On Fri, 2003-07-04 at 16:09, BK [address only for mailing lists] wrote:
Dear Steven,
thanks for your reply
On Saturday, July 5, 2003, at 12:14 AM, Steven Critchfield wrote:
On Fri, 2003-07-04 at 07:23, BK [address only for mailing lists] wrote:
So be
careful to not seem like you
On Saturday, July 5, 2003, at 07:54 AM, Richard Smith wrote:
Secondly and the reason for this email is that I'm looking to pipe the
CDR information generated by Asterisk into a billing system... Can this
be done?
Yes, it can be done.
If you want to use the CDR's for billing, you should set the
Hi all,
I want to get the following functionality: define one extension as a virtual
fax machine.
Every fax redirected to that extension to be converted in a picture file
(bmp/jpg/gif or something else) and then attached to an email and send to an
e-mail address.
Are you aware of a linux based
I repost this message as it does not hit the list first time.
Dan
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, July 05, 2003 9:20 AM
Subject: Virtual fax on the Asterisk box
Hi all,
I want to get the following functionality: define one
I dont know if it's really elegant enough, but if you wanted to set up
a modem and mgetty, it should do most of that. It's not so elegant
and it will tie up a port, but it might be the quickest option.
Dan [EMAIL PROTECTED] writes:
Hi all,
I want to get the following functionality: define
Hi all.
Is there a way to collect the digits dialled in asterisk and stored
them in a variable ? I'm setting a submenu for the user to change his
extension dial in treatment from a standard extension to something like
'automatic transfer' and I need to ask for the number where to
basically, you need to do call transfer, right?
why not using pattern matching?
When your user wants to change is def. exten, it will
be dropped in a submenu (ie another context) with pattern matching.
from here you could match a number and store it in a db.
see
Thank's
Mmmm. that example assumes that you are making the change to
call-transfer dialing from your extension, what if you are away, even
away from the office and you want to transfer all the calls to the phone
where you're at ?
But I agree, I think it can be done with pattern
hello
Newbie, i try examples to understand asterisk.
I have a pb with your macro macro-record-cleanup. the progress of the
macro stops if the macro is execute on a hangup. I try many other
configure with exchange of rules but it seems me that there is no
execute after the first (or second)
hello
Newbie, i try examples to understand asterisk.
I have a pb with your macro macro-record-cleanup. the progress of the
macro stops if the macro is execute on a hangup. I try many other
configure with exchange of rules but it seems me that there is no
execute after the first (or second)
Hi
I think * supports database integration i would be thankful if anyone help
me to configure my asterisk box with database support. One more think can i
store Log in database.
Regards
Obaid Amin Syed
_
Add photos to your
On Sat, 2003-07-05 at 05:43, Ing. Angel Gomez Garcia wrote:
Thank's
Mmmm. that example assumes that you are making the change to
call-transfer dialing from your extension, what if you are away, even
away from the office and you want to transfer all the calls to the phone
On Sat, 2003-07-05 at 08:42, God Knows Well wrote:
Hi
I think * supports database integration i would be thankful if anyone help
me to configure my asterisk box with database support. One more think can i
store Log in database.
asterisk supports a couple of things being integrated with
Hello there
Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making
clean opt in pwlib and openh323 and make clean install in Asterisk i get
an Undefined symbol error when I try to start Asterisk. As far as I can
see its when loading the h323 channel driver the error occurs.
Do
Yes, look into the file cdr_mysql under asterisk cvs directory (
/usr/src/asterisk/configs/cdr_mysql.conf.sample )
God Knows Well wrote:
Hi
I think * supports database integration i would be thankful if anyone
help me to configure my asterisk box with database support. One more
think can i
Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making
clean opt in pwlib and openh323 and make clean install in Asterisk i
get
an Undefined symbol error when I try to start Asterisk. As far as I can
RTFM. Use specified versions of pwlib openh323 instead of latest/CVS
Le sam 05/07/2003 à 14:59, Hervé Thibaud a écrit :
I have a pb with your macro macro-record-cleanup.
Sorry, i didn't look at bugs and now i have seen remark 5 in your extensions.conf
I hope update soon
regards
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[EMAIL
Ok, thx
-Original Message-
From: Peter Zeltins [mailto:[EMAIL PROTECTED]
Sent: 5. juli 2003 19:11
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Runtime error: Undefined symbol, have
fetched new CVS and recompiled everything
Yesterday I updated my pwlib, openh323 and Asterisk from
Hi
thanks to everybody who responded to my earlier post. I have looked at
all the material and links provided and tried everything in there, but
it simply won't work for me.
My SIP phones register with Asterisk, but they cannot be called
(everybody is busy at this time) nor can they call
Why not just do:
[local]
... (local rules)
[switchcontext]
switch = IAX2/
[mycontext]
include = switchcontext
include = local
Mark
On Fri, 4 Jul 2003, WipeOut . wrote:
Hi Karl,
Ok that makes sence as to the way its attempting to look up the extensions for an
exact_match and when the
Hi
thanks to everybody who responded to my earlier post. I have looked
at all the material and links provided and tried everything in
there, but it simply won't work for me.
My SIP phones register with Asterisk, but they cannot be called
(everybody is busy at this time) nor can they call
I am trying to make an in/out trunk group comprised of 4 DS0's using
EM Wink signalling. The first four channels of a DS1 on a T100P
are being used for the group. Outbound calls work fine, but inbound
calls fail. The other 20 DS0 channels are used for a PRI. Does the
configuration shown below
First off, caller ID should be in the q.931 packets and not on the B
channels of a PRI. So if the fsk spill is causing problems, go back to
full PRI and turn off callerid from your telco.
One thing I noticed below is you don't have your D channel defined in
zapata.conf.
I think for em you need
Thanks for the info. My 'telco' is an Adtran Atlas that I have management
control of. I broke out the 4 DS0's into a separate trunk group for
testing. I don't see a way to configure the Atlas to not send caller-id
info on outbound PRI channels, but will look further. Eventually, I need
caller-id
I got this today trying to place a call through FWD:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.11.11.1:5060;branch=z9hG4bK230f856c
From: Iain sip:[EMAIL PROTECTED];tag=as6eaa85fb
To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.3701
I didn't used to have any
I already have a web page to do it, I just wanted to add these feature
upon request by a customer, and yes, he wants the data keyed in.
I was looking at AGI command, GET DATA, will try it.
Thank's
Steven Critchfield wrote:
On Sat, 2003-07-05 at 05:43, Ing. Angel Gomez Garcia wrote:
Why wouldn't SIP show channels display lag jitter, it's always 0ms? Is
there a deeper reason for this or this is just something not implemented
yet?
Peter
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[EMAIL PROTECTED]
You cannot do inband DTMF on anything but ulaw/alaw
Jeremy McNamara
Jay Tyndall wrote:
Hi,
I have setup X-Lite to dial into our Asterisk box using SIP.
It connects, and i can dial the extension no. and hear ring tone.
But cannot hear any of the GSM audio.
There is a message that says:
Iain == Iain Stevenson [EMAIL PROTECTED] writes:
Iain I didn't used to have any trouble with FWD and * is registering
Iain with FWD OK. Has FWD changed or * changed in a way that might
Iain cause this error?
Jeff just announce an upgrade to fwd the other day.
One change is that callers have
Hi all. I bought Digium's dev kit and a used IBM PL300 PC to try it out
in. The X100P works fine, but with the TDM400P I get what I can best
describe as 'interrupt noise'... noise whenever I type a key on the
keyboard, or when something accesses the disk drive, uses cpu, etc.
In my other PC it
On Sat, 2003-07-05 at 21:18, Kevin Herzig wrote:
Hi all. I bought Digium's dev kit and a used IBM PL300 PC to try it out
in. The X100P works fine, but with the TDM400P I get what I can best
describe as 'interrupt noise'... noise whenever I type a key on the
keyboard, or when something
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