We have a situation where it's sometimes taking Asterisk 17-19 minutes to post
CDR's, both over the AMI, and over the MySQL socket. It seems however that they
are logged locally to /var/log/asterisk/cdr-csv/Master.csv right after the call
is terminated.
Anyone got any idea what's causing this?
Just went to order some IVR prompts from the digium web site
From the digium web site:
We have created an easy and cost effective way to have customized recordings
done quickly and with no hassle.
I thought this was rather amusing, as:
1. If you want multiple prompts recorded, you need to
] New Bridge Command/Event in 1.6?
20 jul 2008 kl. 02.55 skrev Douglas Garstang:
I just downloaded Asterisk 1.6 beta 9 because I had read that there
was a new bridge command. After looking through the doc/*
documentation, I see no mention of a bridge application or AMI
command.
Does
I just downloaded Asterisk 1.6 beta 9 because I had read that there was
a new bridge command. After looking through the doc/* documentation, I
see no mention of a bridge application or AMI command.
Does it exist?
I
am trying to take a bridged call, and redirect each to another
destination, which
I just downloaded Asterisk 1.6 beta 9 because I had read that there was a new
bridge command. After looking through the doc/* documentation, I see no mention
of a bridge application or AMI command.
Does it exist?
I am trying to take a bridged call, and redirect each to another destination,
, 2008 at 10:52:53AM -0700, Douglas Garstang wrote:
Wanting to provide a real time call timer on a web page.
Can't you get information about other channels through the manager
interface without this special AGI?
Maybe you just need to somehow mark those channels as interesting
before the Dial
All,
I was able to use the Redirect AMI command to take two bridged channels and
send them elsewhere in the dial plan. Great.
Now... how can I bridge them back together again? Looks like Asterisk 1.6 might
have a bridge command. What about Asterisk 1.4?
Doug.
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 10, 2008 6:07:36 PM
Subject: Re: [asterisk-users] Asterisk as an IVR solution
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
to solve all problems
writing programs, so maybe someone else has a better idea!
--
Cosmin Prund
De
la:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] În numele Douglas
Garstang
Trimis: Thursday, July 10, 2008 7:49 PM
Către: asterisk-users@lists.digium.com
Subiect: [asterisk-users] Tracking Call Time
: [asterisk-users] Asterisk as an IVR solution
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
Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 11, 2008 7:20:40 AM
Subject: Re: [asterisk-users] Asterisk as an IVR solution
On Friday 11 July 2008 01:28:34 Douglas Garstang wrote:
Well I can tell you that it makes a difficult programming
@lists.digium.com
Sent: Friday, July 11, 2008 7:36:54 AM
Subject: Re: [asterisk-users] Asterisk as an IVR solution
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
] Tracking Call Time While in Dial()
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
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 before
: [asterisk-users] Asterisk as an IVR solution
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
] Tracking Call Time While in Dial()
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
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
, 2008 4:29:50 PM
Subject: Re: [asterisk-users] Recharge Dial Limit?
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
So, I've been asked if this is possible.
Someone wants to actively monitor the duration of a call, while the call is
still in progress. Obviously, in Asterisk, once the Dial() application starts,
you lose dial plan control until after the call has ended, successful or
otherwise.
Anyone know
Admittedly I have not used the ExternalIVR app. Is it any good?
I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, it can do
it, but boy it is UGLY. There's also the fact that you can't call Backgound()
in a macro, which forces you to use Read() which won't accept a timeout of
@lists.digium.com
Sent: Thursday, July 10, 2008 12:37:31 PM
Subject: Re: [asterisk-users] Asterisk as an IVR solution
From what I can tell Read allows for a floating point input which uses
ast_waitfordigit that accepts milliseconds as input.
Douglas Garstang wrote:
Admittedly I have not used the ExternalIVR app
,
Douglas Garstang wrote:
Don't know about Asterisk 1.4, but in Asterisk 1.2 it expects the
input in seconds. If you try and use 0, it seems to drop back to a
default of 5s.
- Original Message
From: MFH [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I'm using i6net's vxml browser in Asterisk.
I'm trying to work
out how I can return the inputs from a menu or form back into the
Asterisk dial plan. Is there a variable?
The exit tag apparently can be used to return a value (still trying to
work out how to do that), but what about multiple
So, I need to build a complicated IVR with Asterisk, with a lot of back end
hooks. The dial plan itself has a lot of limitations, not the least of which is
that the dial plan is ugly, hard to maintain, and full of gotchas like all
variables being global etc etc.
I've been involved with
I'm using i6net's vxml browser in Asterisk.
I'm trying to work out how I can return the inputs from a menu or form back
into the Asterisk dial plan. Is there a variable? It's not documented if it is.
The exit tag apparently can be used to return a value (still trying to work out
how to do
So, I'm trying to get the Asterisk vxml (from i6net) working.
Having no luck with it.
My dial plan has:
exten = _X.,1,Answer()
exten = _X.,n,Wait(1)
exten = _X.,n,Vxml(file:///tmp/menu.vxml)
The /tmp/menu.vxml file has:
?xml version=1.0?
vxml version=1.0
form
blockaudio
let you use absolute paths?
Wouldn’t it have the equivalent of a
DocRoot???
Alex
From:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Thursday, July 03, 2008 5:03
PM
To: asterisk-users@lists.digium.com
Subject: [asterisk
Argh! I have this...
[ct_start2]
exten = _X.,1,Set(LANGUAGE()=mig33/en/allison-tts)
exten = _X.,n,NoOp(${LANGUAGE()})
exten = _X.,n,Answer()
exten = _X.,n,Wait(1)
exten =
_X.,n,Playback(/var/lib/asterisk/sounds/mig33/en/allison-tts/please-enter-your-pin)
exten =
I just downloaded the app_vxml for Asterisk 1.2 from i6net.
Couldn't get it to work. We're using Asterisk 1.2 still, and it looks like the
app_vxml binary was linked against libstdc_++-5.x (we have libstdc++-6.x). I
grabbed the 1.4 version of the module hoping in vain that would work, but it
Asterisk 1.2, and Cepstral 5, Allison voice.
I execute:
swift Please enter your pin. -o please-enter-your-pin.ulaw -p
audio/channels=1,audio/encoding=ulaw,audio/sampling-rate=8000
then copy it up to /var/lib/asterisk/sounds, and Play() the file.
The sound file seems corrupted. All I hear is
] Building a Complex IVR
On Mon, 2008-06-23 at 09:54 -0700, Douglas Garstang wrote:
I'm about to build a complex IVR with Asterisk.
Having done it a few times with the dial plan, I know it's going to be
pretty ugly. What are my other options? I guess I could do it in
AGI/FastAGI. What about VxML
I'm about to build a complex IVR with Asterisk.
Having done it a few times with the dial plan, I know it's going to be pretty
ugly. What are my other options? I guess I could do it in AGI/FastAGI. What
about VxML (about which I know almost nothing...)?
Using Asterisk 1.2
Thanks,
Doug.
Sent: Monday, June 23, 2008 10:02:37 AM
Subject: Re: [asterisk-users] Building a Complex IVR
On Mon, Jun 23, 2008 at 12:54 PM, Douglas Garstang [EMAIL PROTECTED] wrote:
I'm about to build a complex IVR with Asterisk.
Having done it a few times with the dial plan, I know it's going to be
pretty
are going to complain about recommended solutions, then why ask
in the first place?
Just use FreePBX and copy over the pertinent parts of your conf files...
Thanks,
Steve T
On Mon, Jun 23, 2008 at 1:31 PM, Douglas Garstang [EMAIL PROTECTED] wrote:
Right, except now I have to go write a multi
I would build it this way:
1) Design the dialplan logically so it is understandable and maintainable.
2) Code up the AGIs in whatever language you are comfortable. I would use
C, but that's what I'm most comfortable with.
3) Confirm everything works like you think it should.
4) Measure to
I'm using the SayNumber() app to read out a users balance for an IVR.
Is there a way I can do that while waiting for DTMF input?
Obviously, read() and Background() don't correctly say a number in number
format.
Thanks,
Doug.
___
-- Bandwidth
at 10:03 -0700, Douglas Garstang wrote:
I'm using the SayNumber() app to read out a users balance for an IVR.
Is there a way I can do that while waiting for DTMF input?
Obviously, read() and Background() don't correctly say a number in
number format.
I don't know of an easy way of doing
Here's a weird one. We have a situation where Asterisk seems to be losing it's
ODBC database connection during idle periods. A workaround was to have a script
connect to AMI and generate a bogus call, which would then generate a CDR and
keep the connection alive. We didn't want to be generating
General Asterisk question.
We are sending CDR's to MySQL via odbc. It seems that Asterisk is sometimes
dropping CDR's, and they aren't being sent to the database (they ARE in the
Master.csv file though). We suspect that when the MySQL socket is idle, it gets
disconnected, either by the MySQL
@lists.digium.com
Sent: Wednesday, May 21, 2008 3:02:07 PM
Subject: Re: [asterisk-users] Asterisk Database Handling
Douglas Garstang wrote:
We are sending CDR's to MySQL via odbc. It seems that Asterisk is
sometimes dropping CDR's, and they aren't being sent to the database
- Original Message
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 21, 2008 3:30:06 PM
Subject: Re: [asterisk-users] Asterisk Database Handling
Douglas Garstang wrote:
I couldn't
So... surely this must be a general problem with ANY Asterisk module that
uses the database. Do all modules use the same common database code or do
they all use their own? If they all use their own, I guess idle database
connection issues may be fixed in some modules and not others. If it's
I personally can tell you I've never had a problem with either the
PostgreSQL or MySQL cdr apps themselves losing records. However, I can't
say personally how well the ODBC method works. I'll just stick to saying
that if you're considering using the cdr_mysql addon, I would highly
suggest it
I personally can tell you I've never had a problem with either the
PostgreSQL or MySQL cdr apps themselves losing records. However, I can't
say personally how well the ODBC method works. I'll just stick to saying
that if you're considering using the cdr_mysql addon, I would highly
suggest it
Not at all, just offering a workaround. If your master.csv is
complete and correct then it makes sense to use that data unless
someone can identify your problem and offer a fix.
Unfortunately, not really feesible. I didn't design the system but we are using
CDR's not only for billing purposes,
On Wednesday 21 May 2008 17:02:07 Alex Balashov wrote:
Douglas Garstang wrote:
We are sending CDR's to MySQL via odbc. It seems that Asterisk is
sometimes dropping CDR's, and they aren't being sent to the database
(they ARE in the Master.csv file though). We suspect that when the MySQL
Anyone know where I can find an Alison recording of the word 'per'?
Seems silly to buy the word 'per' from Digiums web site.
And, I'd rather not open up audio editing software and get my 'per' prompt by
editing it out of something else.
Doug.
I haven't done this for a while... yes, that is my excuse.
What the heck is wrong with this?
[general]
autofallthrough=yes
exten = s,n(prompt),NoOp()
exten = s,n,Background(wish-to-continue)
exten = s,n,Background(1-yes-2-no)
exten = s,n,WaitExten(5)
; User entered nothing
exten =
Is it possible to set an outboundproxy for REGISTER from Asterisk?
register = xxx:[EMAIL PROTECTED]
[foobar]
type=peer
host=sip99.foobar.com
disallow=all
allow=g729
canreinvite=no
secret=yyy
fromuser=xxx
port=5099
outboundproxy=xxx.42.149.69
However, SIP REGISTER still goes directly to
Oops, I got that wrong... should have been
register = xxx:[EMAIL PROTECTED]
Doug.
- Original Message
From: Douglas Garstang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 18, 2008 11:56:27 AM
Subject: [asterisk-users] REGISTER Outboundproxy
Is it possible
Using Asterisk 1.2, still.
We are issuing a callback. User rejects the first two calls, but answers the
third. For some reason, the Manager Interface outputs a CDR with disposition
'NO ANSWER' for all three attempts, eventhough the 3rd call worked.
Is this an asterisk 1.2 bug?
Doug.
It's time to ask this question again. Maybe I will get a reply one day. :)
Asterisk
1.4 has some channel variables that you can inspect after a call is
complete that will give you QoS metrics. Stuff like average round trip
time, etc.
Since there's only one set of variables, and calls will
have
- Original Message
From: Jay R. Ashworth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 13, 2008 9:45:34 AM
Subject: Re: [asterisk-users] LCR in Asterisk
On
Wed,
Feb
13,
2008
at
11:33:19AM
-0600,
Tilghman
Lesher
wrote:
On
Wednesday
13
Ok, so I've asked this question before, and didn't get an answer.
So here I go again!
Asterisk
1.4 has some channel variables that you can inspect after a call is
complete that will give you QoS metrics. Stuff like average round trip
time, etc.
Since there's only one set of variables, and calls
Ok, so I've asked this question before, and didn't get an answer.
So here I go again!
Asterisk 1.4 has some channel variables that you can inspect after a call is
complete that will give you QoS metrics. Stuff like average round trip time,
etc.
Since there's only one set of variables, and
I have Asterisk 1.4 registering via IAX to another Asterisk machine.
How can I change the default registration timeout of 60s?
I need my Asterisk box to register every HOUR Anyone?
Editting source isn't an option.
Doug.
Does anyone know the cause of these BAD BAD BAD messages?
I think I lost all my calls when it happened too. We have nagios running
against IAX and nagios reports that IAX is down. It would seem that the entire
application locks up when this happens and calls are dropped.
Connected to Asterisk
We're doing callback here. Asterisk dials a number, waits for an answer, plays
a prompt, dials a second number, and bridges the channels together.
Calls are initiated from the AMI.
No problems there. Easy stuff.
However, I'd like to know if it's possible to have Asterisk dial the same two
Replying to myself. :)
I just noticed the deadlock message still displayed on the console at the end
of a normal call, so the the deadlock message is not related to the early CANCEL
- Original Message
From: Douglas Garstang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent
Hope someone can help.
I have a situation where asterisk is sending a SIP CANCEL message before the
Dial() timeout has hit. It doesn't always do it.
Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a
180 Ringing, or 183 Session Progress. It seems to be at this
When I saw the subject I thought the poster was maybe asking if was possible to
transfer the live RTP stream from one Asterisk system to another in the event
that power was lost
- Original Message
From: MatsK [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
, 2007 10:23:02 AM
Subject: Re: [asterisk-users] Setting Multiple Values via func_odbc ...?
On Thu, 6 Dec 2007, Douglas Garstang wrote:
I need to insert/update multiple MySQL columns in a single row with
the
func_odbc function at the SAME TIME.
If I understand your question correctly
I need to insert/update multiple MySQL columns in a single row with the
func_odbc function at the SAME TIME.
Someone showed me how to use ARRAY to retrieve multiple values at the same
time, but I need to SET multiple values.
Can this be done?
If not, I will just stick with MySQL, but that's a
So... I'm trying to access CDR(duration) and CDR(billsec) inside h...
I keep getting 0. Can I access the CDR function inside a hangup extensions?
Asterisk 1.4.13
Thanks, Doug.
Be a better friend,
: Douglas Garstang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, December 6, 2007 12:04:29 PM
Subject: CDR Function in Hangup Channel
So... I'm trying to access CDR(duration) and CDR(billsec) inside h...
I keep getting 0. Can I access the CDR function inside a hangup extensions
: Re: [asterisk-users] CDR Function in Hangup Channel
On Thursday 06 December 2007 14:54:14 Douglas Garstang wrote:
Ok, this is a little crazy...
billsec and duration are 0, but disposition is ANSWERED.
Huh?
That's correct. Both of those values depend upon the call be ENDED.
If the call
Got it!
endbeforehexten=yes
Wooo!
- Original Message
From: Steve Edwards [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 6, 2007 2:31:54 PM
Subject: Re: [asterisk-users] CDR Function in Hangup Channel
Not strictly an Asterisk question.
I've tried to install adhearsion on TWO relatively fresh CentOS 5.x systems,
and I get this...
[EMAIL PROTECTED] rubygems-0.9.5]# gem install adhearsion
Bulk updating Gem source index for: http://gems.rubyforge.org
ERROR: While executing gem ...
]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 27, 2007 9:08:50 PM
Subject: Re: [asterisk-users] Multiple Return Values from func_odbc
On Tuesday 27 November 2007 20:05:55 Douglas Garstang wrote:
Is there any way to return multiple
]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 27, 2007 9:08:50 PM
Subject: Re: [asterisk-users] Multiple Return Values from func_odbc
On Tuesday 27 November 2007 20:05:55 Douglas Garstang wrote:
Is there any way to return multiple
Is there any way to return multiple values from functions defined in
func_odbc.conf?
It appears that you can only return one value.
True? Hope not
Doug.
Be a better pen pal.
Text or chat with
I'm a little confused. I'd like to build an RPM for Asterisk 1.4.
Is it better to modify and use the spec file under redhat/asterisk.spec and run
a 'make rpm', OR is it better to build a custom spec file from scratch and use
'rpmbuid -ba' specfile?
How do people normally do it?
The problem I
Does anyone know where I can get an rpm spec file for zaptel 1.4.x?
Thanks,
Doug.
Be a better sports nut! Let your teams follow you
with Yahoo Mobile. Try it now.
I'm a little confused. I'd like to build an RPM for Asterisk 1.4.
Is it better to modify and use the spec file under redhat/asterisk.spec and run
a 'make rpm', OR is it better to build a custom spec file from scratch and use
'rpmbuid -ba' specfile?
How do people normally do it?
The problem I
- Original Message
From: Richard Lyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 1, 2007 8:47:28 AM
Subject: Re: [asterisk-users] AEL2 and Callbacks
Douglas Garstang wrote:
I am originating
I am originating a command via the AMI with this...
Action: Login
Username: xxx
Secret: yyy
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: Local/[EMAIL PROTECTED]
Callerid: 849120
Context: default
ActionID: 849120
My LegA context:
---
context LegA {
Anyone know if the MySQL() application has a configurable timeout?
If it tries to connect to a bogus IP, it's timeout seems to be a few minutes.
I'd like to cut it down to a few seconds.
Doug.
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has
Subject: Re: [asterisk-users] MySQL() timeout
Douglas Garstang wrote:
Anyone know if the MySQL() application has a configurable timeout?
If it tries to connect to a bogus IP, it's timeout seems to be a few
minutes.
I never got a response on that question myself.
Doug
--
Ben Franklin quote
I'm confused about something.
It's the way Asterisk handles the A leg (ie the first party dialed) on an
originate command via the Manager Interface.
Lets say our originate commands looks like this:
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: SIP/[EMAIL PROTECTED]
I'm trying to build an Asterisk rpm from the supplied asterisk.spec file.
Made numerous changes to get it to work.
The architecture of the system I am building on is x86_64. I'd like to build
for i686 though.
I added a --target i686 to the rpmbuild line in the Makefile, but it looks like
it's
environments.
-Philip
Douglas Garstang wrote:
I'm trying to build an Asterisk rpm from the supplied asterisk.spec
file.
Made numerous changes to get it to work.
The architecture of the system I am building on is x86_64.. I'd like
to
build for i686 though.
I added a --target i686
Ah jeez. All I wanted to do was connect to a carrier and then perform fail over
logic based on their SIP response.
Not supposed to be difficult. This is what Asterisk is supposed to be good at.
We have a SIP module, why not have SIP responses available to the module.
Now, I have to look at the
packets coming into ur system and in
those packets you can see the response code.
On 10/25/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I'd like to grab the SIP response code that comes back from an INVITE. The
HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get
the SIP
I'd like to grab the SIP response code that comes back from an INVITE. The
HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get
the SIP response code instead?
Doug.
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the
Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that output
applies to, to the start of each line? If you are proxying multiple systems,
how can it uniquely identify the output from each system?
Thanks,
Doug.
__
Do You
?
Douglas Garstang wrote:
Can the Asterisk Manager Proxy, AstManProxy, prefix the host name
that output applies to, to the start of each line? If you are proxying
multiple systems, how can it uniquely identify the output from each
system?
Thanks,
Doug.
each Event block should have a
Server
Is it well known that setting the ActionID when connecting to AMI has
absolutely no effect?
Is this fixed in Asterisk 1.4?
If you add an ActionID to your Originate command for example, it looks like the
only events that come back with an ActionID associated are the initial
response,
- Original Message
From: SIP [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Wednesday, September 26, 2007 4:31:08 AM
Subject: Re: [asterisk-users] Asterisk Redundancy
Per Jessen wrote:
Atis Lezdins wrote:
This seems nice way
- Original Message
From: Scott Moseman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 26, 2007 6:07:06 AM
Subject: Re: [asterisk-users] Asterisk Redundancy
On 9/26/07, SIP [EMAIL PROTECTED] wrote:
It's nice to see Asterisk redundancy being discussed. A year and half ago, when
I posed the question of Asterisk redundancy, I was looked at like I was from
outer space.
- Original Message
From: Jared Smith [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Nagios that's not redundancy.
- Original Message
From: Dave Walker [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, September 25, 2007 9:09:46 AM
Subject: Re: [asterisk-users] Asterisk Redundancy
On Tue,
- Original Message
From: Atis Lezdins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, September 25, 2007 2:11:10 PM
Subject: Re: [asterisk-users] Asterisk Redundancy
On 9/25/07, Philipp Kempgen [EMAIL
Wow. Polycom phones are STILL doing that? I haven't been involved with Polycom
phones since before January, and it was a problem back then too. Jeez
- Original Message
From: Gregory Boehnlein [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, September 21, 2007
I'm confused about something
In Asterisk 1.4 you can collect RTP QoS metrics at the end of a call with:
${CHANNEL(rtpqos,audio,all)}
Now, when your using the AMI to do a callout, like this...
ACTION: Originate
Async: yes
Timeout: 6
Exten: callback
Channel: SIP/1000
Variable:
I have two Asterisk Systems. One on of those, when I execute this:
Dial(SIP/teleglobe-007931d0,
SIP/[EMAIL PROTECTED]|60|oL(400752:6:3))
... It causes Asterisk to immediately read out the time limit of the call
(66,792 minutes), as soon as the other end answers, even though we aren't
Ok, so I'm no rpm expert, but Asterisk 1.4.11 comes with an asterisk.spec
file. Running rpmbuild against it yields errors, the first one being that
the 'Copyright' tag is unknown, and that I need a License tag instead.
Fixed that, and...
Processing files: asterisk-CVS-1
error: File not found:
I'd like to know why the spec file is even included at all then?
I think we'd prefer to build our own, rather than trust someone elses build.
On 9/19/07 3:22 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Sep 19, 2007 at 02:54:17PM -0700, Douglas Garstang wrote:
Ok, so I'm no rpm expert
in Asterisk
I have used this freeware tool in the past:
http://sineapps.com/sinestatiax.php
maybe you can have a look at it as well
l.
In data Thu, 09 Aug 2007 02:07:49 +0200, John Todd [EMAIL PROTECTED]
ha
scritto:
At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:
At 12:31 PM
:33 PM -0700 2007/8/3, Douglas Garstang wrote:
At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
How can I objectively measure jitter in Asterisk on a SIP
channel?
I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.
Thanks,
Doug
:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, August 09, 2007 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
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