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
This might get you going:
http://www.voip-info.org/wiki/view/Asterisk+cmd+AbsoluteTimeout
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> JR Richardson
> Sent: Friday, August 03, 2007 1:49 PM
> To: asterisk-users@lists.digium.com
> Subject: [a
I am having a serious problem with agents being logged out of the queue after
they finish a call. I am using static agents and agents.conf. I am running
2.1.17. Anyone having these problems or could think of anything that would
cause them.
Jordan Novak
Telecommunications Engineer
__
Hi,
I have a client who has a system with a Sangoma 1 port PRI card with
echo canceling in it.For some reason, when the system comes up the
PRI will stay up for about 4-5 hours, then drop. "zap show status"
shows everything as ok, but we can't make or receive any calls until
the system is reb
Trevor G. Hammonds wrote:
> From: SIP
> Sent: Saturday, August 04, 2007 2:57 PM
>
>
>> Stephen Bosch wrote:
>>
>>> Douglas Garstang wrote:
>>>
>>>
I confused by this. Don't ITSP's have redundancy? Don't they have
multiple edge systems for accepting incoming calls? Don't
On Sat, Aug 04, 2007 at 10:05:34PM -0400, Baji Panchumarti wrote:
> On 8/4/07, Matt wrote:
>
> > This may sound stupid.. so bear with me for a moment.
> >
> > Assuming the only access I have to a machine is through asterisk -rx
> > can I use the ! command?
> >
> > asterisk -rx help
> >
> > incl
On Sat, Aug 04, 2007 at 09:16:22PM -0400, Matt wrote:
> This may sound stupid.. so bear with me for a moment.
>
> Assuming the only access I have to a machine is through asterisk -rx
> can I use the ! command?
>
> asterisk -rx help
>
> includes the ! command, but I can't seem to get it to work i
On 8/4/07, Matt wrote:
> This may sound stupid.. so bear with me for a moment.
>
> Assuming the only access I have to a machine is through asterisk -rx
> can I use the ! command?
>
> asterisk -rx help
>
> includes the ! command, but I can't seem to get it to work ie:
>
> asterisk -rx "! ls"
>
>
satish patel wrote:
> dear all
>
> is there any GUI application with support asterisk 1.2
> version i am useing 1.2 and i have fine more about GUI base
> configuration but i didnt got any GUI package for asterisk 1.2
>
>
If you're a windows user, you can also check out DialplanP
Stephen Bosch wrote:
> John Novack wrote:
>
>> Stephen Bosch wrote:
>>
>>> Well, this is approaching the absurd.
>>>
>>> Do you know how many Meridian systems have radios plugged into them for
>>> "on-hold" background sound? Nobody pays royalties on those.
>>>
>>>
>> IF they ar
This may sound stupid.. so bear with me for a moment.
Assuming the only access I have to a machine is through asterisk -rx
can I use the ! command?
asterisk -rx help
includes the ! command, but I can't seem to get it to work ie:
asterisk -rx "! ls"
Any help?
__
From: SIP
Sent: Saturday, August 04, 2007 2:57 PM
>Stephen Bosch wrote:
>> Douglas Garstang wrote:
>>
>>> I confused by this. Don't ITSP's have redundancy? Don't they have
>>> multiple edge systems for accepting incoming calls? Don't their multiple
>>> edge systems have multiple interfaces, con
Stephen Bosch wrote:
> Douglas Garstang wrote:
>
>>I confused by this. Don't ITSP's have redundancy? Don't they have
>>multiple edge systems for accepting incoming calls? Don't their multiple
>>edge systems have multiple interfaces, connected to multiple subnets,
>>via multiple switches? And, don'
Iax channel can be encrypted.
Not just the authentication, even rtp data, see:
http://www.voip-info.org/wiki/view/IAX+encryption
On 8/4/07, Michael Munger <[EMAIL PROTECTED]> wrote:
>
> IAX is not encrypted. What you're seeing in wireshark is likely the
> authentication method you've chosen. (RSA
I currently have an AGI that calls the Festival text2wave app to write
a wav file that my dialplan plays into a call with the Background()
command. But the voice sounds terrible: like SAM, the 1980s 6502 voice
synthesizer. I tried to slow it down by calling (text2wav -eval
"(Parameter.set '
I have a TDM800 that is installed and working. (TDM800 + 2 X QUAD FXO).
Zttool says it is configured, ok, and there are no issues.
Ztcfg -vvv shows that all the channels are configured.
Zap show channels in the CLI show all 8 channels configured as they are
supposed to be.
When I plug in
SIP wrote:
> There are also lots of big carriers masquerading as big carriers. ;)
*lol*
> If the ONLY people who could get into the business were the ones who
> could, before offering any services to customers, afford to build out
> multiple edge systems for accepting incoming calls, each with
John Novack wrote:
>
> Stephen Bosch wrote:
>> Well, this is approaching the absurd.
>>
>> Do you know how many Meridian systems have radios plugged into them for
>> "on-hold" background sound? Nobody pays royalties on those.
>>
> IF they are discovered by ASCAP and receive a letter demanding
Stephen Bosch wrote:
> Douglas Garstang wrote:
>
>> I confused by this. Don't ITSP's have redundancy? Don't they have
>> multiple edge systems for accepting incoming calls? Don't their multiple
>> edge systems have multiple interfaces, connected to multiple subnets,
>> via multiple switches? And
Steve,
On 8/3/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
> I just tried to call in after creating an account.
>
> After the call connects, enter the show id: 22622# and your_PIN#
>
> I dial in and am asked for the podcast id, I enter 22622# and am told
> that my passcode is not correct. I also t
On 05:27, Fri 03 Aug 07, bilal ghayyad wrote:
> Hi List;
>
> What is the difference between WaitExten function and
> TIMEOUT (response)? As I see that both are used to
> determine the allowed time to enter the digits, any
> one can advise?
WaitExten is waiting for you to type an extension.
TIMEOU
Dear Michael;
I understood it in that way (please advise me if I am
correct):
WaitExten is for the time to complete entering the
digits, while timeout is specified wether user
responded by dialing any thing or not.
Please advise.
regards
The difference is in the scope of the command.
Think of
Exactly, with the amount of royalty free music out there why bother.
Just go searching for some you like, download it and while you are at it
tip the author/performer a couple of bucks into their myspace tip jar or
similar.
For $10 why take the risk with ascap.
Regards,
Dean Collins
Cognation
On 8/4/07, John Vogel <[EMAIL PROTECTED]> wrote:
>
> Hi!
>
> My application needs to look up (by spelling) the first and last names of
> a
> person and then insert the corresponding pre-recorded audio file to
> personalize the message. E.g. "Hi, John Brown. Your book is due back at
> the
> library.
What modules do you want on it?
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MOSBAH
ABDELKADER
Sent: Saturday, August 04, 2007 3:16 PM
To: asterisk-users@lists.digium.com
Subject:
I'm using asterisk 1.2 with debian and I have configured a SIP account in * and
using it trought X-Lite. My problem is that I can't do phone call(voip or PSTN)
to the SIP account(a friend account) because it always answer ocupped.
Using this SIP account on ATA it doesnt happen and i can do and r
Stephen Bosch wrote:
>
> Well, this is approaching the absurd.
>
> Do you know how many Meridian systems have radios plugged into them for
> "on-hold" background sound? Nobody pays royalties on those.
>
IF they are discovered by ASCAP and receive a letter demanding payment
they will. Not abs
Hello,
Have i to buy an asterisk card like TDM400P to connect the two asterisk
servers with frame relay.
Thanks.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
This seems like something I should know... but I don't.
How do you update zaptel / libpri on a Business Edition box running
rPath? Tried running conary, but got 'Insufficient permission to access
server conary.digium.com."
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
Hi,
I have an asterisk and a quintum AFT200 with two FXO ports, and want
to use it as a gateway to handle outgoing and incoming calls.
I have found this thread,
http://lists.digium.com/pipermail/asterisk-users/2005-February/084015.html
But I think I need a little more help, could anyone knows wh
Hi!
My application needs to look up (by spelling) the first and last names of a
person and then insert the corresponding pre-recorded audio file to
personalize the message. E.g. "Hi, John Brown. Your book is due back at the
library." So I need "John" and "Brown" in audio files along with LOTS of
o
Douglas Garstang wrote:
> I confused by this. Don't ITSP's have redundancy? Don't they have
> multiple edge systems for accepting incoming calls? Don't their multiple
> edge systems have multiple interfaces, connected to multiple subnets,
> via multiple switches? And, don't they have multiple upstr
Steve Kennedy wrote:
> On Tue, Jul 31, 2007 at 05:22:20PM -0400, Jon Pounder wrote:
>
>> Quoting John Millican <[EMAIL PROTECTED]>:
>> there are plenty of radio stations with internet feeds of their audio,
>> piping that in would not change any coverage area since anyone with
>> internet could
Doug wrote:
> At 21:59 7/29/2007, Paul Hales wrote:
> >
> >I even got a Polycom here saying "I'll be back" which was funny for
> >about an hour, then not funny at all.
> >
> >PaulH
>
> Kewwl! How do you get the .wav files into the Polycom?
If it's not obvious, I'd be interested in this info
Deepak Naidu wrote:
>>>It would help to know exactly what Dell Poweredge you were considering.
>>>They do vary.
> I have Dell Power Edge 850
>
>>> Also how do I enable DTMF hardware detection.
> There are no drivers which support it. I have the lastest Beta drivers
> installed, they seem to show y
Hi
OutCALL 1.40 is released. It is available in two flavours:
- Without extension authentication
- With extension authentication
Changelog:
OutCALL 1.40 (2007-06-29):
- Multi-language support (French-Canada is included in the setup, while the
English PO file is distributed with OutCALL setup
On 8/4/07, Steve Underwood <[EMAIL PROTECTED]> wrote:
> Why are people so determined to break things. If you want to use
> unicall-0.0.3pre11, use it with spandsp-0.0.2.
Not really determined to break things, but to understand failures,
even when those failures are because of version missmatching :
Steve Totaro wrote:
> It says clearly in the email from Digium. The link is the same.
> ftp.digium.com but no FTP. Sounds silly to me. So it is
> http://ftp.digium.com.
We added http://downloads.digium.com/. We will be using that URL for all of
our
links from now on.
--
Russell Bryant
S
JR Richardson wrote:
> Thank you, Alex.
> As I've said many times, this community has the smartest people in the
> world. It is with great humbleness, I offer this to all.
>
> New Asterisk Proverb:
>
> Asterisk is like an onion with many, many layers. With 160+ applications
> and seemingly endles
It says clearly in the email from Digium. The link is the same.
ftp.digium.com but no FTP. Sounds silly to me. So it is
http://ftp.digium.com.
Thanks,
Steve Totaro
Michael Munger wrote:
> So where will the files be then? What will the new link be?
>
> Yours,
>
> Michael Munger, dCAP
> 404-4
So where will the files be then? What will the new link be?
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Thursday, July 26, 2007 1:11 PM
To: asterisk-users@lists.digium
> On Fri, 3 Aug 2007, JR Richardson wrote:
>
> > Can anyone point me int he right direction?
>
>At the risk of coming off in a gratuitiously self-aggrandising manner
> quoting myself:
>
>http://lists.digium.com/pipermail/asterisk-users/2007-May/188438.html
>
> --
> Alex Balashov
Thank
The difference is in the scope of the command.
Think of it this way:
WaitExten gives the user more time to enter digits before the dialplan
moves on to the next instruction in the dial plan. Timeout is the max
number of seconds to wait at any point in the current context before
deciding the user
ok, i've taken a look at the actual app_chanspy.c and the newest
i've tried to comment
ast_set_flag(&csth.spy, CHANSPY_MIXAUDIO);
and recompile asterisk, now i can hear only the input stream of the
channel spyed.
that's fine!
thansk
On 8/4/07, nik600 <[EMAIL PROTECTED]> wrote:
> i'm taking a l
How do you disable musiconhold for a single sip peer? I see that there is a
musicclass setting, but what do you set it to so that you disable musiconhold?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing l
Sounds like you have call waiting on the phones. You can disable this
on the Asterisk side. To verify, make a call on your phone and then
dial yourself from another phone. Depending on the phone, you will have
some sort of indication that a second call is coming in.
Thanks,
Steve Totaro
sat
I have seen this "No Authority Found" many times. Not sure what the fix
was, I just kept playing with it until I got it working. I suggest
getting rid of the inkeys, auth=rsa, and adding a secret. Make the
username and passwords the same on both sides as well.
If that works, then you know so
You could use SIP if the servers are on routable IPs or the same subnet,
if not you could use IAX but I think OpenVPN is your best choice for
using SIP over different NATed networks.
I do not think you need any hardware except for what is needed for the
Frame Relay. QoS and traffic shaping wou
Moises Silva wrote:
> I would not call that properly a fix. We need to know why is failing
> in newer spandsp versions in the first place. Can you make a diff and
> post it?
>
Why are people so determined to break things. If you want to use
unicall-0.0.3pre11, use it with spandsp-0.0.2.
The la
IAX is not encrypted. What you're seeing in wireshark is likely the
authentication method you've chosen. (RSA or MD5)
You can encrypt it with a VPN as long as you have a pipe fat enough to
deal with the overhead a VPN puts on packets.
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
On Sat, Aug 04, 2007 at 01:42:35AM -0700, satish patel wrote:
> dear all
>
> is there any GUI application with support asterisk 1.2
> version i am useing 1.2 and i have fine more about GUI base
> configuration but i didnt got any GUI package for asterisk 1.2
freepbx?
destar?
voice
I would be grateful for some comments on our proposed machine specs for
a new Asterisk installation at a client with an initial 70 extensions.
The system should be able to handle 100 extensions. The system will have
the following main features:
- PSTN connection via ISDN 30, dealing with all i
Hello all,
I have to connect two Asterisk servers with a frame relay connection but i
do not know what is the hardware to use and how to connect them.
Have anyone an idea about that.
Thanks.
___
--Bandwidth and Colocation Provided by http://www.api-dig
Hi,
I have two asterisk servers and I want to make these servers call each
other as they were internal. I have succeeded in one way. Server B can
call Server A without problem, but Server A cannot call Server B.
Here's the iax configuration of servers
Server A:
==
[ipek]
auth=rsa
dear all
is there any GUI application with support asterisk 1.2 version i
am useing 1.2 and i have fine more about GUI base configuration but i didnt got
any GUI package for asterisk 1.2
-
Sick sense of humor? Visit Yahoo! TV's Comedy with
Dear all
I have setup of asterisk 1.2.14 with 100 SIP phone and it is
working fine but thing is that when i call to somebody on local extention my
asterisk not give me notification like party phone is busy or busy tone alway
it give me rining single how can i justify other part
Hi.
If you are beginner, that is good book. If you get more depth skill, you need
find out
another book.
Han
"clive.chan(atn)" <[EMAIL PROTECTED]> 쓰기:
Hi all,
Can some one tell me about the book name call “ Asterisk Configuration Guide”
comment? Or Review about this
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