On Thu, 2005-12-01 at 22:29 -0800, Luki wrote:
Does anybody know, why it is not possible, to run asterisk within
screen?
Yes, it is possible but you can't scroll up so you only see the last
~40 lines. At least I didn't work for me but I didn't research this
further.
in screen you can
Hi all.
Does anybody use VIP 153
phone with asterisk and has DTMF works.
Thank,
Bob.
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See
http://www.iaxtel.com/setup.html
2005/12/2, Ishanka Anuradha Ranasooriya [EMAIL PROTECTED]:
Hi all,
I have configured two asterisk Boxes.Then I need to communicate
these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via
See
http://www.iaxtel.com/setup.html
2005/12/2, P.G.C.K. Nirukshitha [EMAIL PROTECTED]:
Dear Sir
I have configured two asterisk Boxes.Then I need to communicate these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via asterisk.
Thanks
See
http://www.iaxtel.com/setup.html
2005/12/2, Lakmal [EMAIL PROTECTED]:
Hi all,
I have configured two asterisk Boxes.Then I need to communicate these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via asterisk
Thanks,
Ishanka.
-
so is
there a solution in the next cvs udpate?
Von: René Enskat
[Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag,
1. Dezember 2005 14:47An:
'asterisk-users@lists.digium.com'Betreff: WG: App_rxfax
problem
I just sent the error in full log:
Dec 1 15:01:08 VERBOSE[27950] logger.c:
You can use EAGI
2005/12/2, Alexander Lopez [EMAIL PROTECTED]:
Good Idea. I am doing a similar thing but for a different reason:
I use the system call to bring in mp3 files for music on hold. We make
custom Music on Hold messages and we store them at our DC. I am also
using this to pull mp3
Hi
is there any way i can reduce Bandwidth ussage when iam making outbound??
right now each call taking 80k
ram
On 12/2/05, Giovanni Miano [EMAIL PROTECTED] wrote:
Seehttp://www.iaxtel.com/setup.html2005/12/2, Ishanka Anuradha Ranasooriya
[EMAIL PROTECTED]:Hi all,I have configured two asterisk
2005/12/1, Tony Hoyle [EMAIL PROTECTED]:
The following works in iax.conf for me:
[voipbuster]
host=iax.voipbuster.com
type=peer
username=username
secret=password
qualify=yes
context=inbound
Context=inbound? I'm using from-pstn.
The problem is this: I configure asterisk (through amp)
2005/12/1, Francesco Peeters [EMAIL PROTECTED]:
Mine works just fine. It's a pain though if you have to get a new account,
as minimum amount is now EUR 5... OTOH, for free calls, it might be worth
it...
The account that I'm testing is one of 5 EUR. I'm trying to test it
with asterisk because
2005/12/2, ram [EMAIL PROTECTED]:
is there any way i can reduce Bandwidth ussage when iam making outbound??
right now each call taking 80k
Choose a different codec. The problem is that the codec must be
supported by the other end. I think the better sould be aspeex, or
iblc. But the more
2005/12/2, Giovanni Miano [EMAIL PROTECTED]:
See
http://www.iaxtel.com/setup.html
I'm also interested on doing this. I already set up iax connections to
providers like free world dialup, voipbuster and voipjet, but I don't
know how to configure asterisk to receive the registration from the
On Thu, Dec 01, 2005 at 04:43:25PM -0800, Innocent Evil wrote:
Sorry, to misinterpret.
I also never tried this.
Let me make a simple AGI script that will do this
#!/usr/bin/env ruby
message = ARGV.shift
$stderr.puts \n#{Time.now} #{message}
just put the above lines in a file
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten =
Hi
if you are using
AMP
go to trunk
and start regitering your account
to check the accounts registered
and Go to Console
asterisk sip show registry
ram
On 12/2/05, Alejandro Vargas [EMAIL PROTECTED] wrote:
2005/12/2, Giovanni Miano [EMAIL PROTECTED]: See
On Thu, Dec 01, 2005 at 06:51:54PM -0800, John Todd wrote:
#exec /usr/bin/curl -s
http://webserver.domain.com/privatefiles/username-to-numbers
/etc/asterisk/username-to-numbers
#include username-to-numbers
Nice. However, what happens if curl takes longer than expected? your
reload waits
On Fri, December 2, 2005 9:26, Alejandro Vargas said:
2005/12/1, Tony Hoyle [EMAIL PROTECTED]:
The following works in iax.conf for me:
[voipbuster]
host=iax.voipbuster.com
type=peer
username=username
secret=password
qualify=yes
context=inbound
Context=inbound? I'm using from-pstn.
Are there any Asterisk Users Newsgroup? For me it's much easier to follow
newsgroup then to read all e-mails. Especially with news readers with so many
features.
So, if anybody knows for any newsgroup that has big discussions about Asterisk,
please let me know.
--
Tomislav Parčina
Lama
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
hi guys:
we suffer strange voice shakings after only8
concurrent PSTN calls, any one knows why?
we use g729, which can be cpu intensive, is this
the coz?
Matt
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On Fri, December 2, 2005 9:29, Alejandro Vargas said:
2005/12/1, Francesco Peeters [EMAIL PROTECTED]:
Mine works just fine. It's a pain though if you have to get a new
account,
as minimum amount is now EUR 5... OTOH, for free calls, it might be
worth
it...
The account that I'm testing is
In article [EMAIL PROTECTED],
Nir Simionovich - CTO [EMAIL PROTECTED] wrote:
Hi All,
I have a really funky problem, which I can't seem to pin point.I have a
setup that looks something like this:
SS7 Networks --SS7-- Veraz IGate4000 --SIP-- Asterisk
Now, Asterisk has a second
You'll have to have agents join the queues by issuing the commands
AgentLogin() or AgentCallBackLogin() from an extension in your dialplan.
l.
On Thu, 01 Dec 2005 17:53:19 +0100, gc [EMAIL PROTECTED] wrote:
Thanks. I made change to joinempty=yes. And now I can hear the music on
hold.
I have 3 DID numbers and one E1.
How to limit incoming calls so first DID can accept 10, second 15
and the third 5 councurent calls.
Thanks
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HI, here they are:
--
zapata.conf
[channels]
language=it
;context=incoming
context=default
switchtype=national
pridialplan=unknown
signalling=pri_cpe
echocancel=yes
group = 1
channel = 1-15,17-31
group = 2
channel = 32-46,48-62
group = 3
channel = 63-77,79-93
Hi,
I'm running the last CVS asterisk version (I was running an older version
before with the same problem) and I've a problem with agi scripts. Commands
results are not always correct.
I've made a small agi test script that execute ChanIsAvail on an inexistent
extension:
On Fri, Dec 02, 2005 at 11:00:34AM +0100, FaberK wrote:
HI, here they are:
--
zapata.conf
[channels]
language=it
;context=incoming
context=default
switchtype=national
pridialplan=unknown
signalling=pri_cpe
echocancel=yes
group = 1
channel = 1-15,17-31
On Fri, Dec 02, 2005 at 09:32:35AM +0800, Marcus Deluigi (intern) wrote:
Hi!
I downloaded asterisk 1.2.0 and compiled it myself.
The default behaviour is that calling 'asterisk' will return the prompt
and calling 'asterisk -v' is returning the CLI.
I want to run asterisk within screen,
PRITRUNK1 is defined into the extensions.conf globals:
--
[globals]
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
PRITRUNK3=Zap/g3
--
Well I know what's happening, from my asterisk CDRs, and also from the PRI-CDRs.
We use a Teles.
2005/12/2, Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, Dec 02,
Has anyone used BT's DSS services?
http://www.bt.com/isdn/isdn2e/extra/dss.htm
I have an ISDN 2e comming into my Asterisk and would like to deflect
calls when I am busy (or I can't get my HFC-PCI card to run correctly
LOL) to my PSTN-IAX VOIP number if the Asterisk doesn't answer.
Listening
Tzafrir Cohen wrote:
/usr/bin/screen -L strace -f -o /tmp/trace /usr/sbin/asterisk -v
and I have no screen session running and I also have no asterisk CLI to
connect to.
I can't explain the behaviour and the screenlog is empty.
permissions? If that is what you suspect, strace the
In article [EMAIL PROTECTED],
Tomislav Parèina [EMAIL PROTECTED] wrote:
Are there any Asterisk Users Newsgroup? For me it's much easier to
follow newsgroup then to read all e-mails. Especially with news readers
with so many features.
So, if anybody knows for any newsgroup that has big
Dear guys
My asterisk is giving some error as bellow with some extention.Have any
body
received this type of error.
CDR updated on SIP/200-a6a5
-- Executing Goto(SIP/200-a6a5, ivr-main|s|3) in new stack
-- Goto (ivr-main,s,3)
-- Executing BackGround(SIP/200-a6a5,
Hi,
Do you know from where i can buy g723 codec. for g729 i can buy it
from digium.com. But Please let me know from where i can get g723
codec.
And the codecs purchasing can solved my problem?
--
Thank You,
Code Lover
___
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2005/12/2, ram [EMAIL PROTECTED]:
if you are using
AMP
go to trunk
and start regitering your account
Humm... what I'm trying to do, and what is this thread subject, is to
connect asterisk-to-asterisk.
Then... I go to trunks, create a new iax trunk, invent some
user/password, use the ip of
Try print EXEC ChanIsAvail IAX2/24\n;
Channel type is IAX2 not IAX
Cheers
2005/12/2, Cyrille Demaret [EMAIL PROTECTED]:
Hi,
I'm running the last CVS asterisk version (I was running an older version
before with the same problem) and I've a problem with agi scripts. Commands
results are not
You can use global var and same if condition
Cheers
2005/12/2, omadon [EMAIL PROTECTED]:
I have 3 DID numbers and one E1.
How to limit incoming calls so first DID can accept 10, second 15
and the third 5 councurent calls.
Thanks
___
- Check int call on IRQ
- Check cpu usage
Good Luck
2005/12/2, Matt [EMAIL PROTECTED]:
hi guys:
we suffer strange voice shakings after only 8 concurrent PSTN calls, any one
knows why?
we use g729, which can be cpu intensive, is this the coz?
Matt
hi, I am a newbie to asterisk. I am tryining to connect two sip based soft X-Lite phones to an asterisk server. i made following settings in sip.conf: [general] port=5060 bindaddr=0.0.0.0 allow=all context=bogon-calls [2000] type=friend username=2000 secret=tejas host=dynamic
Is there any way to create various trunks with the same priority.
I'm interested on usingo 2 trunks, but balancing the usage in both
because both has a number of free minutes. If I give preference to one
over other, this one will exceed the free limit much before the other.
--
Alejandro Vargas
I has varios access to pstn but each one has different hours in the
day when the calls are free. Then I want to change the priority with
the time of deay in order to make asterisk to prefer the one where the
calls are free.
Is there an easy way to do this? If there is not, I can place a script
in
Hi Tejas,
what context are the extensinos you included below in your
extensions.conf file?
Tejas Shah wrote:
hi,
I am a newbie to asterisk. I am tryining to connect two sip
based soft X-Lite phones to an asterisk server. i made following
settings in sip.conf:
[general]
port=5060
Alejandro,
You could do something like:
[balance]
exten = _X., 1, Random(50:4)
exten = _X., 2, Dial(Zap/g1/${EXTEN})
exten = _X., 3, Congestion
exten = _X., 4, Dial(Zap/g2/${EXTEN})
exten = _X., 5, Congestion
See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random
Alistair
Official Polycom view seems to be that you shouldn't work at night :)
The phones are crying out for a backlit LCD that only lights when
ambient light is low. I have a cheap radio/weather station with a
large LCD that does that.
___
--Bandwidth and
I have. I have not noticed any major problems with my 800 DIDs or
outgoing with them for about a year now. I don't use them very much
though
-Original Message-
From: Bill Michaelson [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 01, 2005 2:25 PM
To: Asterisk Users Mailing List -
Hi,
I've changed that and it's the same problem. I've this problem with all
applications. Results from agi are not correct.
Regards,
Cyrille
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Giovanni
Miano
Envoyé : vendredi 2 décembre 2005 12:52
À :
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
Yep, tried APIC, NOAPIC, ACPI=OFF, etc. (capitals only for clarity!) but
to no avail! As soon as both share the same IRQ, the zaphfc driver stops
passing data to asterisk...
It is supposed to when you are using APIC, you should obtain many
Alejandro Vargas wrote:
I has varios access to pstn but each one has different hours in the
day when the calls are free. Then I want to change the priority with
the time of deay in order to make asterisk to prefer the one where the
calls are free.
Is there an easy way to do this? If there is
My aterisk is working now. I had some spelling
mistakes in queues.conf.
Thanks for your help.
- Original Message -
From:
Dov Bigio
To: gc ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 12:22
PM
Subject: Re:
Ouch ... error while writing audio data: : Broken pipe
If you are talking about the Ouch message, yes lots of people have seen
the error and its usually the result of some misconfiguration in one of
your files (likely zapata.conf).
Correct me if I'm wrong, but isn't that message from mpg123
Hello to all, i have now two questions
the first is, anybody know some software to emulate a ip phone? or a soft
phone ip to work with asterisk in other computer ?
and the other is
how i do to install 1 rpm from my cd rom?
i accessed with the root password but i navigate for all the
But i have this in astewrisk log:
Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so]
Dec 1 15:01:08 WARNING[27950] loader.c:
/usr/lib/asterisk/modules/app_rxfax.so: undefined symbol:
fax_set_phase_d_handler
Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so
failed!
Dan,
Thanks - that helps...
Now when I run it, I hear my headset ring or beep for the incoming call
and when I answer it I dont get any audio.
I have a kensington dongle and a plantronics head set.
Jerry
Hi Jerry,
- Original Message -
From: "Jerry Geis" geisj at
wath ip hardware phones are the recomended to work with asterisk?
or any phone work fine?
__
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Tutopia es Internet para todos.
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On a polycom 600 which is working perfectly otherwise, I am
unable to use DTMF with IVR or such - not even to dialout of a Sipura setup
elsewhere. Other phones (analogue connected to ATA) are accepted.
I suspectthe phone is not using rfc2833 but I don't know
how to specify that it should
On Fri, 2005-12-02 at 10:24 +0100, Francesco Peeters wrote:
[snip]
(The 0031x are set up in this manner to avoid Cellphone (0031-6.) and
Premium (0031-8. 0031-9.) numbers.)
Afaik 0031-8. are freephone numbers, not premium.
Regards,
Patrick
___
I'm currently using X-Ten as a softphone and I've been having issues
with dialing into IVR's. It seems that my DTMF passes in chirps and not
clear tones. Any solutions?
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Asterisk-Users mailing
I have bout ISDN 4-port BRI Sirrix.PCI4S0 card and I'm unable to make it work.
I have followed instructions that you get with drivers but I'm unable to start
Asterisk ([chan_sirrix.so] Program exited with code 01. Warning, flexibel rate
not heavily tested! Ouch ... error while writing audio
On Sat, Nov 26, 2005 at 10:37:44AM -0500, Tom Rymes wrote:
More specifically, you can make it work using an ATA or a TDM400P
card with an fxs port, but it is not likely to be reliable. If you
TDM400P (FXS) and some ISDN quad bri Card works fine for
approximately 10 months in a small
On Fri, December 2, 2005 15:08, Patrick said:
On Fri, 2005-12-02 at 10:24 +0100, Francesco Peeters wrote:
[snip]
(The 0031x are set up in this manner to avoid Cellphone (0031-6.) and
Premium (0031-8. 0031-9.) numbers.)
Afaik 0031-8. are freephone numbers, not premium.
Regards,
Patrick
Hi
Just wandering what solution worked to eliminate echo on your setup.
I am trying every solutions I can find on the wiki and none is working
perfectly.
We have asterisk 1.2.0
3 x digium TDM400P
30 Snom320 + 5 Snom360
For now the best setup I have is using Mark2 Echo cancel.
Thanks
On Fri, December 2, 2005 14:00, Alejandro Vargas said:
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
Yep, tried APIC, NOAPIC, ACPI=OFF, etc. (capitals only for clarity!) but
to no avail! As soon as both share the same IRQ, the zaphfc driver stops
passing data to asterisk...
It is supposed
2005/12/2, Ishanka Anuradha Ranasooriya [EMAIL PROTECTED]:
Hi all,
I have configured two asterisk Boxes.Then I need to communicate
these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via asterisk
I've found the answer:
Patrick Fortin wrote:
Just wandering what solution worked to eliminate echo on your setup.
I am trying every solutions I can find on the wiki and none is working
perfectly.
I have been (since 1 1/2 weeks) using the ECHO_CAN_MG2.
We have got a different setup (quadBRI, 12 GXP-2000's), and
Hello,
I have a queue for incoming calls with some agents (defined as Agents)
using iax2 softphone. I would like to use the Attended transfer (ATXFER)
feature, however app_queue cannot handle it (I guess because it is not a
channel).
For this reason I put a Local channel in between with /n
I thing u cant use ChanIsAvail with exec command
... as use EXEC DIAL(SIP/40) .. it isnt work
2005/12/2, Cyrille Demaret [EMAIL PROTECTED]:
Hi,
I've changed that and it's the same problem. I've this problem with all
applications. Results from agi are not correct.
Regards,
Cyrille
This must be similar to a problem I have seen here. Some times the main
operator's phone will stop ringing when a call comes in on the queue
while the other phones still ring. I have to reset her phone which
causes a re-login to get it working again. It must stop after she does
an attended
One simple way to overcome this problem would do to make an attended
transfer to check whether the receiving person is available and willing to
take the call, and then an unattended transfer to discharge the operator
of the call.
l.
On Fri, 02 Dec 2005 16:21:39 +0100, James Armstrong
we should be getting a limited number in a couple of weeks time.
Proper stocks will be arriving in January - www.provu.com
Paul.
Senad Jordanovic wrote:
[EMAIL PROTECTED] wrote:
There is a review on the homepage at http://voipspeak.net
It has been available for a few weeks, it
Is there a way in asterisk to configure a sip invite timeout ? It seems
to be about 30 seconds right now which is too long. I would like to
have asterisk return congestion if a host does not respond to an invite
within 5 seconds.
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I am using http://www.gmane.com/ with my newsreader.
You still have to be a list member to post.
You can then turn on the vacation option in the list manager to stop
receiving emails.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
Matthew Simpson wrote:
Is there a way in asterisk to configure a sip invite timeout ? It seems
to be about 30 seconds right now which is too long. I would like to
have asterisk return congestion if a host does not respond to an invite
within 5 seconds.
Asterisk 1.2 will use a T1 timer
Hi list,
I have a problem with a SIP trunk on my * box: I can
originate calls but I cant receive them.
The * box is behind a modem-router and as a private
address.
I think about a NAT problem but I dont know
how to resolve it.
I have included some debug and configuration.
The
Hello,
extensions.conf:
[mytest-in]
exten = 1,1,NoOp(${MYVAR1})
exten = 1,n,Wait(20)
exten = 1,n,Hangup()
[mytest-out]
exten = 1,1,NoOp(${MYVAR1})
exten = 1,n,Dial(Zap/g1/06111,10,H|g)
my test dial.out file:
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
Context: mytest-in
Extension: 1
Ouch ... error while writing audio data: : Broken pipe
If you are talking about the Ouch message, yes lots of people have seen
the error and its usually the result of some misconfiguration in one of
your files (likely zapata.conf).
Correct me if I'm wrong, but isn't that message
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference and do something when the user count reaches zero... And
of course, I can do that directly from the
Just wandering what solution worked to eliminate echo on your setup.
I am trying every solutions I can find on the wiki and none is working
perfectly.
We have asterisk 1.2.0
3 x digium TDM400P
30 Snom320 + 5 Snom360
For now the best setup I have is using Mark2 Echo cancel.
I'm
Hello AllI'm bought VoiceTronix Card (Openswitch), it's bad card and resaller (www.telephonyware.com) give me are old card (for one year old). than, after that, my card is fault. I didn't received any help from telephoneware or voicetronix. I don't like voicetronix and telephoneware. i notice
Guys,
I'm curious if it's possible to asterisk at home and the sangoma T1 cards together. I realize asteriskathome is traditionally used for at home, but I'd like to use it in a small office with a T1 and our hardware is a Sangoma card. I know all I need to do to get the sangoma working is
Jess Coburn wrote:
Guys,
I'm curious if it's possible to asterisk at home and the sangoma T1
cards together. I realize asteriskathome is traditionally used for at
home, but I'd like to use it in a small office with a T1 and our
hardware is a Sangoma card. I know all I need to do to get the
Thanks Saul,
What you do to get the Sangoma to install and how'd you go about compiling the zaptel source did you just download zaptel and extra RPMs? I'm by no means a linux guru...
Jess
On 12/2/05, Saul Diaz [EMAIL PROTECTED] wrote:
Jess Coburn wrote: Guys, I'm curious if it's possible to
@home by no means means it just for the home - its Asterisk nothing more, nothing less. I don't think the @home designation was meant to limit it by perception. I read somewhere it was called @home for another reason, anyone know more?
RegardsRobOn 12/2/05, Jess Coburn [EMAIL PROTECTED] wrote:
Title: Music on Hold Error
Can anyone help with;
Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3'
Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player
Dec 2 12:20:16 WARNING[2562]:
In article [EMAIL PROTECTED], John Daragon [EMAIL PROTECTED] wrote:
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference and do something when the
Before 1.2.0 I used Mark2 with AGGRESSIVE turned on
I would recommend switching to KB or MG in 1.2.0, we have done this with
very good results (using KB now)
Jared Armstrong
-Original Message-
From: Patrick Fortin [mailto:[EMAIL PROTECTED]
Sent: Friday, December 02, 2005 9:17 AM
To:
Dave Morrow wrote:
Can anyone help with;
Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no
files in '/var/lib/asterisk/mohmp3'
Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:488 monmp3thread:
Unable to spawn mp3player
Dec 2 12:20:16 WARNING[2562]:
Has anyone encountered 'bad' cdr logging in * 1.2?
Since upgrading to 1.2 (bristuffed) and asterisk-addons 1.2, sometimes
the clid is 'messed' up. I use AMP to look at the reports, but when I
look in the cdr database, it's the same, here's an example:
2/12/2005 15:06:02 Tech: ÀB ÀB 2
(ENGLISH VERSION AT THE END)
Hola lista:
Requiero saber si alguien tiene un cliente o empresa donde se encuentren montado algun
Asterisk como PBX de tamaño mediano (al menos unas 50 extensiones). Esto para dar una
demostracion a un cliente mio que esta interesado en invertir en Asterisk.
Les
Hello,
I am trying to convert my hint priorities from the old style:
exten = 2130,hint,SIP/0146472130
to Asterisk Extension Language (AEL) style.
I haven't found anything in the docs, wiki or examples about it.
How should I do it?
--
Sigs have been known to cause cancer in California.
Tony Mountifield wrote:
In article [EMAIL PROTECTED], John Daragon [EMAIL PROTECTED] wrote:
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference and
How could a CVS update fix an error you have made during installation?
Steve
René Enskat [Teamware GmbH] wrote:
so is there a solution in the next cvs udpate?
*Von:* René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED]
*Gesendet:* Donnerstag, 1. Dezember 2005 14:47
*An:*
Could you send it patch please.
On 11/30/05, Paradise Dove [EMAIL PROTECTED] wrote:
btw, i've patched this part of code and now its working fine for me.i'm going to upload it.Paradise Dove
On 11/30/05, Kevin Hanson [EMAIL PROTECTED] wrote: Paradise Dove wrote: Yes with version 1.2. I have tried
Louis-David Mitterrand wrote:
to Asterisk Extension Language (AEL) style.
I haven't found anything in the docs, wiki or examples about it.
I don't believe hints are supported in AEL at this time.
___
--Bandwidth and Colocation provided by
Dean Collins wrote:
who's done it? and how much money are they talking about? I've been
looking to pay for something like that for a while.
-Original Message-
From: Neil Stratford [mailto:[EMAIL PROTECTED]
Sent: 24 November 2005 09:30
To: John Martin; [EMAIL PROTECTED]
Subject: Re: Fwd:
On Thu, Dec 01, 2005 at 06:51:54PM -0800, John Todd wrote:
#exec /usr/bin/curl -s
http://webserver.domain.com/privatefiles/username-to-numbers
/etc/asterisk/username-to-numbers
#include username-to-numbers
Nice. However, what happens if curl takes longer than expected? your
reload waits
Hi list:
I'm having problem with some DIAXY ATA FROM DIGIUM, I have 3 of them in different points, all of them register
to a central asterisk server. If i call from any of the ATA's to Asterisk or Asterisk's to ATAs. But when any ATA's want to talk
to another ATA's.. TheATA's rings, but when the
On Fri, Dec 02, 2005 at 12:05:08PM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
to Asterisk Extension Language (AEL) style.
I haven't found anything in the docs, wiki or examples about it.
I don't believe hints are supported in AEL at this time.
Thanks for the heads-up.
I was experimenting with ael and first thing I tried to do was move the
inclusions for the default context form the extensions.conf file to the
extensions.ael file
Can a context that is defined in extensions.conf be included by the ael
parser?
Just asking in case anyone has already discovered
Hello all,
I recently upgraded the kernel on one of the phone servers I have at home
(dual Xeon 2.4) from 2.6.11 to 2.6.14 in the usual way, copying the .config
file across and building the new kernel. Now ztdummy is refusing to run, and
gives the following errors in dmesg:
ztdummy: Unknown
At 10:16 AM -0600 12/2/05, Kevin P. Fleming wrote:
Matthew Simpson wrote:
Is there a way in asterisk to configure a sip invite timeout ? It
seems to be about 30 seconds right now which is too long. I would
like to have asterisk return congestion if a host does not respond
to an invite
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