SO does that mean that if he used BACKGROUND is a SubRoutine he would
get the correct or desired action , from his point of view? It would
jump to the 1 Extension in the SUBROUTINE ?
Tilghman Lesher wrote:
On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
It's a known problem.
Well I can tell you that it makes a difficult programming environment, just a
little more difficult. It means I can't implement a menu as a single reusable
piece of code inside a macro.
- Original Message
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On Fri, Jul 11, 2008 at 8:28 AM, Douglas Garstang [EMAIL PROTECTED] wrote:
Well I can tell you that it makes a difficult programming environment, just
a little more difficult. It means I can't implement a menu as a single
reusable piece of code inside a macro.
I do the IVR stuff in a context
Hi
I have a problem with dtmf recognition an analog lines connected to Sangoma
A200. The digits (in most cases the first one) are doubled and so my IVR is
useless. I tried to adjust the rxgain, toneduration and relxing the dtmf but
nothing worked. I also noticed one thing it only happens during
Hello !
I am wondering if anyone has experiences with the integration of
Asterisk 1.4.19 into Microsoft Dynamics CRM 4.0 ?
Or alternatively integration with Microsoft Office Communications
server (however trying to avoid this, if it isn't really necessary for
the integration).
I would be glad to
I was curious so I took a look at my sounds directory. Most of the
files are 644 except the g729 which are 444. I also noticed that the
ownerid/groupid are a non-existent 1000/1000. I take it that the sound
installer uses something like tar with the option to keep the original
owner and
Call an AGI right before the start of the Dial command to record the start
time and ether use an manager application (makes use of manager API) or call an
DeadAGI once the call has ended (from the h extension). This requires a bit
of programming - but then again some programming is required
Hello,
I've got a problem with rtp handling by siemens c450 and similar. I experience
a couple seconds of silence between early media and normal call (normal call's
rtp is dropped by phone). This is caused by SSRC changing (even though marker
bit is set). I have all relevant patches applied -
On Thu, 2008-07-10 at 10:38 -0500, Jared Smith wrote:
On Wed, 2008-07-09 at 17:54 -0400, Ryan M. Colbert wrote:
I'm trying to build a simple accept/reject screening app for inbound
calls that * forwards to my cell phone. Basically I want * to
announce the caller ID and then let me press 1
Folks:
This is my first post, so please let me know if I transgress in any way...
In updating to 1.4.21 recently, we've encountered a problem, when running
over a satellite connection (where the latency is considerable; a regular
internet connection did not exhibit this problem), where incoming
Thanks, but that won't do what I need. By calling an AGI before the call starts
and after the call ends, all I am doing is accounting the start and the end of
the call, not actively monitoring the duration of the call as it occurs.
- Original Message
From: Cosmin Prund [EMAIL
On Friday 11 July 2008 01:28:34 Douglas Garstang wrote:
Well I can tell you that it makes a difficult programming environment, just
a little more difficult. It means I can't implement a menu as a single
reusable piece of code inside a macro.
That's the point. A Macro is NOT a subroutine.
Yes, and by doing that your compounding the fact that all your variables are
global.
- Original Message
From: randulo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 11, 2008 12:14:28 AM
Subject: Re:
On Friday 11 July 2008 01:05:22 Al Baker wrote:
Tilghman Lesher wrote:
On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
It's a known problem.
If you call Background() in a macro, then Asterisk will look for the
extensions to jump to in the CALLING Macro/context and NOT the
I dont know if this will help, but I have been working with MS OCS at work,
and * 1.6 integrates rather wall tith OCS speech server.
If you need help on that relm, I can try to help. (admitidly I dont have
inbound calls working, but we arnt worried about that, as our appplication
is strictly
On Friday 11 July 2008 09:22:25 Douglas Garstang wrote:
Yes, and by doing that your compounding the fact that all your variables
are global.
No, his variables are local to the channel he's using. Global variables are
a completely different beast.
--
Tilghman
On Friday 11 July 2008 09:21:56 Douglas Garstang wrote:
Thanks, but that won't do what I need. By calling an AGI before the call
starts and after the call ends, all I am doing is accounting the start and
the end of the call, not actively monitoring the duration of the call as it
occurs.
It is
Well, a macro is the closest thing the dial plan has to a subroutine, and
without that, we might as well be programming in Assembler (no subroutines,
local variables, lots of goto's... sound familiar?).
Doug.
- Original Message
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk
On Fri, 11 Jul 2008, Artie Gold wrote:
This is my first post, so please let me know if I transgress in any way...
A more meaningful subject would get more interest. It also helps when
someone else is searching the mailing list archives. For example, SIP
timing out over satellite.
In
In updating to 1.4.21 recently, we've encountered a problem, when running
over a satellite connection (where the latency is considerable; a regular
internet connection did not exhibit this problem), where incoming calls are
being dropped as a result of the sip handshake timing out (dropping down
Ugh. Yes, the variables are local to the current channel. However, they are
global to the entire dial plan within the current channel. I have stepped on
myself many times because I've had a loop counter called $i for example, jumped
somewhere else within that loop, reused the same variable
Hi,
We are using asterisk 1.4.20 load. We have seen that couple of
times Asterisk fails to convert SIP INFO packet in to Inband tone.
Problem Description:
Asterisk behaves as a media proxy between proxy1 and
proxy2.
Proxy1 transmit
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
Hi! I am a newbie using Asterisk. I am developing an IVR using perl
from AGI and Cepstral as voices
The AGI is this
[snip]
My problem is that i cant hear anything when play the file sound
using $AGI-stream_file($filename);
I put
On Fri, 11 Jul 2008, Douglas Garstang wrote:
Ugh. Yes, the variables are local to the current channel. However, they
are global to the entire dial plan within the current channel. I have
stepped on myself many times because I've had a loop counter called $i
for example, jumped somewhere
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
--
#Joseph
GPG KeyID: ED0E1FB7
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25
On Friday 11 July 2008 09:40:55 Douglas Garstang wrote:
Well, a macro is the closest thing the dial plan has to a subroutine, and
without that, we might as well be programming in Assembler (no subroutines,
local variables, lots of goto's... sound familiar?).
I've mentioned Gosub at least twice
Hi all,
I have a problem with my asterisk box and an X100P FXO card. I am able to
place outgoing calls from my SIP phone (Cisco 7940) to any external number
using my PSTN line, but when I call my PSTN line from my cell phone, the
Cisco doesn't ring (and no message appears in the Asterisk CLI).
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them at our respective homes. Rock solid stable.
No issues whatsoever.
N.
On Friday 11 July 2008 09:17:37 Artie Gold wrote:
In updating to 1.4.21 recently, we've encountered a problem, when running
over a satellite connection (where the latency is considerable; a regular
internet connection did not exhibit this problem), where incoming calls are
being dropped as a
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them at our respective homes. Rock solid stable.
No issues
The other thing that baffles me about this setup is that it only seems
to happen to people who are connected to the internal network in the
office. They have about 30 remote users that have not reported this
same problem, their issue is usually bandwidth related from their home
On 07/11/08 18:37, Dave Cotton wrote:
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them at our respective
Dave Cotton wrote:
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them at our respective
Joseph wrote:
On 07/11/08 18:37, Dave Cotton wrote:
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them at
This is a quite promising idea. Many thanks.
I'll post my results to the list...
Cheers,
--ag
On Fri, Jul 11, 2008 at 11:22 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Friday 11 July 2008 09:17:37 Artie Gold wrote:
In updating to 1.4.21 recently, we've encountered a problem, when
I want to track call duration while the call is in progress.
- Original Message
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 11, 2008 7:39:40 AM
Subject: Re: [asterisk-users]
A subroutine with arguments?
- Original Message
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 11, 2008 8:58:46 AM
Subject: Re: [asterisk-users] Asterisk as an IVR solution
On
Dave Cotton wrote:
Joseph wrote:
On 07/11/08 18:37, Dave Cotton wrote:
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them
Fine, I'll call it ${LoopVariable} then... how's that going to fix the problem?
- Original Message
From: Steve Edwards [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 11, 2008 8:43:47 AM
Subject: Re:
On Friday 11 July 2008 12:07:37 Douglas Garstang wrote:
A subroutine with arguments?
In 1.6, yes, or in the 1.4 backport, yes.
--
Tilghman
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25
On Jul 11, 2008, at 10:08 AM, Douglas Garstang wrote:
I want to track call duration while the call is in progress.
To accomplish what? Are you wanting to beep the channel every 10
seconds? Are you wanting to play a you have 60 seconds left message
when they approach some quota? Are you
Tzafrir Cohen wrote
So this is an FXS module.
Guess I mixed it up ;-)
For starters, do you have echo cancellation enabled?
asterisk -rx 'zap show channel 120' | grep 'Echo'
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
How can I turn it on?
Udo
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri, 11 Jul 2008 08:10:38 -0700
Subject: Re: [asterisk-users] Asterisk cant play sounds from AGI
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
Hi! I am a newbie using Asterisk. I am developing an IVR using perl
Unfounded rumors say that ABE doesn't come with app_rnddropcall ;-]
On Fri, Jul 11, 2008 at 12:40 PM, Carlos Chavez [EMAIL PROTECTED]
wrote:
The other thing that baffles me about this setup is that it only
seems
to happen to people who are connected to the internal network in the
On Friday 11 July 2008 12:40:47 Edwin Quijada wrote:
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri, 11 Jul 2008 08:10:38 -0700
Subject: Re: [asterisk-users] Asterisk cant play sounds from AGI
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
Hi! I am a newbie
Wanting to provide a real time call timer on a web page.
- Original Message
From: Daniel Hazelbaker [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 11, 2008 10:17:01 AM
Subject: Re: [asterisk-users]
From: Steve Edwards [EMAIL PROTECTED]
On Fri, 11 Jul 2008, Douglas Garstang wrote:
Ugh. Yes, the variables are local to the current channel. However, they
are global to the entire dial plan within the current channel. I have
stepped on myself many times because I've had a loop counter
On Fri, Jul 11, 2008 at 07:24:23PM +0200, Udo Schacht-Wiegand wrote:
Tzafrir Cohen wrote
So this is an FXS module.
Guess I mixed it up ;-)
For starters, do you have echo cancellation enabled?
asterisk -rx 'zap show channel 120' | grep 'Echo'
Echo Cancellation: 128 taps
On Fri, 11 Jul 2008, Tilghman Lesher wrote:
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
My problem is that i cant hear anything when play the file sound
using $AGI-stream_file($filename); I put asterisk in verbose mode
but just see that it plays the sound but I cant hear anything.
The Asterisk development team has released version 1.4.5 of libpri. This
release was made solely to correct a problem introduced in version 1.4.4.
In February of 2008, a change was made in libpri to support inband audio
(progress) when the far end of a PRI circuit issues a RELEASE message,
I saw this shortly after ssh'ing into a box that was not answering sip inbound
calls:
--- SIP read from 192.168.100.253:5060 ---
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.5;rport=5060;branch=z9hG4bK7a87d233
Max-Forwards: 70
From: xx sip: xx @192.168.100.5;tag=as588c6a60
Joseph L. Casale wrote:
I saw this shortly after ssh'ing into a box that was not answering sip
inbound calls:
--- SIP read from 192.168.100.253:5060 ---
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.5;rport=5060;branch=z9hG4bK7a87d233
Max-Forwards: 70
From: xx sip: xx
I've just added a PREVIEW release of my upcoming how-to guide for Asterisk
PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2.
It addresses all kinks and showstoppers that many people have experienced
over the past year or so. Because this is a preview, it is not the final
On Tue, Jul 08, 2008 at 11:13:02AM -0700, Daniel Hazelbaker wrote:
D-Marc that terminates the 25-pair analog line coming in (this does
not just contain our lines as I can tap into other peoples lines and
hear there conversations, love security).
The T-1's aren't on that, though, right?
Today I had a problem where the internet connection is unstable so
calls are getting dropped all over the place. The one thing I do not
understand is that at least 30 phones on the internal network went to
No Service. Since they are on the same network segment and on the
same subnet I do
Very cool, you've piqued my interest. Since I haven't launched an
instance before, where's the best place to learn to do that? What's the
approximate monthly cost of hosting an Asterisk PBX on EC2?
Ronald Lewis wrote:
I've just added a PREVIEW release of my upcoming how-to guide for
Hi All,
I seen earlier the first guide on deploying asterisk on EC2 in the
list becoming available.
Has anyone deployed a hosted environment like enswitch using EC2? I
was wondering if anyone had any thoughts on concerns on the
feasibility in doing this using cloud computing?
I would have
Still, that's kind of funny though :)
Hilarious :) This CentOS machine running asterisk is in a Xen vm and its not
behaving well.
I am moving it to physical hardware asap and thought that may have been part of
some
indication of the myriad of issues it has. That is a priceless coincidence!
Googlezon will rule the world. http://www.robinsloan.com/epic/
On Fri, Jul 11, 2008 at 3:28 PM, Robert McNaught [EMAIL PROTECTED]
wrote:
Hi All,
I seen earlier the first guide on deploying asterisk on EC2 in the
list becoming available.
Has anyone deployed a hosted environment like
On Jul 11, 2008, at 12:09 PM, Jay R. Ashworth wrote:
On Tue, Jul 08, 2008 at 11:13:02AM -0700, Daniel Hazelbaker wrote:
D-Marc that terminates the 25-pair analog line coming in (this does
not just contain our lines as I can tap into other peoples lines and
hear there conversations, love
On Fri, Jul 11, 2008 at 12:58:59PM -0700, Daniel Hazelbaker wrote:
Really? You have an RJ-21X block that contains both analog AND T1
wires? That's really uncommon. I hope they at least put the red
special service caps on the T1 wires.
Yup. I thought that pretty funny myself. 10 year
Date: Fri, 11 Jul 2008 11:29:58 -0700
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk cant play sounds from AGI
On Fri, 11 Jul 2008, Tilghman Lesher wrote:
On Jul 10, 2008, at 7:54 PM, Edwin
Carlos Chavez wrote:
Today I had a problem where the internet connection is unstable so
calls are getting dropped all over the place. The one thing I do not
understand is that at least 30 phones on the internal network went to
No Service.
When this happens try to capture DNS traffic
That happens all the time when the T1s are purchased from a CLEC as the
RBOCs just deliver the clec pairs wherever.
I can think of at least two or three demarcs that I have been to in the
last few months that were mixed like that. Here in Qwest territory the
T1s use a different color cross
On Fri, Jul 11, 2008 at 12:58:59PM -0700, Daniel Hazelbaker wrote:
Really? You have an RJ-21X block that contains both analog AND T1
wires? That's really uncommon. I hope they at least put the red
special service caps on the T1 wires.
Yup. I thought that pretty funny myself. 10
On Jul 11, 2008, at 1:31 PM, Edwin Quijada wrote:
vm-debian#file tts-hello
example.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM,
16 bit, mono 8000 Hz
Other than the filename being wrong which I would assume is the result
of a copy and paste from the original e-mail, that
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based calls (softphones)
could have and advantage over hardphones, provided there is a parameter
On Fri, 11 Jul 2008 17:13:15 -0400, C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based calls (softphones)
could have and
Good light codecs like speex, and minimal feature sets.
C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based calls
Not sure about magicjack but skype has supernodes that play a large
part in how the system works well.
http://geemodo.blogspot.com/2006/10/dont-be-skype-supernode-or-how-not-to.html
Thanks,
Steve T
On Fri, Jul 11, 2008 at 5:29 PM, Anthony Francis [EMAIL PROTECTED] wrote:
Good light codecs like
Better handling of the packets, that's for sure. Also, the algorithm is
smart, and flexible... that being said, it opens more questions than
answers.
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Friday, July 11, 2008 5:24
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based calls (softphones)
could have and advantage over hardphones, provided there is a parameter
Yes, I have designed two different webphones, granted, using third party
libraries, and magicjack's quality is better. I acknowledge that.
Thank you, but referring me to someone's review won't help me much... I am
interested in the internals. Regardless, their technique has a twist, and
I don't see Magicjack being around long. The business model isn't
sustainable without tons of ads, and even then, people will either
ignore them if they are audio or if they are popups, they will simply
close them or disable them.
I might buy one just to hack it. Has anyone sniffed it or poked
On Fri, 11 Jul 2008 18:28:09 -0400, Steve Totaro wrote:
I don't see Magicjack being around long. The business model isn't
sustainable without tons of ads, and even then, people will either
ignore them if they are audio or if they are popups, they will simply
close them or disable them.
I might
As per the ads, if people ignore them or not, doesn't matter. Advertisers
will fall in love with the idea that the venue reaches 1 million people,
or more. As per the price of the service, they might be calculating the
fact that the average monthly consumption of minutes on a softphone could
Here's an interesting challange.
I need to implement a calling card application, where I call the Dial() command
and pass it (L)imit information. Nothing difficult about that. Except it is a
requirement that rather than ending the call when the limit is reached, the
user gets the option to
As Michael Graves points out, people will hack it to run on thin
clients and why not virtual machines with very limited access? Maybe
an AP with a USB port and OpenWRT or something?
Remember when NetZero really cost nothing? I had a program someone
wrote to close the as Windows, later I figured
On Fri, Jul 11, 2008 at 7:12 PM, Douglas Garstang [EMAIL PROTECTED] wrote:
Here's an interesting challange.
I need to implement a calling card application, where I call the Dial()
command and pass it (L)imit information. Nothing difficult about that.
Except it is a requirement that rather
Thanks, but how does that extend the core functionality of Dial()? If Dial()
can't do it, how does a wrapper do it?
- Original Message
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July
On Fri, Jul 11, 2008 at 7:50 PM, Ronald Lewis [EMAIL PROTECTED] wrote:
I've just added a PREVIEW release of my upcoming how-to guide for Asterisk
PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2.
It addresses all kinks and showstoppers that many people have
Strange prices, look at ovh, 233€/m~~350$ And ovh provide a REAL unlimited.
CPU
Intel Xeon X5355
1x 4x 2.66 GHz
L2: 8Mo, FSB: 1333MHzQuadruple Coeur
Architecture64 bits RAM
8 Go FBDIMM DDR2 HDD
2x 750 Go Type HDD
SATA2 RAID HARD 1Interfaces
2 x 1 Gbps SPEED
2 Gbps Traffic
UNLIMITED
IP fixe2
Quote Seriously though, if your business lives and dies by the phone
system,
get T1 with SIP from your provider directly
If your business lives and dies, get that regular, boring, RELIABLE, TDM-T1.
SIP/VOIP/Whatever - Cool fun, great when it works
TDM-T1 - Unsurpassed reliabilty
Steve Totaro
Could you clarify how you end up with 1.4 Backport ?
If you go to DIGIUM and download 1.4 do you have a backport 1.4 or is
there
a super-secret-non-more-secret-archive one would get it from ?
I have never really understood this.
Thank You
Tilghman Lesher wrote:
On Friday 11 July 2008 12:07:37
On Sat, Jul 12, 2008 at 2:16 AM, Grygoriy Dobrovolskyy
[EMAIL PROTECTED] wrote:
Strange prices, look at ovh, 233€/m~~350$ And ovh provide a REAL unlimited.
And I'm sure there will be someone somewhere who has a better deal as
well. Whenever hosting gets mentioned in tech forums you always end up
C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based calls (softphones)
could have and advantage over hardphones, provided
Douglas Garstang wrote:
Well, a macro is the closest thing the dial plan has to a subroutine,
and without that, we might as well be programming in Assembler (no
subroutines, local variables, lots of goto's... sound familiar?).
Doug.
- Original Message
From: Tilghman Lesher
Thank You - clears up a LOT I did not fully grasp
Tilghman Lesher wrote:
On Friday 11 July 2008 01:05:22 Al Baker wrote:
Tilghman Lesher wrote:
On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
It's a known problem.
If you call Background() in a macro, then
On Fri, 11 Jul 2008, Edwin Quijada wrote:
I recorded the sound using Cepstral. This is my AGI
I thought maybe was my sound card but this works fine
Why would you think it was the sound card?
1) Try enabling AGI debugging. For 1.2, enter agi debug and then execute
your agi. The important part
On Friday 11 July 2008 21:24:10 Al Baker wrote:
Tilghman Lesher wrote:
On Friday 11 July 2008 12:07:37 Douglas Garstang wrote:
A subroutine with arguments?
In 1.6, yes, or in the 1.4 backport, yes.
Could you clarify how you end up with 1.4 Backport ?
If you go to DIGIUM and download
Hi,
I'm having a problem with IMAP storage and asterisk. Here is the error
message I get (in this instance its checking messages):
[Jul 11 23:14:12] WARNING[9888]: app_voicemail.c:8738 mm_log: IMAP
Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[Jul 11 23:14:12] ERROR[9888]:
Steve Underwood wrote:
C. Savinovich wrote:
I am puzzled by the quality of magicjack. I keep trying to figure out how
they can the quality be that adequate. Since Skype also has an excellent
quality, that leaves me to believe that software based calls (softphones)
could have and
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