Hi,
When I build the same asterisk package that I build on i386 on
x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same
set of installed packages. What should I be looking for in the output of
./configure to get a clue of what might be missing?
TIA.
On Wed, 19 Mar 2008 10:10:21 + (GMT), Gordon Henderson
[EMAIL PROTECTED] wrote:
I got free installation for Featureline Compact
on 3 yr contract.
So it saved me £££s!
Intersting... But shouldn't you be using VoIP for your calls anyway...
Then just one basic BT line, and a
On Thu, Mar 20, 2008 at 01:09:36AM -0400, Al Baker wrote:
Not sure if this is the best place to ask this or not...but since it was
mentioned..
Is SwitchVox a alternative to * ?
Were they a competitor to *, and DIGIUM bought them and so DIGIUM
has 2 Totally Different PBX software packages
Date: Wed, 19 Mar 2008 11:31:57 +0200
From: Atis Lezdins [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Handling 3 different call ending causes
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type:
On Thu, Mar 20, 2008 at 12:59:08AM -0400, Alex Balashov wrote:
At the risk of inflaming a lot of passions, including those of
hard-working developers, I must say that where Asterisk may be
production-worthy, the entire constellation of things (like Zaptel) of
which its PSTN hardware
On Thu, Mar 20, 2008 at 12:48:54AM -0600, William F. Acker WB2FLW
+1-303-722-7209 wrote:
Hi,
When I build the same asterisk package that I build on i386 on
x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same
set of installed packages. What should I be looking
Excellent topic and points brought up by all!
On Thu, Mar 20, 2008 at 8:43 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Think of Asterisk not as a PBX but as a PBX toolkit. Various people in
That's always been the way I saw asterisk. I wondered why people
sometimes try to interface it with legacy
On Thu, Mar 20, 2008 at 01:27:47AM -0400, Al Baker wrote:
From a lot of experience - you are not being anywhere near paranoid
enough !!
Think dual RAID controllers, Dual power supplies off of, at a Minimum,
separate isolated circuits, with Hefty UPS that is in-line so it filters
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a ser registration proxy. Everything works
fine so far, but i can´t use the hint feature. Its possible to subscribe
Hello,
You have to set up a hint extension pointing to the Sip user like exten
= 777,hint,SIP/username
That extension is used in the Snom as extension.
if you use the following format of this option field you should be able
to pickup: sip:[EMAIL PROTECTED]|*9
777 is the hint extension
127.0.0.1
I am using asterisk 1.4.18 (server A ) and have it store records in
mysql database . One of my client uses predictive dialer ( asterisk
1.2.26 based and server B ) which makes many calls . B registers with
A over sip and there is no nat involved If i re-invite rtp from
server B to my carrier (
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
thanks for you help,
regards,
Asif
Message: 14
Date: Wed, 19 Mar 2008 10:39:22 -0500
From: Eric Wieling [EMAIL PROTECTED]
Subject: Re: [asterisk-users] How to configure
In article [EMAIL PROTECTED],
Lee, John (Sydney) [EMAIL PROTECTED] wrote:
I am working on a menu to accept input from a caller like as follows:
Exten = 100,1,Answer()
Exten = 100,n,Playback(LONG-MESSAGE)
Exten = 100,n,Read(OPTION,,2)
...
When I tested it, I noticed if I start pressing a
On 3/20/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Date: Wed, 19 Mar 2008 11:31:57 +0200
From: Atis Lezdins [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Handling 3 different call ending causes
To: Asterisk Users Mailing List - Non-Commercial Discussion
Gordon Henderson wrote:
On Wed, 19 Mar 2008, Norman Franke wrote:
As for why a company would purchase hard phones, several reasons. First, we
are replacing many hard phones with computers. We have a custom application
and have been moving folks main numbers to use the computer. We can
Tzafrir Cohen wrote:
Yeah, right. And we have no SIP compatibility issues at all. It is also
funny that you reflect the quality of old PRI card of one company and
yet ignore all the past mishaps of SIP devices.
Oh, no, I didn't mean to imply that. There are plenty of SIP interop
problems
On Wed, 19 Mar 2008 16:38:23 -0500, Bill Andersen [EMAIL PROTECTED]
wrote:
Although this is a users list, I think it is more of a list
for Asterisk resellers. I'd be interested in how many of you
are simply using Asterisk as your phone system and NOT selling
your services or an Asterisk
Although this is a users list, I think it is more of a list
for Asterisk resellers. I'd be interested in how many of you
are simply using Asterisk as your phone system and NOT selling
your services or an Asterisk based solution?
I'm responsible (development, maintenance, support) for an
On Thu, Mar 20, 2008 at 06:45:14AM -0400, Alex Balashov wrote:
Tzafrir Cohen wrote:
Yeah, right. And we have no SIP compatibility issues at all. It is also
funny that you reflect the quality of old PRI card of one company and
yet ignore all the past mishaps of SIP devices.
Oh, no, I
Andreas Sikkema wrote:
I've literally got _thousands_ of users and Asterisk is rock
solid for us.
I think most of the instabilities are from the use of queues and
mixmonitor/chanspy.
I don't use either and have no real issues. I still restart the
Asterisk service once a week though,
On 3/20/08, Stefan Schmidt [EMAIL PROTECTED] wrote:
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a ser registration proxy. Everything works
fine so far, but
- Original Message -
From: Lee, John (Sydney) [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 19, 2008 11:48 PM
Subject: [asterisk-users] Newbie IVR: How to read() before playback()
isfinished?
I am working on a menu to accept input from a caller like as
20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a ser registration proxy. Everything
works
fine so far, but i can´t
Appologies for top-posting. This is the most interesting thread in a
long time. Alex, yours is the most well considered opinion I've seen in
a long while. I exactlt reflects my own, moerw limited experience.
Thank you for chiming in.
Two weeks ago on the VOIP Users Conference weekly call we had
MeetMe() has the K option that kills the conference,
how do I do that in app_conference() as there no kill the conference option?
Jerry
___
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asterisk-users mailing list
To
Must be having a DOH! week.
Problem turned out to be the Fedora core firewall that was turned on.
Sorry folks.
On Wed, Mar 19, 2008 at 3:01 PM, Aaron Fransen [EMAIL PROTECTED]
wrote:
Finally got my Cisco Call Manager link going; what it turned out to be was
having the same extension on the
On 20/03/2008, Johansson Olle E [EMAIL PROTECTED] wrote:
20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a
I reboot every evening :) Drew, what's the uptime on your
asterisk process on that box that's been up for 193 days?
I too restart the asterisk process every night as part of the cron process.
Many people here seem to be under the impression that restarting the
application every day is a bad
On 3/20/08, Steve Davies [EMAIL PROTECTED] wrote:
On 20/03/2008, Johansson Olle E [EMAIL PROTECTED] wrote:
20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt
Perhaps in a similar thread, is it possible to somehow SET the state
of a hint from the dialplan? Perhaps a bit like:
Set(${ChanIsAvail(hint,234)}=Busy)
or perhaps have a pseudo-device facility where you can add
it to the
end of the hint list to hint-the-hint. Something
Although this is a users list, I think it is more of a list for
Asterisk resellers. I'd be interested in how many of you are simply
using Asterisk as your phone system and NOT selling your services or
an Asterisk based solution?
I actually work as a software engineer for a big telecom
On Thu, Mar 20, 2008 at 09:31:03AM -0500, John Faubion wrote:
I reboot every evening :) Drew, what's the uptime on your
asterisk process on that box that's been up for 193 days?
I too restart the asterisk process every night as part of the cron process.
Many people here seem to be under
Holy Mackeral. Ignore that last message. I still do NOT know how to route
calls with the same extension being used in two locations, however the issue
I've resolved is getting Cisco CallManager and Asterisk talking together
properly.
Sorry folks AGAIN.
So if anybody has ideas on how to have
I've got a couple of extensions in users.conf that have both SIP and IAX
access(IAX softphone, SIP hard phone).
I'd like to setup my dial string to check to see which they are actively
registered with, and send the call appropriately.
Right now I have:
Exten =
On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED]
wrote:
Sure some others on here may disagree, but I am also over on the
trixbox
forums, and have often seen talk about the 2.6.9 kernel having
interrupt
issues, and such that cause asterisk issues. One reason I think
they moved
forward
On Mar 19, 2008, at 5:56 PM, [EMAIL PROTECTED]
wrote:
Anyone? Just a user?
I'm just a user, although I also develop things for internal use.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
___
-- Bandwidth and Colocation
On Thu, 20 Mar 2008, Tzafrir Cohen wrote:
On Thu, Mar 20, 2008 at 12:48:54AM -0600, William F. Acker WB2FLW
+1-303-722-7209 wrote:
Hi,
When I build the same asterisk package that I build on i386 on
x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same
set of
Still grasping at straws trying to solve DTMF detection issues with one
of my asterisk servers. This particular server is now running Asterisk
1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P
cards. I have tried adjusting channel gains, turning call progress and
I was running Trixbox 2.2 up until about 2 months ago, and had persistent
interrupt issues. I upgraded to 2.4, with the updated kernel, and its been
complete smooth sailing ever since.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Norman Franke
Sent: 20 March 2008
On Thu, Mar 20, 2008 at 11:10:21AM -0400, Norman Franke wrote:
On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED]
wrote:
Sure some others on here may disagree, but I am also over on the
trixbox
forums, and have often seen talk about the 2.6.9 kernel having
interrupt
issues, and such
Thank you all for the great advice. Although fairly new to Asterisk, and
relearning systems administration, it has helped put some perspective on
the matter for me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, March 19,
List,
Question about the Polycom 650: when dialing the digits for a phone number,
and an incoming call comes in, does the phone prevent you from completing
your outgoing call until the phone stops ringing, like a Cisco 79X0 does?
--Brent
___
--
To add some further details to this thread I set up a Monitor command
that records just the IVR portion of an incoming call. I left the m
flag off so I could listen to the incoming audio separate from the
outgoing recording. On calls where the DTMF detection works correctly I
only hear
John Faubion wrote:
Although this is a users list, I think it is more of a list for
Asterisk resellers. I'd be interested in how many of you are simply using
Asterisk as your phone system and NOT selling your services or an Asterisk
based solution?
I actually work as a software
Am Donnerstag, den 20.03.2008, 08:59 -0600 schrieb Aaron Fransen:
Holy Mackeral. Ignore that last message. I still do NOT know how to
route calls with the same extension being used in two locations,
however the issue I've resolved is getting Cisco CallManager and
Asterisk talking together
Hi,
I'm getting this error when registering with SIP server using Asterisk
1.4.10 and Freepbx...
I'm getting this error no matter what I try to setup in sip.conf :
- I'm getting confused whether options are maxexpirey=36000 or
maxexpiry=36000 ?
- Can I solve this with some settings in
Hello,
I am having some troubles with Snom phones and maybe someone can help
me.
Let me say this: BLF and pickup works great with Polycomes and
Grandstream etc... So I think my problem might not be Asterisk related
but I am not 100% sure.
The snom phones subscribe to my extensions (hint
For such a small system there is no earthly reason for it to
be 10 percent of that, even on a 5 year lease.
I know that EVERYTHING is big in Texas, but that is nothing
more than highway robbery.
I fully agreed, that's why we built her an Asterisk based system. Splitting
this up they wanted
No, I meant if I leave this office, what to do when the cpu fan or power
supply breaks on our current * box :) They might just be so worried
that they'd *want* something like the 3Com V3000 :)
Steve Totaro wrote:
Call your dealer as I am sure you would have a support contract.
Haven't
Am Donnerstag, den 20.03.2008, 16:59 +0200 schrieb Tzafrir Cohen:
And what happens if at the time of the shutdown there was a
___
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ROTFL
Trafrir, you made my day.
(BTW: I
hello,
you have to use following format in den extension key of the snom:
sip:[EMAIL PROTECTED];user=phone|*7
the |*7 is the extension to dial if you want to pickup the ringing
(blinking) line.
maybe you should try sip:[EMAIL PROTECTED]|*7 where 100 is your hint
extension
and *7100 is a
Mian M Asif wrote:
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
exten = s,n,Set(OLD_EXTEN=${EXTEN})
Then later, just use ${OLD_EXTEN}
___
-- Bandwidth and
On March 20, 2008 02:33:52 pm Anselm Martin Hoffmeister wrote:
Am Donnerstag, den 20.03.2008, 16:59 +0200 schrieb Tzafrir Cohen:
And what happens if at the time of the shutdown there was a
ROTFL
Trafrir, you made my day.
Oh god, I didn't realize that wasn't a typo until you wrote that...
[EMAIL PROTECTED] wrote:
I am planning to write a module to find if a Special Information was detected
or not.
Can anyone please help me to figure out the below fields?
1. The Frequency of a frame
2. Length of frame in milliseconds
Aren't all the frames in asterisk 20ms long, no
On Thu, Mar 20, 2008 at 3:01 PM, RE Kushner List Account [EMAIL PROTECTED]
wrote:
Al Baker wrote:
Quote
This code is pre-Asterisk 1.0... It processes quite a few calls daily, I
have about 1,800 DID numbers pointed at it,
Are you SURE on that figure. Since you cold have at MOST
Mojo with Horan Company, LLC wrote:
[EMAIL PROTECTED] wrote:
I am planning to write a module to find if a Special Information was
detected or not.
Can anyone please help me to figure out the below fields?
1. The Frequency of a frame
2. Length of frame in milliseconds
hi:
In my zapata.conf i have 4 fxo configured channels,for fxo number 1 to 3 i
added polarity reversal property but for fxo number 4 i didnt add polarity
reversal property but it still giving me on cosole that fxo number 4 is
polarized (because the line on fxo number 4 is not polarized).
what
Steve Totaro wrote:
On Thu, Mar 20, 2008 at 3:01 PM, RE Kushner List Account [EMAIL PROTECTED]
wrote:
Al Baker wrote:
Quote
This code is pre-Asterisk 1.0... It processes quite a few calls daily, I
have about 1,800 DID numbers pointed at it,
Are you SURE on that figure.
On Thu, Mar 20, 2008 at 09:09:05PM +0200, wassim darwish wrote:
hi:
In my zapata.conf i have 4 fxo configured channels,for fxo number 1
to 3 i added polarity reversal property but for fxo number 4 i didnt
add polarity reversal property but it still giving me on cosole that
fxo number 4
Al Baker wrote:
Quote
This code is pre-Asterisk 1.0... It processes quite a few calls daily, I
have about 1,800 DID numbers pointed at it,
Are you SURE on that figure. Since you cold have at MOST 4 T1's coming into
that box, 1,800 DIDs pointing to it sems like
one hell of a congestion
Hello All,
I've been trying to get BLF working with Asterisk 1.6-LatestBeta, and My
Cisco 7970 (Latest SIP Firmware).
Has anyone successfully completed this?
I got the patch to merge in from
http://www.voip-info.org/wiki/view/Asterisk+Presence+for+Cisco+79x1+Phones
With a bit of hackery to
On Thu, 20 Mar 2008, Norman Franke wrote:
On Mar 20, 2008, at 12:59 AM, [EMAIL PROTECTED] wrote:
Sure some others on here may disagree, but I am also over on the trixbox
forums, and have often seen talk about the 2.6.9 kernel having interrupt
issues, and such that cause asterisk issues. One
On Thursday 20 March 2008 05:06:29 am Mian M Asif wrote:
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
thanks for you help,
regards,
Asif
Message: 14
Date: Wed, 19 Mar 2008 10:39:22 -0500
From: Eric
That probably includes 5 years of support but still expensive.
John Faubion wrote:
Although this is a users list, I think it is more of a list for
Asterisk resellers. I'd be interested in how many of you are simply
using Asterisk as your phone system and NOT selling your services or an
Hello,
Do your verify, the codecs, of both clients, in your sip.conf?
What codec do you use?
Best Regards
On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay [EMAIL PROTECTED] wrote:
Hi,
I am sorry my questinos are too fundamental. I am new to Asterisk, and
hope to catch up as fast as I can.
Pete,
I have never done it but it would seem that running a SIP client on
your SIP server may be problematic.
Also, 58.251.75.228 is certainly not on your 192.168.x.x subnet. Is
your machine dual homed?
Thanks,
Steve Totaro
On Thu, Mar 20, 2008 at 9:34 PM, Carlos Rojas [EMAIL PROTECTED]
Sorry, I am tired and missed the virtual IP part. I am not quite sure
what that means or why you are sending traffic to the routeable IP.
Are you using a FQDN with external DNS or the IP in your client?
Thanks,
Steve Totaro
On Thu, Mar 20, 2008 at 10:42 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
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