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 1.
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.
_
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 calls w
- 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
Feb
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 tw
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.
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 sip99.fo
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
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 => t,1
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 are
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: /tm
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
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 2:3
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: callid=8491
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
Sent: Tuesday, September 25, 2007 9:09:46 AM
Subject: Re: [asterisk-users] Asterisk Redundancy
On Tue, 2007-09-25 at 18:01 +0200, Phil
>- Original Message
>From: Atis Lezdins <[EMAIL PROTECTED]>
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>
>Sent: Tuesday, September 25, 2007 2:11:10 PM
>Subject: Re: [asterisk-users] Asterisk Redundancy
>
>On 9/25/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote:
>> Adrian M
>- 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:
>>
>>
>>> Th
>- Original Message
>From: Scott Moseman <[EMAIL PROTECTED]>
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>
>Sent: Wednesday, September 26, 2007 6:07:06 AM
>Subject: Re: [asterisk-users] Asterisk Redundancy
>
>On 9/26/07, SIP <[EMAIL PROTECTED]> wrote:
>>
>> No. It's not.
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 Yahoo!?
Thanks, just realised that...
- Original Message
From: Richard Lyman <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, October 24, 2007 10:45:25 AM
Subject: Re: [asterisk-users] AstManProxy Host Prefix?
Douglas Garstang wrote:
>
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, OriginateSu
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
g 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
th
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 lo
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]
Variabl
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 st
cross-development 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
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 the
erisk-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 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 {
>- Original Message
>From: Richard Lyman <[EMAIL PROTECTED]>
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>
>Sent: Thursday, November 1, 2007 8:47:28 AM
>Subject: Re: [asterisk-users] AEL2 and Callbacks
>
>Douglas Garstang wrote:
>> I
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' ?
How do people normally do it?
The problem I see with a
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' ?
How do people normally do it?
The problem I see with a
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.
http://mobile.yahoo.com/sports;_yl
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 fr
TED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion
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 values
TED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion
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 values
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 ... (Errno::ENOENT
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, this ca
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, news
--
From: 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 functi
-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
Sent: Thursday, December 6, 2007 2:31:54 PM
Subject: Re: [asterisk-users] CDR Function in Hangup Channel
On Thu, 6 Dec 2007, Joshua
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 - N
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
number
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.co
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 point
I'm trying to find a high port count ATA device. We have a lot (up to
110) analog devices that we need to bridge to IP. Rather than go out and
buy 110 ATA's, it would make more sense to buy a single chassis type
device with some number of ports and blades. Anyone know if such a
device exists?
T
drews
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, May 31, 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] High Port Count ATA
I'm trying to find a high port count ATA device.
I previously worked for a company that did some heavy load testing with
Asterisk on multiple core Sun systems. We saw that no matter how many
cores you threw at Asterisk, it always used ONE core to process calls,
even at very high loads.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[
Speaking of SQLite, is there an Asterisk SQLite command?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, June 01, 2007 9:41 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk mysql support
On Fri, Jun
Where's Steve Murphy when you need him? :-)
This doesn't seem to work in AEL2...
Macro foo(arg1) {
.
Includes {
Hangup;
}
}
The error is: File: /etc/asterisk/extensions.ael, Line 59, Cols: 5-12:
Error: syntax error, unexpected KW_INCLUDES, expecting '
Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in
Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will
clear any extensions that have been created by regexten. This is VERY
bad!
Doug.
___
--Bandwidth and Co
context defined as both
your regcontext and as a context in extensions.conf (or an .ael, or
whatever).
- Brad
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Douglas Garstang
> Sent: Wednesday, June 06, 2007 7:08 PM
> To
I don't know if this is possible, and I can't quite get my head around
how to do it...
If I am using DUNDi for redundancy in a cluster, when Phone1 makes a
call to Phone2, both Asterisk A and B will be in the RTP stream:
+---+ +---+
| A |-| B |
/+---+ +---+\
Does anyone know how the Linksys PAP2T ATA's can be mass provisioned?
Documentation seems to be sketchy, even on the Linksys web site.
Thanks,
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBS
How do you get PAP2T-NA's? They aren't even on Linksys's web site.
-Original Message-
From: Doug [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 07, 2007 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Douglas Garstang
Subject: Re: [asterisk-user
sys PAP2T ATA's
On Thu, 7 Jun 2007, Douglas Garstang wrote:
> Does anyone know how the Linksys PAP2T ATA's can be mass provisioned?
> Documentation seems to be sketchy, even on the Linksys web site.
If it's like the pap2, you can use tftp and xml. This should get you
sta
...
>
> On 6/7/07, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> > If I am using DUNDi for redundancy in a cluster, when Phone1 makes a
> call to
> > Phone2, both Asterisk A and B will be in the RTP stream:
>
> Correct so far... although once the call is made,
...
>
> On 6/7/07, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> > If I am using DUNDi for redundancy in a cluster, when Phone1 makes a
> call to
> > Phone2, both Asterisk A and B will be in the RTP stream:
>
> Correct so far... although once the call is made,
k-users] DUNDi and reinvites...
>
> Douglas Garstang wrote:
>
> >
> > Let's just say we only configured the originating phone with
> > canreinvite=yes, which hopefully means the originating phone would
> > reinvite with the second Asterisk server. That's all fine
Can anyone recommend any wholesale SIP termination providers that will
automatically charge a credit card? Most seem to want upfront payment
and a credit balance but that sucks when you have to watch it like a
hawk to make sure the balance never hits zero. I'm looking for a
provider that can automa
We're dialing a disconnected number via Level 3's vector network, and
are receiving this. The response has SDP in it. Apparently, Level 3 is
playing early media. Asterisk doesn't seem to know what to do with SDP
in a 180 RINGING, and just plays ringing. What am I missing here? How
can Asterisk see
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Alex Balashov
> Sent: Monday, June 18, 2007 5:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] 180 Ringing with SDP
>
> On Mon, 18 Jun
I have this in my dialplan...
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten => 5000,1,Answer
exten => 5000,n,Wait(1)
exten => 5000,n,NoOp(${CALLERID(num)})
exten => 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this...
== Parsing '/et
I have this in my dialplan...
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten => 5000,1,Answer
exten => 5000,n,Wait(1)
exten => 5000,n,NoOp(${CALLERID(num)})
exten => 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this...
== Parsing '/et
I have this in sip.conf:
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
progressinband=yes
[19256002182]
type=friend
username=19256002182
callerid="Test hone 1" <+19256002182>
host=dynamic
canreinvite=no
secret=password
context=test
disa
Is it possible to have Asterisk allow the From address in a SIP invite
to be different to the required digest username?
The auth parameter supposedly allows it, but whether or not I set auth
to be what the UA sends as the digest username, Asterisk just complains
that the from and the digest are di
have asked this questions,but have no answer :) I also want Asterisk
do not check "to" head with digest username in registration,how can we
do that?
On 7/12/07, Douglas Garstang <[EMAIL PROTECTED]> wrote:
Is it possible to have Asterisk allow the From address in a SIP invite
to be
ubject: Re: [asterisk-users] Different SIP From and Auth?
Hi
I have asked this questions,but have no answer :) I also want Asterisk
do not check "to" head with digest username in registration,how can we
do that?
On 7/12/07, Douglas Garstang <[EMAIL PROTECTED]> wrote:
Is it possi
Don't know about the 320, but we provisioned the 301's. They're
provisioning is basically the same as the 501's and 601's. What problems
are you having?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Doug
> Sent: Wednesday, August 01
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of SIP
> Sent: Wednesday, August 01, 2007 1:05 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Retail DID provider ?
>
> IdeaSIP, Voxbone, G
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ira
> Sent: Thursday, August 02, 2007 10:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Teliax Quality of Service
>
> At 09:23 AM 8/2
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.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asteri
[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, August 03, 2007 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Measuring Jitter in Asterisk
How can I objectively measure jitter in Asterisk on a SIP channel?
I don't just want t
n Asterisk
>
> On Fri, 3 Aug 2007, Douglas Garstang wrote:
>
> > If it COULD, you could leave a tshark process running, constantly
> > measuring jitter in real time. You'd run one for each ITSP you use,
and
> > voila, you have real time jitter metrics on a provide
terisk
>
> On Fri, 2007-08-03 at 12:31 -0700, 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.
>
> You can use Wiresha
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Alex Balashov
> Sent: Friday, August 03, 2007 2:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
>
> On Fr
risk
>
> 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,
>
risk
>
> 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,
>
I don't think creating a network without a single point of failure is
unreasonable.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Stephen Bosch
Sent: Sat 8/4/2007 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
> I don't think creating a network without a single point of failure
> is unreasonable.
It's impossible. I can&#
>
> Steve Totaro wrote:
> > Anthony Francis wrote:
> >
> >> Tim Panton wrote:
> >>
> >>
> >>> On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
> >>>
> >>>
> >>>
> >>>
> >>>> I do
Service
>
> Eric "ManxPower" Wieling wrote:
> > Douglas Garstang wrote:
> >> Let's assume for a moment that it's impossible. That does not mean
> adding additional servers and additional networking equipment does not
add
> value, or is a worthless endea
>
> Steve Totaro wrote:
> > Anthony Francis wrote:
> >
> >> Tim Panton wrote:
> >>
> >>
> >>> On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
> >>>
> >>>
> >>>
> >>>
> >>>> I do
Quality of Service
>
> Douglas Garstang wrote:
> >> -Original Message-
> >> From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> >> [EMAIL PROTECTED] On Behalf Of SIP
> >> Sent: Monday, August 06, 2007 8:56 AM
> >> To: Asterisk Users Mailing List
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Stephen Bosch
> Sent: Tuesday, August 07, 2007 10:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Teliax Quality of Service
>
> Brian
Quality of Service
>
>
> On Aug 6, 2007, at 10:42 AM, Stephen Bosch wrote:
>
> > Eric "ManxPower" Wieling wrote:
> >> Douglas Garstang wrote:
> >>> Let's assume for a moment that it's impossible. That does not
> >>> mean adding
easuring Jitter 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:
&g
Jitter in Asterisk
>
> At 3:33 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?
> >> >
> >> >
:[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
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?
The Asterisk Manager Interface is driving me nuts.
Whoever wrote it should be drawn and quartered.
Sometimes the data comes back separated by \r\n, and sometimes it's separated
by \n.
The whole thing is completely inconsistent, and trying to write any kind of API
for it is -GHASTLY-
Doug.
_
> -Original Message-
> From: Michael Collins [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, November 29, 2006 11:20 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
>
>
> > Sometimes the data comes back
> -Original Message-
> From: Matt Florell [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, November 29, 2006 12:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] What's up with the Manager Interface?!?!
>
>
>
> -Original Message-
> From: Douglas Garstang
> Sent: Wednesday, November 29, 2006 12:26 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
>
>
> > --
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