- each sample is an 8-bit value
- resulting in 64,000 bits/second (like g711 codec standard)
Thank
Hi,
I am trying to configure asterisk to send all calls which come in on the second port of my linksys pap2(which isattached to a fax)to send out my FXO trunk line.I have setup seperate sip profiles for both ports, the second port defaults to a [fax] context.
If I have the context configured as
Spec sheets are available for the TigerJet 320 pci chipset as well as
the Silcon Labs 3050, 3210 chip sets used on the TDM card. If you dig
through those I think you'll find that it would difficult if not impossible
to change the card's infrastructure since its based on the standards
Mantu Jha wrote:
Hi I am have two Asterisk server at two different location one is
having static ip 203.101.42.14 and other is having static ip
10.42.16.1 how can i integrate both so that i can use the others dial
plan.
It's all here on this page.
Then, how about Acer? Does it work well with asterisk?
Simone Cittadini wrote:
Matt Florell ha scritto:
The best Dell for a production environment Asterisk server is no Dell
at all. They make some great workstations, but I've had many problems
with their servers(as have many others on this
How big of RAM for Asterisk server? My production environment will be
about 400 users in the office.
Matt Florell wrote:
The best Dell for a production environment Asterisk server is no Dell
at all. They make some great workstations, but I've had many problems
with their servers(as have many
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date:
12/16/2005
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
Dear comunity:
I use a r2mf converter called SignalPath model 201-SA from Encore
Networks, i configure my Asterisk Box and i receive calls wothout
problem, but when i try to make a Outgoing call, sound a busy signall
after few seconds.
I Think is a lost parameter in my zapata.conf file, but if
Dear comunity:
I use a r2mf converter called SignalPath model 201-SA from Encore
Networks, i configure my Asterisk Box and i receive calls wothout
problem, but when i try to make a Outgoing call, sound a busy signall
after few seconds.
I Think is a lost parameter in my zapata.conf file, but if
Dear comunity:
I use a r2mf converter called SignalPath model 201-SA from Encore
Networks, i configure my Asterisk Box and i receive calls wothout
problem, but when i try to make a Outgoing call, sound a busy signall
after few seconds.
I Think is a lost parameter in my zapata.conf file, but if
Hi All,
I want to control a sip phone from my pc.
I found a draft for this.
http://www.faqs.org/ftp/pub/internet-drafts/draft-mahy-sip-remote-cc-01.txt
Can someone let me know sip phones supporting this
protocol or similar one?
Thanks.
Jason.
__
HI allHow to configure voice mail in asterisk . pls do the needful. regards ramakrishnan.n __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
i have tried
rsgain=100
txgain=100
recording volume improved but still not good
--- Steve Totaro [EMAIL PROTECTED]
wrote:
Hi All
I have a call center working on 8 FXO Channels,
everything working fine except one little problem,
I
am using asterisk queues with
monitor-format =
On Dec 18, 2005, at 12:01 PM, Andrew Kohlsmith wrote:
On Sunday 18 December 2005 14:32, Mohammad Shokuie wrote:
As a matter of fact im serious to know where is the source of echo
in a
pure VoIP connection, i think the most of echo problems come from
hybrid
circuits which are not an issue
On Thu, 2005-12-15 at 09:33 -0800, John Biundo wrote:
I'm particularly worried about acceptance of this shared line (or lack
thereof) aspect of the system. My wife will get the idea of
extensions, transfers, parking, etc. because she uses a PBX at work,
though I worry that the habits of
Hi,
I already contacted what I inputed on my softphone but we
both can't hear each other. I used X-lite and the other is a hardware
SIP phone. What could be the problem?
Thanks,
Ryan
At 03:03 PM 12/16/05, you wrote:
yes
$AGI-exec('Dial', SIP/[EMAIL PROTECTED]);
Diyanat
From:
JP Carballo wrote:
Mantu Jha wrote:
Hi I am have two Asterisk server at two different location one is
having static ip 203.101.42.14 and other is having static ip
10.42.16.1 how can i integrate both so that i can use the others dial
plan.
It's all here on this page.
Tom Lynn wrote:
Their trunk works fine as of the time this email is sent.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
___
--Bandwidth
60 matches
Mail list logo
