There are a number of ways to do this that I can think of.
1.) By ip address. You could possibly look up the peer in Asterisk
or maybe it's statically assigned.. number of ways.
2.) In each phone's config set the URL to include the identifier for
that phone.
On Thu, 3 Mar 2005 18:53:13 -1000
>Hi
>I get the following error when i
dial a sip extension, please help
>NOTICE[1681]: app_dial.c:746
dial_exec: Unable to create channel of type 'SIP'
> == Everyone
is busy/congested at this time
The SIP extension you are trying to dial
has not registered with
We learnt as we went, and to be honest it went pretty well.
One of our contractor took the job on in order to learn her way around asterisk
and add it to her list of skills.
Later,
PaulH
Melbourne
Australia
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
Hi
I get the following error when i dial a sip
extension, please help
NOTICE[1681]: app_dial.c:746 dial_exec: Unable to
create channel of type 'SIP' == Everyone is busy/congested at this
time
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I do this test :
exten => _[123456789],1,Dial(SIP/${EXTEN},20,L(0))
exten => _[123456789],2,Hangup
when I use 12345 dial 12346 , it should be hangup.
but it don't link I think. why?
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Title: Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question,therethick!
Look. Lets make it
simple.
In most cases, if a guru is bored or not
interested in a noob question they just ignore it. Personally, I find
myself answering some of these specifically because I am not a
- Original Message -
From: "Eric Wieling" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, March 04, 2005 11:45 AM
Subject: Re: [Asterisk-Users] Why ${EXTEN} variable changes after Goto ?
On Thursday 03 March 2005 04:11 pm, Robert Rozman w
Title: RE: [Asterisk-Users] Newbie Question
As someone who started out
using * when I was just slightly educated on Linux, I can say that you are
probably many steps ahead of the average new * user. You have people that
know Linux so that is a major plus. The actual configuration of the
On Thursday 03 March 2005 04:11 pm, Robert Rozman wrote:
> Hi,
>
> I'm trying to implement dynamic routing of incoming calls to local
> extension if previous outgoing call was unanswered.
> But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
> 's-NOANSWER'. I guess this is normal, but I
Hi
I have asterisk running on a server out side the office, this is with real
ip. i have 1 realip to office and we share internet through nat. i have 5
SIP clients registered to asterisk from behind nat.
when one of the sip cleints dial another sip clients extention the call does
not come. when
Does anyone figure out how to send the mac address or line number in the
browser automatically? So each phone would get their own page?
Thanks,
Richard
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Chris HARIGA
> Sent: Wednesday,
Hello, Users.
I loaded module chan_h323.so, chan_vpb.so.
I have met a message : "No one is available to answer at this time".
I don’t know what I do..
My 'h.323 trace 5' result is :
== vpb/1-8: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR]
-- Executing Dial("vpb/1-8", "h323/192
Steven Critchfield wrote:
On Thu, 2005-03-03 at 17:59 -0700, Paul Fielding wrote:
- Original Message -
Look, don't answer lame questions if you don't want to. Flaming a newb
for being a newb is just mean. (they will eventually RTFM or STFW or
they will fail). This is the way of the
Thanks for the quick reply Kevin (and Dean!)...
It's the kind of answer I was hoping for.
The E1 has been used on a Hardware based PBX until a week or so ago, so I
can't see there being an issue there.
We are looking at replacing our existing PBX with an Asterisk Machine and
the E1 live for anot
Callum McGillivray wrote:
Can anyone tell me from experience how long it might take to get it up
and running so that we can make some basic test calls ?
If the hardware is functional and the E1 is provisioned properly, a
decent admin should be able to have Asterisk running and making test
calls
Hi all,
Just a quick question from someone who is reasonably new to
the Asterisk server.
We have ordered the hardware for a test environment, and plan
on setting it up at the start of next week.
At the moment, we have a couple of VOIP handsets, a Digium TE110P
card, an E1 line a
On Thu, 2005-03-03 at 17:09 -0800, Jeff Busch wrote:
> As someone who is new to Asterisk and Linux (I guess I am a newbie), but
> who has been doing a ton of research, Google searches, and is getting to
> intimately know the wiki, I take offense to Steven Critchfield's
> commentary about newbies.
On Thu, 2005-03-03 at 17:59 -0700, Paul Fielding wrote:
> - Original Message -
> > Look, don't answer lame questions if you don't want to. Flaming a newb
> > for being a newb is just mean. (they will eventually RTFM or STFW or
> > they will fail). This is the way of the open source communi
[EMAIL PROTECTED] (Duane) writes:
> There is now a peering arrangement between e164.org and FreeWorldDialup
> which means any and all subscribers on FWD are now easily able to make
> enum calls by prefixing their call with **164, like wise it's almost as
> simple to make a call to FWD by hitting 8
All-
I am considering an Asterisk implementation in Brazil. Unfortunately,
this presents something of a challenge to plan sitting in Chicago,
USA. I know there is a large section of Brazillian Asterisk users who
actively read this list- so I'd love to pump out a few questions-
note, I'm not nece
The way I did this is to simply build a tree in the database in the
shape of...
4 - 2 -> Sweden
- 4 -> UK
- 9 -> Germany
1 - 8 - 0 - 0 -> US TollFree
- 9 -> Dominican Republic
Etc. I just traverse the tree until I get to a node, and I store the
rate with that as
As someone who is new to Asterisk and Linux (I guess I am a newbie), but
who has been doing a ton of research, Google searches, and is getting to
intimately know the wiki, I take offense to Steven Critchfield's
commentary about newbies.
It is interesting... There seems to be a passion surrounding
I have looked through the archives, and can only find old references to
this problem that appear to be no longer relevant, so I thought I'd ask
again.
I am having a problem with periodic breaks in audio over an IAX trunk.
The interruption only happens in one direction, and (I think) only with
- Original Message -
Look, don't answer lame questions if you don't want to. Flaming a newb
for being a newb is just mean. (they will eventually RTFM or STFW or
they will fail). This is the way of the open source community.
Here Here, I'm with you. I find it a constant source of amazement
Fixed my own issue after a little more work and testing.
Ended up that I was missing the "context=localextensions" line in my
sip.conf for the new extensions I had setup.
Thanks!
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff Busch
Sent: Thur
On Thu, 2005-03-03 at 11:00 -0800, David Shaw wrote:
> Hello All, I have one X100P card for inbound calls. I use two Broadvoice
> SIP accounts for all my outbound calls. I'm unable to place calls using
> BV. Inbound BV calls are ok.
>
> Verbosity is at least 3
> -- Executing Macro("SIP/201-365
If you really want to do this the asterisk list is based off of mailman.
You can learn all about mailman here:
http://list.org/
But really, what are the odds that newbs will know to go there first?
Are you going to moderate it? Someone has to actually answer the
questions you know, if a newb onl
Robert Rozman wrote:
I've updated my Asterisk 3 times with :
cvs checkout -r v1-0 zaptel asterisk asterisk-addons
and then do
cd asterisk
make clean && make && make install
make samples
make progdocs
Try this
cd zaptel
make update; make; make install
cd ../asterisk
make update; make; make instal
First, I apologize if this info has been covered, I tried doing a search
on Google and the wiki and found nothing, but I could be searching for
the wrong info.
Second, I am new to asterisk and linux. My knowledge is Asterisk is
much better than linux, so excuse the ignorance in some areas.
Confi
Two of these in ten minutes?
1. Read about Digium and Asterisk and decide. You found this list. Did
you find the Digium site and the Wiki?
2. Google site:lists.digium.com or look on the Wiki. Hardware
recommendations are covered repeatedly.
3. Again, did you go to Digium? Look at their produ
Well, actually I guess it is help-aah not aah-help but I think you saw
the response by mmiranda which will probably cover the sentiments of
most here. He is correct to that if you are a linux noob, you need to
go get a book and figure out the basics of linux before you embark on
this project. Oth
- Original Message -
From: "Umar Sear" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, March 03, 2005 11:01 PM
Subject: Re: [Asterisk-Users] Why ${EXTEN} variable changes after Goto ?
On Thu, 3 Mar 2005 22:11:23 +0100, Robert Rozman <[E
Thanks a lot for all the suggestions!
Unfortunately, it still gives problems.
Most common error message is "ast_realaudio_callback Failed to write
frame" after "paying the beep". Then it says "User disconnected".
Also, it doesn't react to any extension entered and doesn't do any
forwarding (as
Hi All.
I am beginning a project of Call center and predictive diales, my call
center have 50 operators, I have 50 analog phone line with the company PTT
in my country.
I have the following questions:
1- Can I to work this project with Asterisk?
2- What caracteristic of hardware need for my serv
Dose anybody on the list use the Vovida Load Balancer? If
so how stable is it? How many simultaneous calls can it handle? I am looking
to load balance two local PSTN gateways each with a 4 port 410P T1 card. I
currently have 1 gateway that passes 11 to 12 thousand calls a day and I want
On Thursday 03 March 2005 16:55, Matthew Boehm wrote:
> Care to give an example of where it fails? We've been using it for 6
> months, no problems.
Turkey (90) and North Cyprus (90392). Doubtlessly there are others,
but I'd rather trust my business to a database lookup.
B
PS.
Next time, please
This is probably the worst forum to beg for help or be a noob. There is
a strong sentiment that we should each do as much as possible to help
ourselves before we come to the community for assistance. Not trying to
be mean but you should just know that about this list.
In your case, I am wonderi
> Dear Users,
> I am begging for help.
> I just installed [EMAIL PROTECTED] This went amazingly well, this program is
> made for beginners like me. I do not know a thing of Linux and related
> programs and installation went very easy .
Willy, start with a good book on unix basics, * is not a plu
Has anyone been able to get this phone working with * ?
- Gary
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Dear Users,
I am begging for help.
I just installed [EMAIL PROTECTED] This went amazingly well, this program is
made for beginners like me. I do not know a thing of Linux and related
programs and installation went very easy .
Now I am so far that I can login to the GUI on a remote computer, but no
On Thu, 2005-03-03 at 16:20 -0500, Amit wrote:
> Hi Everyone,
>
>
>
> I am student and I have to study about the source code of Asterisk. I
> have downloaded asterisk and was able to install it on Red Hat Linux.
> My study is to go into the source code of asterisk and see how it
> works, how th
On Thu, 2005-03-03 at 12:23 -0800, Dustin Moore wrote:
> I've been trying to figure out if it's possible to connect
> Asterisk to a parent Inter Tel Axxess system through the
> MGCP protocol. The archives for this list aren't searchable
> and I'm wondering if anyone has a simple answer...
The arch
Dustin Moore wrote:
I've been trying to figure out if it's possible to connect
Asterisk to a parent Inter Tel Axxess system through the
MGCP protocol. The archives for this list aren't searchable
and I'm wondering if anyone has a simple answer...
The short answer: NO. Asterisk does not act as an
Hi,
I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch
(www.cirpack.com).
--<*>-[TE405P]--
ISDN interco is EuroIsdn, ports are configured ccs, hdb3, crc4. Cirpack
is Network, * is Terminal/User.
As I encountered some pb with Sip to Zap transcoding (* to Cirpack way
poor qualit
I have a couple of quick questions that I do not see answered on
the wiki or the Asterisk archives.
I dropped an note to the Dev list and got an "awating moderator"
notice. I realized I had left out some important info, so I cancelled
the message and resubmitted it, but have not gotten another
"
I have been successfully connected (incoming and outgoing)
to FWD for a very long time. A few months ago, I changed
from SIP-based FWD service to IAX2-based, and that went fine
as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable.
I rarely use the service, so othe
Christopher wrote:
I just would like to not have that damn status light flashing all the
time. It hard to explain to people who walk in the server room :)
A small piece of electrical tape works wonders!
Also works well on "check engine" lights!
John Novack
___
Christopher wrote:
It's the server's status light that's flashing, and the lcd display
also reads an error as well (PCI parity error).
Plus, I prefer fixing a hole in the floor rather than covering it with
newspaper :).
Just give it a decorative edge and label it waste disposal. ;)
_
-Original Message-From: Christopher
[mailto:[EMAIL PROTECTED]Subject: Re: [Asterisk-Users]
kernel error with Zaptel cards
I just would like to not have that damn status light flashing all the
time. It hard to explain to people who walk in the server room :)
I know w
I do not see a problem there.
You just have to make sure that Asterisk box is set up with correct DNS server
address.
Rudolf
> Glenn Powers <[EMAIL PROTECTED]> wrote:
>
> Giovanni Powell wrote:
>
> >Can i use a domain name instead of an IP address for externip
> >(sip.conf) Because im using
Hello:
I would like to know if there's a way to request free chanels from remote
asterisk servers ?
My idea is to make an agi returning a dial to inter-asterisk connected servers
when there's not enought chanels on local server, maybe like a ping to all of
them or maybe requesting to a central
On Thu, 3 Mar 2005 22:11:23 +0100, Robert Rozman <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm trying to implement dynamic routing of incoming calls to local extension
> if previous outgoing call was unanswered.
> But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
> 's-NOANSWER'. I guess t
You want him to plug a loopback plug into the front of his Dell server? I
think he means the system status LED on the server, not the port LED on the
Zaptel...
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To:
Sent: Friday, March 04, 2005 8:42 AM
Subject: Re: [Aster
It's the server's status light that's flashing, and the lcd display
also reads an error as well (PCI parity error).
Plus, I prefer fixing a hole in the floor rather than covering it with
newspaper :).
Andrew Kohlsmith wrote:
On March 3, 2005 04:28 pm, Christopher wrote:
I just w
Giovanni Powell wrote:
Can i use a domain name instead of an IP address for externip
(sip.conf) Because im using dynamic dns. Not sure what i'm trying to
achieve as yet but, i want to know if it is possible?
I do it and it seems to work.
-glenn
___
Ast
For anyone using CVS HEAD, if you are using queue member persistence or
agent persistence, your next update will cause the persistence to break.
The storage format for these elements has been changed so that it can be
more easily extended in the future, but this required breaking
compatibility.
On March 3, 2005 04:28 pm, Christopher wrote:
> I just would like to not have that damn status light flashing all the
> time. It hard to explain to people who walk in the server room :)
Plug a damn loopback plug in and be done with it. (TDM4xxP has no flashing
lights, just a green "configured"
${EXTEN} is the extension you are currently at.
exten => s,1,Goto(444) ; ${EXTEN} = s
exten => 444,1,NoOp(bleh) ; ${EXTEN} = 444
exten => someotherextension,1,NoOp() ; $EXTEN = someotherextension
-Matthew
- Original Message -
From: "Robert Rozman" <[EMAIL PROTECTED]>
To: "Aste
I just would like to not have that damn status light flashing all the
time. It hard to explain to people who walk in the server room :)
David Brodbeck wrote:
-Original Message-
From: Christopher [mailto:[EMAIL PROTECTED]]
I see that there is a lot of discu
On Thu, 3 Mar 2005 16:20:54 -0500
"Amit" <[EMAIL PROTECTED]> wrote:
Hi Everyone,
I am student and I have to study about the source code
of Asterisk. I have
downloaded asterisk and was able to install it on Red
Hat Linux. My study is
to go into the source code of asterisk and see how it
works,
Just setting up Asterisk. I'd like to be able to dial
out through VOIP providers and have customers type in
a code in response to a prompt.
So far, I've been able to set things up to make the
call and play the prompt. However, my problem now is
the DTMF tones; they don't register when I call
get
Hi Everyone,
I am student and I have to study about the source code of
Asterisk. I have downloaded asterisk and was able to install it on Red Hat
Linux. My study is to go into the source code of asterisk and see how it works,
how the asterisk works when anybody calls to it or when it ma
David,
Which version of Asterisk do you have running?
Do you have it loaded in the default /etc/asterisk location?
Thanks,
...Paul
(Yes, I read the list - :-) )
[EMAIL PROTECTED] wrote:
Hello,
I have a version of asterisk running on my server for more than 1 year. I
wanna update it to the latest ve
Hi,
I'm trying to implement dynamic routing of incoming calls to local extension
if previous outgoing call was unanswered.
But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to
's-NOANSWER'. I guess this is normal, but I don't understand why ? How to
workaround on this one ?
Thanks in a
I am also using Hosted platform from Amarfone.com for one and a half month now. The platform is ok so far and yes they response quickly , though their implementation time is quite lengthy.
Thanks.
Yahoo! India Matrimony: Find your life partner
online.___
And it would not hurt to just ftp off your confs to another box for
safety.
Or use WinSCP3 if you want that GUI app feel to connect to * from
Windows.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time
Bandit
Sent: Thursday, March 03, 2005 1:48 PM
Hello!
I'm try to implement a fax service using spandsp (0.0.2-pre10) and
NVFaxDetect (since I'm using a SIP channel).
I receive the call from pstn on my SIP/PSTN gateway (welltech 3804).
The fax is detected by NVFaxDetect and than a macro is started.
The welltech use Alaw codec.
The problem is th
> I have a version of asterisk running on my server for more than 1 year. I
> wanna update it to the latest version without over-writing any of the
> config files.
>
> How can I do this?
just don't type "make samples" when you rebuild it
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Hello Giovani,
I'm using Mandrake 9 with rebuild kerner versione 2.4.22 and zaptel 1.0.6
How did you solve the problem??
Thanks
- Original Message -
From: "Giovanni Powell" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, March 03, 2005 7:
Hello,
I have a version of asterisk running on my server for more than 1 year. I
wanna update it to the latest version without over-writing any of the
config files.
How can I do this?
Thanks
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Yes, you can, but asterisk needs to be reloaded (sip reload) when your
ip changes.
Julian J. M.
On Thu, 3 Mar 2005 14:57:15 -0500, Giovanni Powell
<[EMAIL PROTECTED]> wrote:
> Can i use a domain name instead of an IP address for externip
> (sip.conf) Because im using dynamic dns. Not sure what i
I've been trying to figure out if it's possible to connect
Asterisk to a parent Inter Tel Axxess system through the
MGCP protocol. The archives for this list aren't searchable
and I'm wondering if anyone has a simple answer...
Dustin Moore
___
Asterisk-U
is a billing problem
i only have to pay for the calls that begin by 9
I didn't understand the example can you explain it again??
wert
David Boyd <[EMAIL PROTECTED]> wrote:
Why not simply delete the cdr via AGI script (ie delete cdr from tablename where number dialed ) for those call
Stuart Ford wrote:
I'm at my wit's end!
I've spent 2 days now trying to get what I thought was a very simply SIP
+ NAT arrangement working. I've trawled the web and picked brains, but
nothing anyone suggests work.
My setup is very simple. I have a * server in a datacentre, with a
public IP address.
Can i use a domain name instead of an IP address for externip
(sip.conf) Because im using dynamic dns. Not sure what i'm trying to
achieve as yet but, i want to know if it is possible?
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Nice! Ok, Ill give it a try.. But, from the last part of what you said:
"agents" then it mean that somehow the software and the web browser act as a
sip agent of some sort and when a call comes in, it gets dialed into the
phone and the browser agent.. Something like that I guess.
Ill give it a tes
Hello All, I have one X100P card for inbound calls. I use two Broadvoice
SIP accounts for all my outbound calls. I'm unable to place calls using
BV. Inbound BV calls are ok.
Verbosity is at least 3
-- Executing Macro("SIP/201-365c", "dialout-default|XXX") in new
stack
-- Executing Goto
Alex G Robertson wrote:
Ok. I got that. But when I put another PRI from another telco (I use it
for Dial clients) it "syncronizes". The clock differnce appear, but this
Telco doesn't send me any alarms.
Yeah, I could see that working. As another responder already, spans from
multiple telcos are
Hello,
you should download the user manual to see exactly how URL launching
works. It works in any case, no matter what telephone you are using,
because you keep a browser open that launches the requested URL when the
call comes in.
XC-AST is not open source, but is available freely for small
How can I see how many lines I have free to connect with the PSTN?
I am using FXO device. I need to know how many lines I have to connect
PSTN.
I am wanting that my GK and SER cann´t send something if there aren´t
lines PSTN available.
___
I'm at my wit's end!
I've spent 2 days now trying to get what I thought was a very simply SIP
+ NAT arrangement working. I've trawled the web and picked brains, but
nothing anyone suggests work.
My setup is very simple. I have a * server in a datacentre, with a
public IP address. There is no fire
Nice idea, good job
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Are u using zaptel-1.0.6 & what OS? cuz i got the same errors this
morning when trying to modprobe ztdummy.
Im using gentoo + 1.0.6, on redhat9 everything is working ok.
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Yea, Im using fxo ports (x100p) cards... So I guess I can only get the info
as far as the zap card goes, beyond that it is out of my control :(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Jueves, 03 de Marzo de 2005
I see that XC-AST has some nice reports but is there any section that has
some screenshots or something on how it can handle popups on incoming calls
that come thru sip hardphones for example and it would be integrated?
Also, I see a price list on XC-AST but you can download it so, is it open
sour
Sounds awesome.
Hadn't come across these before, lots of interesting possibilities.
Do you have a link to the IAX project?
I found
http://sourceforge.net/mailarchive/forum.php?thread_id=6720059&forum_id=3894
0 which was the most informative. Only a couple of mention on this list.
I take it update
Anton Krall wrote:
When you dialout using zap lines and sip phones, the sip connects to the zap
channel and then dials the number, on the logs its shows sip => zap channel
and when zap picks it up shows as answered but how can you really tell if
the dialed number was answered or busy?
If you are u
Guys.
When you dialout using zap lines and sip phones, the sip connects to the zap
channel and then dials the number, on the logs its shows sip => zap channel
and when zap picks it up shows as answered but how can you really tell if
the dialed number was answered or busy?
Steve Underwood wrote:
Daryl G. Jurbala wrote:
I'm looking for an application that can monitor a channel for voice
input and then proceed on. The closest thing I've found is
BackgroundDetect, which expects DTMF.
Here's what I'm doing:
-Call file generated which calls someone and connects them to a
Hi,
I am using asterisk as a GW between SIP and the PSTN. So SIP INVITEs are
coming from and sent to my SIP proxy.
The SIP proxy asks for authentication of the INVITE requests. So I have
configured a SIP peer that I use for outgoing:
[outgoing]
type=peer
host=myproxy
username=asterisk
secret=som
I am in the process of upgrading a Cisco 7960 to the P0S3-06-3-
00.bin image. It's hanging when it reaches the status message
"upgrading". I have two other phones that successfully upgraded.
When I researched on the internet it appears that there is a
problem with upgrading directly from the image
I think this could help :
http://lists.digium.com/pipermail/asterisk-users/2002-August/004180.html
Dpto. Técnico (Softec). wrote:
Hi everybody,
We are trying to develop a new blacklist application using Perl and AGI, in
order to cut the users that call more than a number of minutes. We want that
Kevin P. Fleming wrote:
Andrew Kohlsmith wrote:
Wow you're the first person I've seen using a combination of T1 and E1
on the same card. I wonder if this has something to do with it (i.e.
a bug in the drivers).
Based on my understanding of these cards, I don't think it's possible
for this to w
Dean,
Have you considered using MRCP rather than inventing a new XML protocol?
The need for ASR is one of the reasons I sometimes end up recommending
a Cisco router running VoiceXML to my customers. I then use an MRCP
server for ASR and TTS.
Alistair Cunningham,
Integrics Ltd,
Telephony, Data
Why not simply delete the cdr via AGI script (ie delete cdr from table
name where number dialed ) for those calls that don't adhere to the
dialed number that you want to capture, or am I missing something ? This
would allow you to remove the cdr at the completion of the call, and
preserve stora
Hello all,
All my extensions are dead and when you try to dial any extension number on
the LCD it shows "Loop detected".
On the Asterisk I don't see any of my extension register. Whole system was
working fine and now it's dead. I ran "sip debug" and I don't see any
request sent by my Snom phones.
Andrew Kohlsmith wrote:
On March 3, 2005 07:28 am, Alex G Robertson wrote:
span=1,0,0,esf,b8zs
span=2,0,0,esf,b8zs
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
Wow you're the first person I've seen using a combination of T1 and E1 on the
same card. I wonder if this has something to do with
Hi everybody,
We are trying to develop a new blacklist
application using Perl and AGI, in order to cut the users that call more than a
number of minutes. We want that this blacklist application will be
transparent for the real application, so the application don't have to do
anything else.
thanks to all
i think yuo are rigth
thanks
wertNir Simionovich <[EMAIL PROTECTED]> wrote:
Hi,I think you are going the wrong way, let asterisk register all the calls,and then simply query accordingly. In example, lets say you use the MySQLCDR backend, after all the CDR's are in the DB, simply run
Very simple, lets take Israel and Palesatnian authority. Israel's country
code is 972, mobile area codes are 2 digits, local area codes are 1 digit,
and there is a special area code, which is a subset of a 1 digit area code,
which is East Jerusalem, which uses the following 9722201 and 9722202 and
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