I have several providers that do NOT allow using your own CID. On one
of these, I suppress outgoing CID simply because I don't want people
calling us back on that number.
When I dial out through this SIP provider to an incoming number of
another SIP provider (for testing dialplan stuff, since
On 3/10/06, AR Tarzi [EMAIL PROTECTED] wrote:
That's because the duration is counted from the time of dialling. billsec is
what you want if it's to calculate the duration the call was active.
To change what shows you need to change call-log.php in
/var/www/html/admin/cdr/
Instead of
I'm having an apparent issue where caller id name isn't coming through
my IAX2 channels. The name shows up in the asterisk cdr log, but my IAX2
application doesn't receive it. I'm running asterisk 1.2.4.
Just tried it on IAXCOMM with 1.2 and it worked fine.
Is this a known problem or config
I am using asterisk-1.2.1-15 and want to use itto replace my normal PBX with it. For creating IVR menus i tried festival, the text which was passed into it was said, but the problem was at stating of every line a "tick" sound comes.As festival app in asterisk connects with festval server at
This should be called auto-secretary or auto-receptionist or
something like that since it's exactly what a receptionist does.
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11 mar 2006 kl. 07.54 skrev Douglas Garstang:
Hi JR. I'm dying to know... where'd you find your DUNDi
documentation? Has something new appeared since I looked at it 2-3
months ago? The O'Reilly book's DUNDi section was impossible to
follow, and the examples in the Asterisk DUNDi config
I have two issues...
First I am working with a hotel software vendor to include an automated way to
turn vmail on and off while clearing it at the same time. The vendor is looking
to interface via serial cable as they currently do with Mitel systems. i am
willling to work with them on an IP
There's the g-option for the Dial-cmd that allows to execute the next
extensions in the current context when the callee hangs up.
I would need the same for a call where the caller hangs up, concretely
i have to inform a agi-application of the end of a call. Does someone
know a way to do this
You can use DeadAGI.
exten = _X.,1,DeadAGI(agicall.agi,${EXTEN})
now in that AGI (pseudocode)
$exten=Get parameter 1
$dialstring=SIP/mytrunk/.$exten;
$res=$agi-dial($dialstring),
//If we used deadagi, if the _caller_ hangs up, the agi keep runing here
$chres = $agi-channel_status();
Hi Chris,
Chris Earle (CBL) wrote:
I've got a Junghanns QuadBRI card which I'm going to install on a system in
Germany
Anyone give me some tips on the Jumper settings? I'm guessing it's going to
be NT mode with p2p? I haven't used ISDN before.
I'm going to also put a Digium TDM400P card in
I'm working on an IVR that gives the users the option (number 5 in the main
menu) to dial by extension:
exten = 5,1,Set(TIMEOUT(digit)=5) ; Dial Extension
exten = 5,2,Set(TIMEOUT(response)=10)
exten = 5,3,Background(LCL/prompt-60)
exten = 5,4,WaitExten(15)
When going option 5 you
Greetings from Hack in The Box -- We are pleased to announce that the
Call for Paper (CfP) for HITBSecConf2006 - Malaysia is now open! Set to
take place from September 18th - 21st 2006 at The Westin Kuala Lumpur,
this years conference promises to once again deliver an International
deep-knowledge
Hi all,
Can somebody help me out, to get the call going through to my provider?
I connected my A104D Sangoma card to E1/isdn and each I tried to make call I
get the errors below.
My protocol analyzer can see only setup info and release complete.
WIRELESS2*CLI set debug 9
Core debug was 0
Jordan Novak wrote:
Warning: This message has had one or more attachments removed
Warning: (winmail.dat).
Warning: Please read the NetConcepts-Attachment-Warning.txt attachment(s) for
more information.
I have two issues...
First I am working with a hotel software vendor to include an automated
When going option 5 you can dial some extensions such as 2802, it goes to the
extension (all extens start with 28 on the
system). However, just dialing something random like 2929 sends the caller
to option 2 of the main menu or 1010 sends
the caller to menu option 1 from the main menu.
http://nerdvittles.com/index.php?p=122
On 3/10/06, Sharath Chandra [EMAIL PROTECTED] wrote:
How can i configure the following scenario,
- User 'A' dials into Asterisk,
- Asterisk puts user 'A' on hold
- Dials Out to User 'B'
- Consults user B' if he wants to take the call (Press 1) or
On 10 Mar 2006, at 21:25, Michael van Rooyen wrote:
I had a working 1.0.9 asterisk installation and tried to get a
Firefly IAX phone to register, but it was failing. I upgraded to
asterisk 1.2.5 and the PBX is working fine, but the IAX phone still
won't connect. Below is my iax.conf and
The G.726 codec is the current Asterisk 1.2 version (revision 7221). I
am using G.711a (alaw) between a Sipura ATA and Asterisk at each end of
the link and am testing alternative codecs on an IAX link (not in trunk
mode) between the two Asterisk servers. (Yes, I know that the Sipura can
do
Brillian, thanks very much. I had actually tried the second context myself but
looking back at my attempts I can see
where I went wrong.
Thanks again for the help.
Cheers!!!
Time Bandit wrote:
When going option 5 you can dial some extensions such as 2802, it goes to the
extension (all extens
Anthony Rodgers wrote:
Greetings,
I have just updated our test server to 2.6.9-34.EL and get the following
error messages when compiling zaptel:
make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686'
CC [M] /usr/src/zaptel/zaptel-1.2.1/zaptel.o
Yes, nice website. But, where's the documentation regarding setup and
configuration? Where's the examples?
-Original Message-
From: Olle E Johansson [mailto:[EMAIL PROTECTED]
Sent: Sat 3/11/2006 1:23 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi,
I successfully connected 2 servers via IAX but I'm pulling my hair to
connect 2 extra servers , Anyone connected 3 or 4 servers together ? is
it possible ?
I d like to share the dialplan so _2 goes to server A _3 goes to serverB _4x goes to server C etc from the 4 servers
any
Is it possible that
putting the incorrect switchtype in zapata.conf can cause Asterisk to
crash?
I have a Definity
Generic 3 connected to a TE405 port 1, and a Nortel BCM connected to port
4. In zapata.conf, I configured the Definity connection (group 1)to
switchtype=national as per
Doug, All,
There is no hidden documentation that I know of, I just kept reading over
and over the existing posts and wiki docs. I'm a little embarrassed but
truthfully, I hacked on this for 10 plus hours trying to understand the
context inclusions between dundi.conf mappings and extension.conf
I had no ideal this thread would get this big! Im going to look more
into the regcontext. I planned on using DUNDi since im going to have
lots of * servers that will be tied over a ATM backbone, so DUNDi can
then do on-net calling much easier and never have to hit the internet,
can just ride our
Matt:
Do you know if it can transcode between H.263 and H.264
Thanks for the info.
Erick W
- Original Message -
From: Matt Riddell [NZ] [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, March 11, 2006 12:55
Really interesting Olle
We are expecting :)
Miguel escribió:
Olle E Johansson wrote:
Asterisk won't be an T.38 endpoint, but will handle T.38 calls
properly, regardless
if the T.38 was offered in the original call setup, or if the caller
suddenly sends a fax
in the middle of a call (a
On Mar 11, 2006, at 6:25 AM, Whisker, Peter wrote:
The G.726 codec is the current Asterisk 1.2 version (revision 7221).
I am using G.711a (alaw) between a Sipura ATA and Asterisk at each end
of the link and am testing alternative codecs on an IAX link (not in
trunk mode) between the two
Hi,
yes, i do believe that T38 on Asterisk is a huge step! I've red about
propriatery solutions with it , i was just wondering when Asterisk
would get it.
This project is just perfect, every day keeping in track with every one needs.
Tthank you for your excellent work Steve Underwood!
Best
Hi all,
i have:
out side PSTNOldPBX-Analog-Asterisk (X100P)
^
|
Local Ext
What is happening is:
Calls from Local ext goes to Asterisk and everything is fine.
Calls from Out
Before everyone from around the world shoots off
their resumes and portfolios to me this must be someone living within Boca Raton.
You will be freelancing on a project to project basis
but the guy you will be working with needs someone local and you will be
meeting with him for each
On Sat, 2006-03-11 at 14:53 -0500, Dean Collins wrote:
Before everyone from around the world shoots off their resumes and
portfolios to me this must be someone living within Boca Raton.
Then why don't you go looking for a mailing list or newsgroup for people
[looking for jobs] in Boca Raton
It's in zapata.conf
is that
transfer=yes
if I set this to no, does this keep the # transfer functionality that is
setup w/AAH?
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Olle E Johansson wrote:
Friends in the Asterisk.org community,
There is a lot of cool stuff going on in Asterisk development, things
that will change Asterisk and
make it work better in your organisation, make it easier to sell in
your area or give you more
consulting oppurtunities - in
OK apart of my beleive that sonicwall is a piece of crap (personal), try
to do a port forwarding for the IAX port (4569)
Saul,
Why do you consider Sonicwalls to be crap? Aside from this odd issue
(which is fixed by using an obscure setting) they've been rock solid for
me, for years.
- Mike
I guess the title says it all. I have a few dozens SIP devices, but I want
to limite devices 10 to 20 to 3 concurrent calls max.
How can I do this with Asterisk without limiting everybody else?
Mick
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Does Unicall support the faxdetect directive? In an Asterisk server that
has both an E1 running Unicall and two lines on a TDM02B, I can only send and
receive faxes on the analog lines. I think the problem may be that Unicall is
not detecting the fax tones and does not disable echo
Does anybody know what devices really support t.38 ? I've seen a few
claiming they do on the box, but most do not seem to support it at all.
Zoa.
Kristian Kielhofner wrote:
Olle E Johansson wrote:
Friends in the Asterisk.org community,
There is a lot of cool stuff going on in Asterisk
*** ITU T.38 -- Fax over VoIP
It's not clear from the bug tracker if the problem with a T.38 endpoint
(say ATA) behind NAT is working yet (with sip.conf specifying
nat=yes/qualify=yes). Is this working or do both T.38 endpoints have to
have public routable IP addresses still?
g.
--
Has anyone else noticed this?
I have LCDial() setup for least-cost routing. It works fine for the most
part. However, when I issue a reload command and then if I try to make a
call, it says it cannot find a provider for the number dials. If I issue a
restart now, everything goes back to
In my remember, the artdio ipf-3000 phone support instant message.
Regards,
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman
Sent: Thursday, February 23, 2006 8:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I'd like to setup an autodial that will do the
following:
1. Firsttry callmy multiple SIP phones
at my house - if no answer withinn seconds, try my mobile phone.
Alternatively call bothSIP phones at home +mobile at the same
time.
2. Once someone answer in step 1 (on any phone)
initiate
I have ordered one (for $71 from the supplier you mentioned, although I
have since found another supplier who appears to have them for $55!!!)
and will run whatever testing I can.
Someone from Cologne has commented that because it us a USB device,
there may be some latency issues (which will
This is not an Asterisk specific question but doesnt
anyone know if you can automatically prepend a 9 on the call lists so clients
can return dial without having to repunch in the number? If you go to
directories now it just shows the number without a 9 (obviously).
Maybe on the
Actually I found the solution when trying one last
time to search for some information to help me out.
Solution is to put in LOCAL/0001 (for example) as
the channel parameter in the call file - and then have this extension registered
in the dial plan with whatever further dial commands
Hi,
I remember reading some people talking about making calls betwen
Asterisk servers and MOH stopping. I was playing with this today and
noticed the same thing. Was there a solution to this?
- Gabe
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Use the callerid rewite rule to prepend a 9 on the asterisk side
-Original Message-
From: Bill Gibbs [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: 3/11/06 6:52 PM
Subject: [Asterisk-Users] Polycom - directory dial
I cannot find any info anywhere about regexten. The Wiki page for it is
404.
- Gabe
- Original Message -
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, March 10, 2006 2:04 PM
Subject:
Doug,
How did get the RTP stream to fail over in progress?
Also, you mentioned your using OpenSER. Why did you choose this over
the standard SER?
- Gabe
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Being USB 1.1 is not a problem - there is more than enough bandwidth for a
BRI in USB. The handsets used in the BRI install are Snom 360's with
firmware 5.3 and internal users have complained of slight echo, however I
believe this is more to do with the Snoms than the Draytek adapters. For
On 3/11/06, Kristian Kielhofner [EMAIL PROTECTED] wrote:
Olle E Johansson wrote:
Friends in the Asterisk.org community,
There is a lot of cool stuff going on in Asterisk development, things
that will change Asterisk and
make it work better in your organisation, make it easier to sell in
Use the group application. SetGroup CheckGroup
On 3/11/06, Michaël Gaudette [EMAIL PROTECTED] wrote:
I guess the title says it all. I have a few dozens SIP devices, but I want
to limite devices 10 to 20 to 3 concurrent calls max.
How can I do this with Asterisk without limiting everybody
Gabriel. We are using OSPF on our asterisk box. When an interface fails, OSPF
switches the default route over to the other interface. :) Fortunately Polycom
phones are smart enough to wait for the RTP stream to be re-established.
As for OpenSER vs SER... I'm not sure. It really shouldn't make
On 3/10/06, Michael Welter [EMAIL PROTECTED] wrote:
My customer with Cortelco phones is *very* unhappy. They expected
Polycom speaker phone quality on a crap $50 phone. The only reason I
didn't have to take the phones back is because the IT guy that ordered
this system quit the company.
My
Doug,
Thanks for this info. Can I bug you for one last question?
From what I can find online, OSPF seems to be a technology or method,
not necessarily a program. What are you using to perform OSPF?
- Gabe
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To:
I don't remember exactly the dialplan needed, but this is something
that you would do in the Dialplan of the Sipura, it should be in the
sipura docs.
On 3/10/06, Anton Krall [EMAIL PROTECTED] wrote:
Guys.
Anybody using sipuras 2002 knows if there is a way to make the phones
connected to it
use the following in queues.conf:
periodic-announce-frequency
periodic-announce
context
Home this helps.
On 3/10/06, Poul Møller Hansen [EMAIL PROTECTED] wrote:
I'm wondering how I can let the caller choose to leave a voicemail
message or continue to wait.
Of course I can leave the queue and
From what I can find online, OSPF seems to be a technology or method,
not necessarily a program. What are you using to perform OSPF?
OSPF is a routing protocol. Quagga (quagga.net) is a good open source
implementation of OSPF for Unix.
David
I am trying to write a demonstration dialplan and I need it to call a
number, connect for X number of seconds and then hang up and continue to the
next priority. However, the S option and the L option for Dial() drop the
call after it hangs up.
This is what I have now:
exten =
Kernel is 2.6. zaptel and ztdummy load with Linux, so I can check in
lsmod that they are loaded. They load without any problem, I've loaded
them manually too.
Zach A
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Friday, March 10, 2006 6:13 AM
To: Asterisk Users
So you are actually able to maintain a call in progress even if the server
its connected to fails (by routing to another)?
- Gabe
- Original Message -
From: David Coulson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Think you need to be careful reading his previous post. OSPF can be used
to fail over to another interface on the same box, but it is _not_
going to fail over to a second box and maintain rtp sessions.
Gabriel Afana wrote:
So you are actually able to maintain a call in progress even if the
That's what we're using. :)
-Original Message-
From: David Coulson [mailto:[EMAIL PROTECTED]
Sent: Sat 3/11/2006 8:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Clustering
No, only if a network interface in the server fails. We have two network
interfaces per system (actually we have four, but two are on a private network
with a MySQL server). If one of the network interfaces fails, OSPF will switch
the default route over to the other interface pretty quick
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