Hello all,
i followed for the following step.
1)extract AsteriskWin32-0.60-Setup.exe into cygwin folder
2)extract asterisk-1.2.14 into /usr/src
3)applied the patch like that patch -p0 < awin32-0.60.patch
(here it will ask YES or NO.Is it possible to give YES recursively?)
4)make
that time i
Stephen Bosch wrote:
Bob Smither wrote:
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
Hi,
I have to set call length to 3min, but before hangup have to warn
caller. There are many IVRmenu and submenu options with different
warning audio.
I have to measure somehow the audio file length
On Thu, 12 Apr 2007, Greg Woods wrote:
Aside from this, I love my new asterisk system, and my wife has almost
gotten used to having to dial 9 to get out of the house :-)
Can't help you with your zaptel & ssh issues - I use them both on my
systems without any issues at all.
But why force you
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Alejandro,
SSH to your box like this:
ssh [EMAIL PROTECTED] -L 8080:127.0.0.1:8080 (in Putty it's something called
portforwarding)
This will bind 127.0.0.1:8080 from your asterisk-box to 127.0.0.1:8080
of your local box. So you can access the websi
On Thu, Apr 12, 2007 at 10:17:25PM -0700, Yuan LIU wrote:
> >From: Tzafrir Cohen <[EMAIL PROTECTED]>
> >Date: Thu, 12 Apr 2007 09:18:46 +0300
> >
> >On Wed, Apr 11, 2007 at 08:09:16PM -0700, Yuan LIU wrote:
> >> >From: Sanjay Rajdev <[EMAIL PROTECTED]>
> >> >Date: Thu, 12 Apr 2007 01:29:51 +0530 (I
On Fri, Apr 13, 2007 at 12:32:06AM -0400, dave cantera wrote:
> I had a similar problem... I forget exactly how I resolved it because it
> happened twice... here is the solution from memory.
> the sequence of the zaptel, libpri, and asterisk is important. if you
> compile zap before libpri, zap
alex,
thanks for the note... oh well, fun times ahead! :)
daveC
Alex Balashov wrote:
Dave,
On Fri, 13 Apr 2007, dave cantera said something to this effect:
alex,
I have been considering linux clustering for *... am not ready today
and expect it in about 6-9 months... was wondering if anyone
Hi,
I just compiled and installed Asterisk-1.4.2 along with zaptel-1.4.1
and libpri-1.4.0 on a Debian machine with a TDM400P card.
Everything goes ok but when I try to make a call through the ZAP
channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and
zttool show the card correctly
From: "LKS GMAIL" <[EMAIL PROTECTED]>
Date: Thu, 12 Apr 2007 13:02:24 +0200
Hi guys!
Im using Asterisk 1.2 with mISDN support.
I have problems with Pickup calls with my Grandstream Buttons . I set up on
Dial Plan this:
Exten => _**XXX,1,Pickup(SIP/{EXTEN:2}) but it doesnt work if the call
co
From: Tzafrir Cohen <[EMAIL PROTECTED]>
Date: Thu, 12 Apr 2007 09:18:46 +0300
On Wed, Apr 11, 2007 at 08:09:16PM -0700, Yuan LIU wrote:
> >From: Sanjay Rajdev <[EMAIL PROTECTED]>
> >Date: Thu, 12 Apr 2007 01:29:51 +0530 (IST)
> >
> [good stuff sniffed]
> >and downloaded zaptel 1.4.1, after that e
From: ismir saljic <[EMAIL PROTECTED]>
Date: Thu, 12 Apr 2007 07:42:13 -0700 (PDT)
Hi all,
I have asterisk 1.2.13 and problem is about DTMF.When i have incoming call
on Toll-Free number asterisk accept DTMF digits but dial only first in
context.
Per instance:
When i press 1 it is OK,but when
From: pedro noticioso <[EMAIL PROTECTED]>
Date: Thu, 12 Apr 2007 12:02:52 -0700 (PDT)
Hi there list!
I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a bus
From: "Ronaldo Zacarias Afonso" <[EMAIL PROTECTED]>
Date: Thu, 12 Apr 2007 11:54:51 -0300
Hi all,
Is it possible to configure an extension number to dial a sip address?
Nothing prevents you from doing this.
Yuan Liu
For example:
exten => 101,1,Dial(SIP/sip:[EMAIL PROTECTED])
That way I ca
Hi,
I am newbie to asterisk. I would like to use asterisk using VoIP. I don't
want to use any hardware. I have installed Asterisk 1.2.13. I would like to
record using AGI command RECORD FILE. I would also like to do conferencing
and recording in asterisk. How could this be done? Help me out with
Dave,
On Fri, 13 Apr 2007, dave cantera said something to this effect:
alex,
I have been considering linux clustering for *... am not ready today and
expect it in about 6-9 months... was wondering if anyone had put * on a
cluster and what experiences they gained? have you done this?
dav
alex,
I have been considering linux clustering for *... am not ready today and
expect it in about 6-9 months... was wondering if anyone had put * on a
cluster and what experiences they gained? have you done this?
daveC
Alex Balashov wrote:
On Wed, 11 Apr 2007, Andrew Joakimsen said someth
I had a similar problem... I forget exactly how I resolved it because it
happened twice... here is the solution from memory.
the sequence of the zaptel, libpri, and asterisk is important. if you
compile zap before libpri, zap doesn't know how to make it and then
asterisk doesn't include it beca
brandon,
engineers are BIB on monitoring... count me as one who wants to monitor!!!
daveC
Brandon Kruse wrote:
Yes,
I have actually written a resource module for asterisk and the gui to
use rrdtool to make REAL pretty gradient shaded graphs based on asterisk
data.
So, if you want the cacti scr
The media streams are still RTP, right?
--
Alex Balashov <[EMAIL PROTECTED]>
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/aste
On 4/12/07, J. Oquendo <[EMAIL PROTECTED]> wrote:
> Drew Gibson wrote:
>> The Aastra 480i is a good quality phone, on par with Cisco and probably
>> with Polycom (though I've never used them). Voice quality is good,
phone
>> feels robust. Config is well documented and contained in two text
check your sip.conf, or it might be a UA misconfigured with an IP
and/or external IP address.
On 4/12/07, Mike <[EMAIL PROTECTED]> wrote:
Hi,
I`m getting this (from one of my registered phone that has been installed at
some location I can`t access at the moment) in the Asterisk CLI. Notice t
On 4/12/07, J. Oquendo <[EMAIL PROTECTED]> wrote:
Victor Hoodicoff wrote:
>
>
> I think your impressions of Aastra are outdated. Install the latest
> firmware, download the latest documentation and test and THEN give an
> opinion!
Did you miss the part when I wrote I have Asstras sitting on my
Can you do a packet capture and see what the actual contact (Via) in fact
says right before it hits Asterisk?
On Thu, 12 Apr 2007, Mike said something to this effect:
Hi,
I`m getting this (from one of my registered phone that has been installed at
some location I can`t access at the moment)
Hi,
I`m getting this (from one of my registered phone that has been installed at
some location I can`t access at the moment) in the Asterisk CLI. Notice the
last 3 digits of the IP address in the error message:
Apr 12 23:06:16 WARNING[25593]: chan_sip.c:7036 check_via: '67.39.117.611'
is not a
On Thu, 12 Apr 2007, Mike Lynchfield said something to this effect:
No you are being misled.. SER can NOT DO IAX, SER = SIP only
No, you are not being misled. In my most recent response, I was
referring to the specific dialplan example for rerouting that you provided
that used an IAX chan
No you are being misled.. SER can NOT DO IAX, SER = SIP only
but you would need SER to do that yes.
On 4/12/07, Alex Balashov <[EMAIL PROTECTED]> wrote:
Certainly. Any signaling / trunking protocol will do, in principle.
On Thu, 12 Apr 2007, Edgar Guadamuz said something to this effect:
> T
No entirely true.. but yeah realizing we are talking about asterisk .. you
can't do reliable T38 faxing..
check out openpbx its embeded , all you need then is a ITSP to do TDM
termination as T38
A bit more expensive then voip.. but you still get the bulk /automation part
without the requirements
Greg Woods wrote:
I hope I don't get flamed the first time I post to a new list. I have
spent a couple of hours poking around without seeing anything like this.
The problem is, as soon as I load the Zaptel drivers (with a TDM-31B
card), ssh into or out of the server is broken. Trying to ssh in,
I have been trying to setup a PAP2 adapter on a remote network but can't
seem to get it to work. The unit will register with the server and it can
make calls to extensions on the Asterisk server but it can't receive any
calls and it can't make any calls outside of the Asterisk server.
I also h
Victor Hoodicoff wrote:
>
>
> I think your impressions of Aastra are outdated. Install the latest
> firmware, download the latest documentation and test and THEN give an
> opinion!
Did you miss the part when I wrote I have Asstras sitting on my desk
collecting dust. I program on average about 5
Certainly. Any signaling / trunking protocol will do, in principle.
On Thu, 12 Apr 2007, Edgar Guadamuz said something to this effect:
Thank you Alex and It would be possible to do that using IAX too,
wouldn't it?
I mean something like
exten=>_9NXX,1,Dial(Zap/g0/${EXTEN:1})
exten=>
Hi there,
I am new to this ML. Recently I started working on Asterisk 1.4 + RAGI +
Ruby on Rails to create a call history browser.
To record call history, I am trying to capture dialup, answer and hangup
events. To check what status a call is, I use channel_status() that RAGI
provides.
I am havi
Hi All
Has anyone managed to get Asterisk 1.2 faxes working reliably with
spandsp 0.0.3? I am running Asterisk 1.2.17 and spandsp 0.0.3pre28 with
a Digium b410p card. Everything compiled smoothly but only about 70% of
faxes come through ok. Debugging shows nothing more than: app_rxfax.c:
F
Has anyone tried to pass sccp through a "cheap" router / nat box?
I have gotten sccp to go through a cisco pix just fine, but I can't seem
to get it to go through a ipfilter box or a basic netgear / linksys
router. I was under the impression that sccp was a lot more nat
friendly, but at the mome
I hope I don't get flamed the first time I post to a new list. I have
spent a couple of hours poking around without seeing anything like this.
The problem is, as soon as I load the Zaptel drivers (with a TDM-31B
card), ssh into or out of the server is broken. Trying to ssh in, I get:
RSA_public_d
Thank you Alex and It would be possible to do that using IAX too,
wouldn't it?
I mean something like
exten=>_9NXX,1,Dial(Zap/g0/${EXTEN:1})
exten=>_9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
___
--Bandwidth and Colocation provid
On Fri, 2006-09-29 at 11:12 -0500, Pato Valarezo wrote:
> Hi, sorry for the question, i've been searching for a real time billing
> system for asterisk with zap/sip support, for use in post paid systems
> like "locutorios", do you know of or use any ?
>
Give a try to StarshopOSS:
http://www.s
Ken,
You have certainly had experience with a broader range of phones, so I have
no doubt you can lend more insight on this count.
But for what it's worth, my experience is largely confined to the Cisco
7960s. I've never had any trouble getting any SIP firmware image to
register with Aster
Another way is to run the calls through a SIP proxy such as SER which can
hunt through two Asterisk UA endpoints, depending on a variety of
parameters including failure at a primary and fallback to a secondary.
--
Alex Balashov <[EMAIL PROTECTED]>
_
I've had experience with quite a few different phones, so I think I'm
qualified to drop my two cents:
Alex is quite right that the Cisco phones are only designed to be used
with Cisco Call Manager. They are capable of being decent SIP
telephones, but Cisco won't provide the documentation so that
On Thu, 12 Apr 2007, Edgar Guadamuz said something to this effect:
I've been looking for a way to share trunks between two asterisk servers.
Provided that the Asterisk servers can be set up to hold identical
SIP contacts (URIs), you can just set up a dialplan such that it fails
over if a pri
Hello eveybody,
I've been looking for a way to share trunks between two asterisk
servers. I guest I have to use Dundi, but I've not found the exact
method yet. I need a way to allow users registered in one server to
use the another server's trunks in the case the first server's trunks
were busy
On 4/12/07, Brandon Kruse <[EMAIL PROTECTED]> wrote:
Hey guys,
What are some of the numbers you guys want graphed?
Curious how you are going to do this and will it be backwards portable. One
of our engineers wrote an app that queries the manager interface to build
RRD data. That's sent over
Hi,
It's really a simple question!
I've just started playing with asterisk too, and I think what you want
could be found in the 4th chapter of "Asterisk: The Future of the
Internet". It's a open book you can download from
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11.
I hope it
Hey guys,
What are some of the numbers you guys want graphed?
Anything that is a number, or any kind of information.
Now I have
Agents logged in and out
# of queues
total calls
total channels
What else?
___
--Bandwidth and Colocation provided by
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Asterisk Gurus!
I have a very simple question. I've just started playing around with
Asterisk and BSD box. I also have grandstream ip phone and installed
asterisk from ports. Now I'm on my very first steps to configure
Asterisk. The question is:
Thanks all... Looks like I will have to let them know that FOIP is a no
go and that we can automate on Asterisk though...
Thanks!
Wiley E. Siler
Director of Information Technology
Education 2020
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
(866) 320.2083 ext. 1003
mailt
Hum, I know Stefan, he is an asterisk-java dev, but he is not online
right now, I will let him know ASAP. Thanks!
On 4/12/07, Doug Garstang <[EMAIL PROTECTED]> wrote:
Does anyone know who maintains the Asterisk-java web site at
asterisk-java.org? The site seems to have been unavailable for a cou
Hi there list!
I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a busy signal, something
I don't like
You can search for the word irc to see my comments,
th
> Drew Gibson wrote:
>> The Aastra 480i is a good quality phone, on par with Cisco and probably
>> with Polycom (though I've never used them). Voice quality is good,
phone
>> feels robust. Config is well documented and contained in two text files
>> (one global, one MAC specific). Good web int
Jessee J Holmes wrote:
> Got off the phone with Polycom on this I have the same problem with
> my new 601 phone here (haven't seen the problem on the 650).
I am using an IP650 with the latest firmware (and the corresponding
sip.cfg file) and I have seen this behavior. It is most noticeable w
Hi,
When I tried to use speex (8 khz) codec I got
following warning messages on the Asterisk console.
The other end was pjsip and I was testing this in
local network.
Here is a exact message:
WARNING[6055]: codec_speex.c:237 speextolin_framein:
Out of buffer space
Has anybody had success in usi
Alberto Pastore wrote:
But why does 8.6 seem to work with previous asterisk 1.2.13??
That I wouldn't be able to answer.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety."
_
Wiley Siler wrote:
Basically, I want to send bulk faxes to a list of my clients.
It is time consuming for a person to individually fax so a blast type
solution seems best.
Over IP is of course to save money...
Thanks for the link, reading now...
Any suggestions for the blast then?
My sug
Either analog modems or a PRI, and Hylafax for automation, no VOIP
involved there.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, April 12, 2007 10:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [as
Does anyone know who maintains the Asterisk-java web site at
asterisk-java.org? The site seems to have been unavailable for a couple
of days now.
Doug
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCR
People, I have a Debian box with Asterisk and I've installed the Destar
package in order to get web managing of my voip system.
After I installed Destar, it runs on "localhost:8080", but my server
does not have X-Window to access to it so I can engter the web interface..
So how can I change localh
Eric "ManxPower" Wieling wrote:
> I'll be sending Digium support the info they requested later today. I
> hope it helps.
We have a developer working on extending Zaptel to support pre-echo
audio capture right now, so that we can work on debugging these issues
with real data instead of just conjec
Dear Jason,
Here in my company we use an applet it java IAX, and it functions very well!
If to want to visit the URL is http://www.virgos.com.br, calls the service
as 0800Web.
Leonardo Silva
2007/4/5, Jason Wolfe <[EMAIL PROTECTED]>:
I need to decide on the best way to add a voip SIP or IAX
Noah,
I am just using a dlink router for dhcp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Wednesday, April 11, 2007 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom - Static I
Stephen Bosch wrote:
Lee Jenkins wrote:
Stephen Bosch wrote:
Sidetone can be set in the phone configuration; before you do that,
though, I need to know what you mean by feedback.
Sorry, should have been more detailed. It's a sort of background
humming noise, almost like that if you placed th
Mike,
Got off the phone with Polycom on this I have the same problem
with my new 601 phone here (haven't seen the problem on the 650).
I'm trying to find answers and Polycom's only got one reported case
of this (which I find bazaar, but whatever). The problem was
resolved, the proble
From overall apprecation feedback :
#1 Polycom (Any)
#2 Aastra 480i
#3 Cisco 7940+
#4 Linksys SPA-94x
On 4/11/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
I'm t
The Cisco phones are quite good. The thing that most people don't tend to
appreciate about them is that they all are designed essentially for
mass-provisioning in large environments, and to operate with Call Manager.
Provisioning them using their GUI/configuration interface on a one-off
basis
Hello Francis,
I also hev asterisk and sipura. Can we chat online on gmail/yahoo. Let's
make some experiments... I hev the same problem like you.
On 4/12/07, Francis Augusto Medeiros <[EMAIL PROTECTED]> wrote:
On 10 de abr de 2007, at 23:05, James Harper wrote:
>> 2 - How can I gain full con
Basically, I want to send bulk faxes to a list of my clients.
It is time consuming for a person to individually fax so a blast type
solution seems best.
Over IP is of course to save money...
Thanks for the link, reading now...
Any suggestions for the blast then?
Wiley E. Siler
Director of Infor
Doug Lytle ha scritto:
Alberto Pastore wrote:
Firmware on 7940 is 8.6 (the latest one).
I had the same issue. I ended up moving back to firmware P0S3-07-4-00
on the phone. I did a telnet into the phone, did a show register and
shaw some very weird info. Normally, I would see:
>...
But w
Wiley Siler wrote:
Can anyone recommend software that will allow me to utilize my VoIP
provider and send fax over IP?
No, but I can recommend that you read this to see why you shouldn't bother:
http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
Lee.
On Thu, Apr 12, 2007 at 11:59:00AM -0400, jameson asterisk wrote:
> I'm currently looking to interconnect my Asterisk PBX system with the PSTN
> via a digital PRI/T1.
> I know a multitude of options exist for internal PCI cards
> (Digium/Sangoma/Rhino), I was wondering if anyone has any experience
> On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
>> Hi,
>>
>> I have to set call length to 3min, but before hangup have to warn
>> caller. There are many IVRmenu and submenu options with different
>> warning audio.
>> I have to measure somehow the audio file length and subtract it from 3
>>
Drew Gibson wrote:
> We have Cisco, Aastra 480i and Grandstream GXP2000 phones in house.
>
> I only recommend the Cisco phones to people I don't like, overpriced and
> far too much work.
>
> The Aastra 480i is a good quality phone, on par with Cisco and probably
> with Polycom (though I've never
Lee Jenkins wrote:
> Stephen Bosch wrote:
>> Sidetone can be set in the phone configuration; before you do that,
>> though, I need to know what you mean by feedback.
>>
>
> Sorry, should have been more detailed. It's a sort of background
> humming noise, almost like that if you placed the phone n
Bob Smither wrote:
> On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
>> Hi,
>>
>> I have to set call length to 3min, but before hangup have to warn
>> caller. There are many IVRmenu and submenu options with different
>> warning audio.
>> I have to measure somehow the audio file length and s
Drew Gibson wrote:
> Stephen Bosch wrote:
> > Stephen Bosch wrote:
> >
> >> I need to buy some new phones for our own offices.
> >>
> >> I've used only Polycom phones until now, but I'd like to broaden my
> >> experience.
> >>
> >> I'm trying to decide which phones to experiment with. I have the
On Thu, 12 Apr 2007, Wiley Siler said something to this effect:
Can anyone recommend software that will allow me to utilize my VoIP
provider and send fax over IP?
Asterisk can send faxes, if you make it interoperate with a few
well-known open-source utilities and/or software packages, depen
That's just the thing. There are manifold options, but they are all quite
expensive.
--
Alex Balashov <[EMAIL PROTECTED]>
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi,
I'm having problems installing codec g729 on my Asterisk that's running on
FreeBSD 6.0
codec_g729a.so module loads ok, but the register utility doesn't seem to
register the license key correctly, because when I issue "show g729" under
Asterisk's CLI it says that the command is invalid.
It doesn
Playing with hints/presence/BLF on asterisk I've made the following
"discoveries".
1. The wiki at http://www.voip-info.org/wiki/view/Asterisk+presence says:
"If you add incominglimit=1 to your peer in sip.conf, the SIP
channel will notify you when that extension is busy."
As
Can anyone recommend software that will allow me to utilize my VoIP
provider and send fax over IP?
I use Asterisk now for my phone system.
Thanks!
Wiley E. Siler
Director of Information Technology
Education 2020
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
(86
May i ask why not internal?
On 4/12/07, jameson asterisk <[EMAIL PROTECTED]> wrote:
I'm currently looking to interconnect my Asterisk PBX system with the PSTN
via a digital PRI/T1.
I know a multitude of options exist for internal PCI cards
(Digium/Sangoma/Rhino), I was wondering if anyone has an
Mike wrote:
I found *something*. I've gone into my CPU graph (on the phone, in status -
diagnostic). Two phones, one running 1.6.7 and one running 2.1.0, both on
the same Hub, with the same general configuration (different SIP
registration, and each using it's version-specific sip.cfg file).
I'm currently looking to interconnect my Asterisk PBX system with the PSTN
via a digital PRI/T1.
I know a multitude of options exist for internal PCI cards
(Digium/Sangoma/Rhino), I was wondering if anyone has any experience or
recommendations of external PRI media gateways that support SIP.
So f
I'll be sending Digium support the info they requested later today. I
hope it helps.
Greg Siemon wrote:
No luck yet. No response from Digium support so I guess that they are still
waiting for the Zaptel test code.
Greg
-Original Message-
From: Stephen Bosch [mailto:[EMAIL PROTECTED]
Hi Frederico,
I sometimes have the same problem tooI think the problem is related
to VoIP providers registrations. Are you using VoIP services on your PBX?
Thank you.
Giorgio Incantalupo
Frederico Madeira wrote:
Hi,
My asterisk was working fine but today my calls won't out of my
aster
I found *something*. I've gone into my CPU graph (on the phone, in status -
diagnostic). Two phones, one running 1.6.7 and one running 2.1.0, both on
the same Hub, with the same general configuration (different SIP
registration, and each using it's version-specific sip.cfg file).
The pre-2.x ph
Hi,
My asterisk was working fine but today my calls won't out of my asterisk box.
Restarting asterisk i need to wait around 10 min to can run sip show
registry command.
If i try to run this command before, i receive a error like: no such command.
Why this happen ?
Thanks.
--
Frederico Madeir
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
> Hi,
>
> I have to set call length to 3min, but before hangup have to warn
> caller. There are many IVRmenu and submenu options with different
> warning audio.
> I have to measure somehow the audio file length and subtract it from 3
> minut
Alberto Pastore wrote:
Firmware on 7940 is 8.6 (the latest one).
I had the same issue. I ended up moving back to firmware P0S3-07-4-00
on the phone. I did a telnet into the phone, did a show register and
shaw some very weird info. Normally, I would see:
LINE REGISTRATION TABLE
Proxy Regi
Hi all,
Is it possible to configure an extension number to dial a sip address?
For example:
exten => 101,1,Dial(SIP/sip:[EMAIL PROTECTED])
That way I can dial to a sip name using my Hardphone that is not able
to dial using names just numbers.
Thanks in advance.
Ronaldo.
(I hope putting my sip
Hi All,
I have 2 GXV-3000 phones. Working fine when I manually call the phones.
However, if I use a call file to initiate my call to phone 1, then the
dial plan calls
the second phone only the second phone shows video not the first phone.
How can I get video showing on the first phone also?
On Wed, 11 Apr 2007, Kevin P. Fleming wrote:
> Alan Ferrency wrote:
>
> > This means that all queue activity is associated with a SIP channel
> > in the logs, which is not acceptable.
>
> This is why we added the 'membername' argument to the
> AddQueueMember application, so that queue logs can ref
Hi all,
I have asterisk 1.2.13 and problem is about DTMF.When i have incoming call on
Toll-Free number asterisk accept DTMF digits but dial only first in context.
Per instance:
When i press 1 it is OK,but when i try to dial extension 700 asterisk dial only
first digit(1) and i receive from aster
Dovid B wrote:
I wrote this ages ago. You may want to get more current software than
the URL's that are listed.
I just changed the version numbers before doing the script ;)
--
Warm Regards,
Lee
___
--Bandwidth and Colocation provided by Easy
Hi,
Let me join all of you, interested in such monitoring tool.
Cheers
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-us
also I've seen that not having the correct version of sip.cfg and
phone1.cfg could cause weird problems. Make sure you are using the ones
that came with the firmware.
Mike wrote:
Exactly. It's a weird issue, and I can't imagine what the problem is,
except maybe for bad phones (but then again,
Hi.
I'm stuck into an odd situation.
Here's what happens:
4 Thomson ST2030S
2 Cisco 7912
3 Cisco 7940
2 AAstra 480i
Asterisk 1.2.17
Diva 4BRI + chan_capi
I've just upgraded yesterday from Asterisk 1.2.13 to 1.2.17.
Until yesterday, everything was just fine with 1.2.13.
Immediately after the
Somehow, I ended up with BootROM 3.2.3.0002 (which as far as I can tell hasn't
been released yet...) and SIP version 2.1.0.2708.
I do see the "sluggish buttons" from time to time. Rarely, but I do see it.
--TS
>>> Mike <[EMAIL PROTECTED]> 4/12/2007 9:59 AM >>>
Exactly. It's a weird issue, a
Hi Nivlekch,
Thanks for that, just a comment:
What do you mean by new packages? new for spandsp, libmfcr2, unicall?
chan_unicall?
On 4/12/07, nivlekch <[EMAIL PROTECTED]> wrote:
moises, guys,
just an update, steve released new packages early april.
i just did a successful compile, tomorrow i
Stephen Bosch wrote:
Stephen Bosch wrote:
I need to buy some new phones for our own offices.
I've used only Polycom phones until now, but I'd like to broaden my
experience.
I'm trying to decide which phones to experiment with. I have these options:
- A combination of Polycom, Aastra and Sn
Exactly. It's a weird issue, and I can't imagine what the problem is,
except maybe for bad phones (but then again, why would the phones be only
bad with 2.x?)
UnlessI have bootrom 3.2.2.0019. Is that what people running thelatest
have?
Mike
-Original Message-
From: [EMAIL PROTECTE
1 - 100 of 125 matches
Mail list logo
