don't thank me it's documented in the app just remembered stumbling on
it in the network tab.
On Sun, 2004-02-29 at 15:46, asdasd wrote:
> sweet, cheers
>
> - Original Message -
> From: "William Suffill" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, February 29, 2004 8:44 P
Hi, Nick
I believe * can connect
PABX through VoIP, but my question is
Whether it can switch calls
between PABXs directly, within the
TE405P,
without conversion to IP. And on the other hand, all PABX
To
be able to make calls to the analogue PBX through VoIP.
All E1 lines are distant
SD
Fran Boon wrote:
Olle's chan_sip2 introduces a 3rd possibility:
Using templates & autocreate peers for the majority of user options &
storing just the passwords in the MYSQL database.
Combining this with MYSQL_FRIENDS, storing template= settings in a database
would be very powerful.
/O
__
On Sun, 2004-02-29 at 09:18, Olle E. Johansson wrote:
> > Olle's chan_sip2 introduces a 3rd possibility:
> > Using templates & autocreate peers for the majority of user options &
> > storing just the passwords in the MYSQL database.
> Combining this with MYSQL_FRIENDS, storing template= settings in
Thanks for the suggestions on the hotel wake-up! Actually, I don't have
a hotel, but my earlier request was unanswered because I suppose it was
uninspiring. So I used a hard example that was readily identifiable.
Your helpful responses led me to the facility I had not managed to find
by mysel
Fran Boon wrote:
I guess I need to implement this with astdb instead of MySQL, since this
can be queried direct within the dialplan.
Would be lovely to have dbget/dbput routines for MySQL as well as just
db1!
Brian was working on odbcget/put. I think there's a beta uploaded on his
web site.
/O
__
I want Asterisk to call my cell phone after someone leaves me a voice mail
message. How do I do this?
I cannot use Dial after the Voicemail application, e.g.,
[Step 1] exten => 100, 1, Dial( SIP/100, 15 )
[Step 2] exten => 100, 2, Voicemail( u100 )
[Step 3] exten => 100, 3, Dial( Zap/g1/CELL_P
> I cannot use Dial after the Voicemail application, e.g.,
>
> [Step 1] exten => 100, 1, Dial( SIP/100, 15 )
> [Step 2] exten => 100, 2, Voicemail( u100 )
> [Step 3] exten => 100, 3, Dial( Zap/g1/CELL_PHONE )
>
> because the caller will hang up after leaving the voice mail in Step 2
> above, and
Hi Bill-
I've built some load testers for asterisk, using the outgoing call facility.
It's been a little while, so you may want to test this yourself, but I
recall finding a couple of problems:
(a) I don't think it manages queuing very well if there are a limited amount
of outbound resources. F
Could someone find the
time to tell me whether ALL functions in Asterisk are programmed using scripts
and contexts ?
What I need to find out
is whether the user CAN configure services for themselves.. Below, I chose
a sampling of a fresh question and answer (just as an example).
In this exam
[Please don't top post]
Nikolay Koev wrote:
Hi, Nick
I believe * can connect PABX through VoIP, but my question is
Whether it can switch calls between PABXs directly, within the
TE405P, without conversion to IP. And on the other hand, all PABX
Yes, as long as that's how your dial plan is set
Hi AR-
(I've converted your message to plain text from HTML, preferred by some of
the members of this group)
In response to your general question, I think the answer is "yes". I think
of asterisk as primarily "call-driven". By this I mean that most actions in
the system are triggered by either
Nicholas Bachmann wrote:
Bill Michaelson wrote:
Anybody know how to implement a hotel wake-up call feature with *?
It seems like it could be accomplished with an AGI and a script that
wrote call files. Have the AGI prompt for the wakeup time (or have a
web interface for a front-desk person d
Hi,
> -Original Message-
> What about putting this in a special context and using 'h'?
>
> i.e.
>
> exten => 100, 1, Dial(SIP/100, 15)
> exten => 100, 2, Voicemail(u100)
> exten => h,1,Dial(Zap/g1/CELL_PHONE)
>
> ? h will get executed on hangup. The only caveat is that if
> no voicem
Thanks, Scott. I'm in a general exploration mode, but I do have a small
broadcast application in mind. My limited experimentation leads me to
suspect that there is no queue management at all. I was testing with
only a single call file just minutes ago, and the system tried to redial
the dest
You can manage when the call starts by setting the atime (or is it
ctime?) of the blah.call file. See my sample script at
www.fnords.org/~eric/asterisk It's the Callback script. You can also
manage it WITHIN the AGI application run by the blah.call script, of
course.
On Sun, 2004-02-29 at 10:49
Hey! For some reason, it looks like two of my channels on my TDM400
stopped working for no good reason.
Asterisk stopped working on my main extension today (it does this
every week or so). I usually remedy this by stopping asterisk,
removing the kernel modules, reinstall them, and restart aste
Its the mtime of the file
James
On Sun, 29 Feb 2004, Eric Wieling wrote:
> You can manage when the call starts by setting the atime (or is it
> ctime?) of the blah.call file. See my sample script at
> www.fnords.org/~eric/asterisk It's the Callback script. You can also
> manage it WITHIN the
Bob Knight wrote:
Nicholas Bachmann wrote:
Bill Michaelson wrote:
Anybody know how to implement a hotel wake-up call feature with *?
It seems like it could be accomplished with an AGI and a script that
wrote call files. Have the AGI prompt for the wakeup time (or have a
web interface for a
On Sun, Feb 29, 2004 at 04:26:17AM +0100, Serge wrote:
> Hello,
>
> Pls. help !
> I have server on Freebsd 5.2 and don't may install asterisk , following errors: (
> gmake clean ; gmake install )
> -
> include/mpool.h:53: error: syntax error be
Hi!
> Could someone find the time to tell me whether ALL functions in Asterisk
> are programmed using scripts and contexts ?
I guess the answer is NO - "applications" are written in C and not
scripted. Still you have verious options for configuration and scripting
as Scott already pointed out.
Hi all,
I would like to have some information about your TE410p and TE405p
cards compatibility with telephony protocols adopted in Brazil.
- When in E1 mode, does it support R2 DIGITAL MFC 5C ?
- When in PRI (Primary Rate ISDN), does it support CSS7 DSS1 ?
- Is asterisk pabx enough to implement th
Hi Alex,
Alex G Robertson wrote:
Hi all,
I would like to have some information about your TE410p and TE405p
cards compatibility with telephony protocols adopted in Brazil.
- When in E1 mode, does it support R2 DIGITAL MFC 5C ?
You need a R2 converter. R2lib is "under construction".
- When in PR
We are going to do this test next week. I will say the result
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jim
Rosenberg
Enviado el: domingo, 29 de febrero de 2004 1:15
Para: Asterisk
Asunto: [Asterisk-Users] PCphoneline FXO to FXS b
You need a R2 converter. R2lib is "under construction".
- When in PRI (Primary Rate ISDN), does it support CSS7 DSS1 ?
- Is asterisk pabx enough to implement these protocols o we need
other drivers/software?
- Do you know if any brazilian telco (Telemar, Vesper, Embratel, GVT,
Telefonica, BRT,
I can't start *. I'm receiving the following error:
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Feb 29 19:01:46 WARNING[1024]: chan_zap.c:673 zt_open: Unable to specify channel 1: No
such device or address
Feb 29 19:01:46 ERROR[1024]: chan_zap.c:5324
Hello All,
I forgot this question in my last post ..
Where is the primary site to R2Lib ?
Best Regards,
Marcio Gomes
Marcio Gomes wrote:
You need a R2 converter. R2lib is "under construction".
- When in PRI (Primary Rate ISDN), does it support CSS7 DSS1 ?
- Is asterisk pabx enough to implem
Marcio,
Marcio Gomes wrote:
You need a R2 converter. R2lib is "under construction".
- When in PRI (Primary Rate ISDN), does it support CSS7 DSS1 ?
- Is asterisk pabx enough to implement these protocols o we need
other drivers/software?
- Do you know if any brazilian telco (Telemar, Vesper, Em
Marcio,
Marcio Gomes wrote:
Hello All,
I forgot this question in my last post ..
Where is the primary site to R2Lib ?
There is no "primary site". A scratch have been release last year but
its developer said to me this source is a "junk". He will release a new
beta version in the future.
Dani
On the wiki pages it suggests that clients on the outside of NAT can connect
to an Asterisk server behind nat. (option no 3). The note suggests that this
can work with port forwarding and some 'header mangling magic'.
I have the port forwarding configured however, when I try to connect an
external
Hi,
Has anyone attempted to convert vmail.cgi to php?
I'll take it in any stage of development.
This, or any other php interface to VM, would be greatly appreciated!
Thx
Ryan
___
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[EMAIL PROTECTED]
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Why not just use the cgi?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan
Courtnage
Sent: Sunday, February 29, 2004 4:16 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] vmail.cgi -> .php?
Hi,
Has anyone attempted to convert vmail.cgi to php?
I'l
Hi,
vmail.cgi isn't working properly for me. My PERL skills are pretty
weak, so I thought I'd check on the existence of a PHP.
FYI - the following is not working in vmail.cgi (for me ... perhaps
there is a single problem causing domino effect):
- can't set audio format in Preferences screen.
Check permissions on the files.
Try changing the group of the files and the vm directory to that of the
user running the script (hence, apache)
See where that takes you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan
Courtnage
Sent: Sunday, Februar
On Sat, Feb 28, 2004 at 11:03:41PM -0600, Robert Lawrence wrote:
> I would be interested in the AGI Script. As for the voice prompts, I am
> having Allison record some stuff for me on Monday, including prompts for
> such a wake up system, that I plan to donate back to the Asterisk
> community.
I haven't figured out yet how to make * wait until the call in answered
before playing a recording (without the recipient pressing #).
Show application dial.
Use option A
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Fugina
Sent: Sunday, February
Daniel Bichara wrote:
Hi Alex,
Alex G Robertson wrote:
Hi all,
I would like to have some information about your TE410p and TE405p
cards compatibility with telephony protocols adopted in Brazil.
- When in E1 mode, does it support R2 DIGITAL MFC 5C ?
You need a R2 converter. R2lib is "under cons
On Sun, Feb 29, 2004 at 06:35:54PM -0500, Matthew B Marlowe wrote:
>
> I haven't figured out yet how to make * wait until the call in answered
> before playing a recording (without the recipient pressing #).
>
> Show application dial.
>
> Use option A
Unfortunately, there doesn't seem to be an
It would be nice if there were timestamps to the left of the messages in the
console...
Anyone know if this is possible?
Kind regards,
Matt Riddell
(CEO - www.sineapps.com)
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That's right, my mistake. Forgot you were using a call file. Sorry :)
Definitely looking forward to previewing your script though!
Good luck.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Fugina
Sent: Sunday, February 29, 2004 6:52 PM
To: [EMAIL
Rob Fugina wrote:
On Sun, Feb 29, 2004 at 06:35:54PM -0500, Matthew B Marlowe wrote:
I haven't figured out yet how to make * wait until the call in answered
before playing a recording (without the recipient pressing #).
Show application dial.
Use option A
Unfortunately, there doesn't seem
I never tried out call files so I just tried one out and when I tried it
as soon as I picked up on my cell phone the audible file started to play
and didn't require a # to start playing.
I wonder why yours operates differently
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PRO
On Sat, Feb 28, 2004 at 11:03:41PM -0600, Robert Lawrence wrote:
> I would be interested in the AGI Script.
Here it is, along with the shell script I run from cron every minute.
It's sloppy -- I didn't expect to be sharing it so quickly.
Rob
--
Rob Fugina, Systems Guy
[EMAIL PROTECTED] -- http:
On Sun, Feb 29, 2004 at 07:10:07PM -0500, Matthew B Marlowe wrote:
> I never tried out call files so I just tried one out and when I tried it
> as soon as I picked up on my cell phone the audible file started to play
> and didn't require a # to start playing.
>
> I wonder why yours operates differ
It should not be too tough to make it wait if you are dialing a channel
that supports answer supervision (like an analog FXS port or a digital
FXO port). Analog FXO ports don't support answer supervision, of
course.
On Sun, 2004-02-29 at 17:52, Rob Fugina wrote:
> On Sun, Feb 29, 2004 at 06:35:54
On Sun, Feb 29, 2004 at 06:58:01PM -0600, Eric Wieling wrote:
> It should not be too tough to make it wait if you are dialing a channel
> that supports answer supervision (like an analog FXS port or a digital
> FXO port). Analog FXO ports don't support answer supervision, of
> course.
In my situa
This will definitely work for a wakeup call, processed from a call file:
;file sample.call
Channel: SIP/1234
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: wakeup
Extension: s
Priority: 1
In extensions.conf:
[wakeup]
exten => s,1,Wait(2)
exten => s,2,Playback(tt-monkeys)
exten => s,3,Hangup
It
Hi
What is the relationship between when CDR recording occurs and the
hangup extension is executed. Normally CDR happens before the h
extension is executed.
I use the h extension to clean up for routines, but sometimes it gets
called to quickly before the CDR is dumped into a DB. I would like
Is there any way to "prove" that an EnumLookup is actually being done?
I've been trying to get ENUM working, and have gotten to the point where
I'm pretty sure the NAPTR records are resolving the way they ought to
be, and "manual" lookups using dig return just what they "ought" to.
But asterisk
Hello,
Does anyone have any information on the zaptel driver under freeBSD? I
know that there has been a 1200$ bounty posted, but wasn't sure if
anyone with any talent has taken up the project. (I don't really have
any talent... :| )
Michael
___
Hi,
Being a long time mangler of RPM based distros with visions of porting my
main numbers to our PRI, I have been working on packaging * for a little
while. Now that they look like they are working I've posted them to my ftp
site. Currently the packages are built on Fedora Core 1 using
CVS-02.29
Hi,
How does one get the Background application to recognize more
than one digit? A similar question to this was asked on the list
on 13 Nov 2003 but that post seems to have gotten no responses.
Basically I have the following in extensions.conf:
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s
asdasd wrote:
> It would be nice if there were timestamps to the left of the messages
> in the console...
>
> Anyone know if this is possible?
>
> Kind regards,
>
> Matt Riddell
> (CEO - www.sineapps.com)
>
Should not be too difficult if everything that writes to the console
uses a wrapper fu
Hi,
I am very new to *. I did searched the list to see whether asterisk has a
way to support ANI for known caller-IDs. I know there is a function called
authenticate. I thought of modifying authenticate to support ANI. Where
could I find the code for authenticate function in the code base ? Has
an
Abraham Lincoln wrote:
> Hey Andrew,
>Hi! Thanks for the info! your sample config works with my iax.conf
> im getting the following response whenever my 2 client windows
> machine adds an account
>
> -- Registered '711' (AUTHENTICATED) at 192.168.1.2:4569
> -- Registered '712' (AUTHENTICATED
I want to pull the current extension into a php script, but can't seem
to figure out the syntax. I've tried:
$agi->agi_exec("GET VARIABLE EXTEN $EXTEN");
$EXTEN = $agi->response_var($EXTEN);
$EXTEN = $agi->request("agi_extension");
$EXTEN = $agi->get_variable('EXTEN');
and some other variations,
The following suggested sequence does not work:
exten => 100, 1, Dial(SIP/100, 15)
exten => 100, 2, Voicemail(u100)
exten => h,1,Dial(Zap/g1/CELL_PHONE)
The Dial command in the 3rd step will fail because the channel it uses to
dial out on has hung up already! For example, if extension 101 leaves
I have this working, with not much work...
SIP CONF
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; address to bind to
externip = ; Address that we're going to
put in SIP messages if we're behind a NAT
localnet = 10.100.254.0 ; Intern
Steve Beaumont wrote:
On the wiki pages it suggests that clients on the outside of NAT can connect
to an Asterisk server behind nat. (option no 3). The note suggests that this
can work with port forwarding and some 'header mangling magic'.
I have the port forwarding configured however, when I try
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