On Wed, 2003-12-31 at 05:53, Tilghman Lesher wrote:
> What happens when you change the configuration of the GS phone to
> send DTMF via SIP INFO?
I've just checked my voicemail with 1.0.4.30 and get the same multiple
digits problem. sip.conf and GS config are both at info, for me this is
a new pr
Hi,
I have my GS set to in-audio for DTMF and as bellow for my sip.conf: -
[7001] ; SIP Phone
type=friend
insecure=yes
host=dynamic
reinvite=no
canreinvite=no
nat=1
mailbox=7001
dtmfmode=inband
callgroup=1
pickupgroup=1
disallow=all
allow=ulaw
allow=alaw
allow=gsm
I am using 1.0.4.26 and all is
### connect to asterisk manager through telnet
$t = new Net::Telnet (Port => 5038,
Prompt => '/.*[\$%#>] $/',
Output_record_separator => '',);
#$fh = $t->dump_log("./telnet_log.txt"); # uncomment for telnet log
$t->open("$server_ip");
i got error in this line $t->open("$server_ip");
my ip
Hello, all
Sorry to correct you on this Matt, but I am currently doing this with
the Polycom 600 phones. You need the latest version of both the SIP
software and bootrom to do it, (and that stuff ain't easy to get) but it
is workable.
The latest version of software provides for distinctive ring
Greg Boehnlein wrote:
Hello,
I have been retained by a Building Management Company to install a
combined Voice/Data solution for a Tennated Office Space. This space will
rent offices, with telephone and internet service to inviduals or small
groups of individuals. As fate would have it, the se
Hi all,
Let me be the first to wish everyone, especially the Digium team, an
awesome year in 2004..
Later..
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Lance Arbuckle wrote:
I added the playback line to my [macro-stdexten] context but when I dail
an extension I don't get the "please hold while I try that extension"
message. It just dials the extexsion. Do I have a syntax problem
somewhere ?
exten => 8005,1,Macro(stdexten,8005,Zap/2)
exten =>
When using mysql cdrs, are all legs of a call session logged in the cdr
table? i'm building an app that requires billing on both the incoming and
outgoing (3rd-party transfers) legs.
here's a snapshot of my cdr table:
+-+-+-+-++--+---
--
On Tue, 2003-12-30 at 23:47, Greg Boehnlein wrote:
> Hello,
> I have been retained by a Building Management Company to install a
> combined Voice/Data solution for a Tennated Office Space. This space will
> rent offices, with telephone and internet service to inviduals or small
> groups of
Outbound calls are not logged seperatly in the CDR. So the duration is
the "time the hangup extension ends" minus "time someone dials in on *".
Although I think it could be usefull to seperatly log an outgoing
call-session when it was bridged... My .02 euro.
with kind regards,
Steven
On Wed, 200
Yes.
P.S. Someone shoult set this sticky :)
Jorge R. Constenla wrote:
Hi,
Anybody knows if Asterisk work fine with ser ?
We are using SER (iptel) for VoIP and we want to use Asterisk for PSTN
termination for inbound and outbound calls.
Jorge
___
Asteri
Scott,
Thanks a lot ! this is exaclty what I wanted. Both my E1's came up without
problems.
Best regards,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel
Sent: Tuesday, December 30, 2003 3:42 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asteris
do you have the manager interface turned on?
You need to make sure your /etc/asterisk/manager.conf file looks something
like this:
;
; Asterisk Call Management support
;
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
[testuser]
secret = test
;deny=0.0.0.0/0.0.0.0
;permit=192.168.0.1/255.2
Cool, haven't looked that in depth into the new firmware(is that the 2.4.1
firmware?) I'll have to try that.
I'll post your instructions on the Wiki page later today.
Thanks,
MATT---
-Original Message-
From: John Baker [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 3:07 AM
WipeOut wrote:
> Hi all,
>
> Let me be the first to wish everyone, especially the Digium team, an
> awesome year in 2004..
>
> Later..
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
Senad Jordanovic wrote:
WipeOut wrote:
Hi all,
Let me be the first to wish everyone, especially the Digium team, an
awesome year in 2004..
Later..
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-us
WipeOut wrote:
> Senad Jordanovic wrote:
>
>> WipeOut wrote:
>>
>>
>>> Hi all,
>>>
>>> Let me be the first to wish everyone, especially the Digium team,
>>> an awesome year in 2004..
>>>
>>> Later..
>>>
>>> ___
>>> Asterisk-Users mailing list
>>> [
On Tue, 30 Dec 2003, Tilghman Lesher wrote:
> On Tuesday 30 December 2003 22:16, Greg Boehnlein wrote:
> > On Thu, 18 Dec 2003, Aaron Martin wrote:
> > > I have upgraded my grandstream phone from firmware 1.0.3.78 to
> > > 10.0.4.30 and now I am having problems with early dial. On the
> > > older
Greg Boehnlein wrote:
On Tue, 30 Dec 2003, Tilghman Lesher wrote:
On Tuesday 30 December 2003 22:16, Greg Boehnlein wrote:
What happens when you change the configuration of the GS phone to
send DTMF via SIP INFO?
I had that set originally. I get the same behavior no matter wether I u
I've added it as a separate page:
http://www.voip-info.org/wiki-Polycom+auto-answer+config linked from the
Polycom phones page.
Could you possibly send me a quick line or two(example code) on setting the
ALERT_INFO variable in Asterisk?
Thanks,
MATT---
-Original Message-
From: mattf [
When you push your services button, are there any "" slots.
If not, delete the 3rd or 4th ones
Don't know if it matters, but I have:
Comedian Mail
Asterisk PBX
The only "selectable" ones are the Comedian Mail and the Asterisk PBX.
You might try deleting them all and reloading.
And "#" is
Greg,
> I have been retained by a Building Management Company to install a
> combined Voice/Data solution for a Tennated Office Space. This space will
> rent offices, with telephone and internet service to inviduals or small
> Now.. This is our first deployment of Asterisk, and I n
Hi
I'm trying to configur a grandstream BT101 to connect to asterisk, both
behind different NATs, I realise that a double Nat is a problem, I have
tried using fwd forwarding to iaxtel as a solution but cannt seem to
get them to connect as I think there is a codec problem as IAXTEL
doesn't see
Hi,
we have implemented a first version of call support from a web based
system for Asterisk (via the manager interface) and other, callto: and
tel:, providers.
Now I am looking at the other way around. If a call comes in, I want our
web based system to automatically detect the number and pres
Yes, there are 2 available slots. When I push services, I see:
Services
> Asterisk PBX
Asterisk PBX
As I mentioned, the only soft-key I seem to have is 'VMail', and when I
select that I still get:
Comedial Mail
download refused
Services is full
And yet somehow
Justin,
> I have been trying to get my 7960 & 7960G to register with two seperate *
> servers.
>
> Asterisk box 1 is on out on the Internet running: Asterisk CVS-10/30/03-20:08:15
>
> Asterisk box 2 is on the Lan running: Asterisk CVS-12/19/03-19:28:30
>
> 7960 is on the LAN running: P0S3-0
Peer Oliver schmidt wrote:
Hi,
we have implemented a first version of call support from a web based
system for Asterisk (via the manager interface) and other, callto: and
tel:, providers.
Now I am looking at the other way around. If a call comes in, I want
our web based system to automaticall
> And "#" is the Transfer button. 8-)
Any way to map that to a soft button? How do I use # in a call if not?
Regards,
Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Are there any Snom 200 users that have two extns defined on their phone?
I've been trying to get two (or more) extns defined in such a way that
when extn #1 rings, LED #1 flashs; extn #2 rings, LED #2 flashes, etc.
(Answer greating will be different depending upon which extension is
called.)
I c
The cisco 7960 phone works great. Reliable and fully functional. The
even have support, sort of. In a real business environment I couldn't
imagine using anything else right now. The extra money you spend will be
paid back immediately on the service calls you won't have to make. I have a
bun
You may wish to try Java. It is a client OS independent programming
language. With a server side part and a client side part, it might provide
a solution. It also has some nifty event based handling features that make
the client side work well with the server side. In other words, when events
h
Hi!
> Now I am looking at the other way around. If a call comes in, I want our
> web based system to automatically detect the number and present the call
> information to the user.
Look at this for a start:
http://www.voip-info.org/wiki-Asterisk+tips+managing+CID+names
Then code your part and
Dear all,
I read across Asterisk's lists archives, and found out various
discussions about how nice it would be to have a (SQL) database
abstraction layer enabling the use of various SQL backends, for
various purposes inside of Asterisk.
As far as I see, there is no such thing yet, although there
Wow! Thanks John for the detailed information.
This is such an awesome system... and great support here, too.
On Dec 31, 2003, at 12:07 AM, John Baker wrote:
Hello, all
Sorry to correct you on this Matt, but I am currently doing this with
the Polycom 600 phones. You need the latest version of bo
Strange.. I have my 7960 registered with 3 diffrent asterisk servers...
its ROCK SOLID!! version 6 firmware.
bkw
On Wed, 31 Dec 2003, Rich Adamson wrote:
> Justin,
>
> > I have been trying to get my 7960 & 7960G to register with two seperate *
> > servers.
> >
> > Asterisk box 1 is on out on the
Happy New Year Wipeout and all the other Asterisk Users around the globe
from the Shore Linux Solutions Team
Special kudos and a Happy New Year to the Digium Team
P.S. Wipeout in 2004 are you going to do a Fedora Howto like your RH9
Howto of 2003?
AJ
__
> P.S. Wipeout in 2004 are you going to do a Fedora Howto like your RH9
> Howto of 2003?
Why would you need a howto in the first place? I never did! :P
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/aster
Quite frankly, I don't need a howto, I have it running on my Fedora core 1
system as well as my RH 9 system. I just thought it might be a good idea
for newbies or other people not very familiar with Linux or asterisk,
needed packages, dependencies, etc. Personally I think thorough
documentati
Ok, first off, Asterisk is the coolest piece of software I have EVER had
the pleasure of using in my 8 years of running linux !! and I know I
haven't even scratched the surface feature wise.
Before I get too excited, I wanted to get all you experts to look at the
how I implemented my after hours
I've just checked my voicemail with 1.0.4.30 and get the same multiple
digits problem. sip.conf and GS config are both at info, for me this is
a new problem voicemail has always worked perfectly with the GS.
This has come up many times in this list, with no consensus for a
solution. According to
Where can I find that Howto? I'm new to Asterisk and am looking for all the
doc I can find.
TIA,
Eric
On Wednesday 31 December 2003 12:29, [EMAIL PROTECTED] wrote:
> Happy New Year Wipeout and all the other Asterisk Users around the globe
> from the Shore Linux Solutions Team
>
> Special kudos
Hi all!
> Mr. West I've been called arrogant, egotistical, self centered and even an
> asshole but I think i have enough sense to realize that ...
Heat up that flame, turn it into a nice firework and celebrate - 2004 may
be closer than you think! :-))
Cheers, Philipp
__
I wanted to post the beginings of my latest IVR Project for an automated
Time Clock software.
The customer has over 300 Field Reps that call in everytime they arrive
on location and whey they leave that location. This is handled by the
receptionist now and she logs in them and out of there Time C
We needed the client browser to be open all the time for dynamic data to
load without the page refreshing. After looking at all of our options we
decided on programming it ourselves using flash rather than java.
We have a flash frontend thats tied to our backend mysql DB. We use it
for loading we
I seem to need an upgrade to my Audiocode MP108 FXO for improved loop
supervision. Anyone have a copy they are willing to share? Audiocode is
just ignoring my requests for tech support.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.dig
> Where can I find that Howto? I'm new to Asterisk and am looking for all
> the
> doc I can find.
>
> TIA,
>
> Eric
>
Eric,
You will find at at:
http://members.lycos.co.uk/wipe_out/asterisk/
Robert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http:
Have you tried deleting those services and adding them back in by
checking voicemail 1st (dial 8500) then dial your 8600 for AdsiProg
You should be able to just hit the "#" during the call, but you will
also have to make sure you have the |Tt defined in your extensions.conf
file as well.
Tim
I am putting together a solution that will employ the Digium TE410P with
one T1 going out the PSTN and the other front-ending a PBX. The idea is
that based on a URL, Asterisk will dial an employee behind the PBX. When
the employee picks up, Asterisk will dial the customer (detailed in the
URL). I a
Update #3.
Sayson tell me that this is likely a result of Comedian Mail not knowing
anything about the phone's slots or lock codes, causing the download to
fail.
When I was programming the slots originally, I needed to change the
following information in asterisk.:
SECURITY "_AST"
After looking a litter deeper in the code, it looks as though the
Comedian Mail will only load in the 1st slot of your ADSI phone and the
asterisk loads in the 2nd.
Soif you delete the 1st slot that you had listed as Asterisk, then
dial the 8500 extension it will hopefully work.
Tim Thomps
Tim,
Thanks for your continued participation in this thread.
The truth is, it's not clear to me how to delete a service ... the services
menu only allows me to 'Select' or 'Quit'.
It's also not clear to me how you managed to get Comedian Mail downloaded to
Slot 1 without unlocking it with a code
> Mr. West I've been called arrogant, egotistical, self centered and even an
> asshole but I think i have enough sense to realize that not everyone is on
> the same level just because you didn't need a howto guide doesn't mean
> others don't or won't. By the way, two closing points. First of all
Phillip,
Nope I should have been a bit more clear in my response.. but
its one of those days The howto should be generalized... Not just for
X or Y distro... its not helpful and cuses duplication of documentation
efforts.
bkw
On Wed, 31 Dec 2003, Philipp von Klitzing wrote:
> Hi
> I am putting together a solution that will employ the Digium TE410P with
> one T1 going out the PSTN and the other front-ending a PBX. The idea is
> that based on a URL, Asterisk will dial an employee behind the PBX. When
> the employee picks up, Asterisk will dial the customer (detailed in the
>
As a newcomer to Asterisk, you will not be welcomed
with open arms. First, you will find almost no
documentation on it's features. Second, if you try to
ask questions, you will be flamed and pointed to
worthless how-tos and 'the wiki'. These worthless
documents can only be useful for explaining
There are many people on this list that are more than happy to help you with
a problem if you know how to ask the question. But if you've tried to keep
up with this mailing list over any amount of time, you will see how quickly
it becomes frustrating when people ask the same questions over and ove
> As a newcomer to Asterisk, you will not be welcomed
> with open arms. First, you will find almost no
> documentation on it's features. Second, if you try to
> ask questions, you will be flamed and pointed to
> worthless how-tos and 'the wiki'. These worthless
> documents can only be useful for
> Nope I should have been a bit more clear in my
> response.. but its one of those days The howto should be
> generalized... Not just for X or Y distro... its not helpful
> and cuses duplication of documentation efforts.
I would tend to disagree with your statement that it is not
> As a newcomer to Asterisk, you will not be welcomed
> with open arms. First, you will find almost no
> documentation on it's features. Second, if you try to
> ask questions, you will be flamed and pointed to
> worthless how-tos and 'the wiki'. These worthless
> documents can only be useful for
Sure, here's my extension for paging on the intercom:
[ext-intercom-one];
exten => _87XXX,1,SetVar(ALERT_INFO="Ring Answer")
exten => _87XXX,2,Dial(SIP/${EXTEN:1},20,r)
exten => _87XXX,103,Congestion
exten => _87XXX,104,Congestion
exten => t,1,Hangup
Internally, I use a four digit extension here,
If you are a person who likes all things easy, and if you don't need to know
nothing to be better professional, well, run now, and let us continue our
journey. Who cares? People likes you don't help to our community.
Regards,
Gus
- Original Message -
From: "Me" <[EMAIL PROTECTED]>
To: <[
> > With a community so 'anti-n00b', don't expect your
> > problems to be fixed anytime soon.
> >
> > RUN!!! Don't walk... away from Aterisk.
...
> There are many people on this list that are more than happy
> to help you with a problem if you know how to ask the
> question. But if you've tri
Dear newbies,
As a newcomer to woodworking, you will not be welcomed with open arms.
First, you will find no documentation on how to make your completely custom
ceiling-height cabinets perfectly the first time that your wife will
appreciate. Second, if you ask any woodworker for assistance, yo
There are many reliable Asterisk installations, some running hundreds of
phones. It's easy to stop at a CVS version and build a very stable system.
We have many happy customers running Asterisk. I certainly prefer it to my
Cisco Call Manager installations. It's already a much better product than
> Dear newbies,
>
> As a newcomer to woodworking, you will not be welcomed with open arms.
> First, you will find no documentation on how to make your completely
custom
> ceiling-height cabinets perfectly the first time that your wife will
> appreciate. Second, if you ask any woodworker for ass
As a new asterisk user myself, I would agree with you that the learning
curve is steep, but that was my expectation coming into this. I took
the time to browse the list archives before signing up so no surprises
there. There are some real experts here and they obviously help those
who ask inter
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 31 December 2003 03:24 pm, asterisk wrote:
> Here's the deal:
> It does almost anything. I can make it open my garage door. My
> installation records all conversations and then archives them as
> timestamped stereo MP3s. Our VB windows
Hello,
I am not a veteran here, but would like to share my thoughts on this
subject.
True, * is opensource and freely available, but it is not a computer program
that you download and run. It is a very versatile telecommunication product
you would otherwise pay at least 100 K to buy from a teleco
Well said.
- Original Message -
From: "SW" <[EMAIL PROTECTED]>
To: "[EMAIL PROTECTED] Digium. Com" <[EMAIL PROTECTED]>
Sent: Wednesday, December 31, 2003 2:13 PM
Subject: [Asterisk-Users] New to asterisk? RUN... don't walk.
> Hello,
>
> I am not a veteran here, but would like to share m
Me wrote:
As a newcomer to Asterisk, you will not be welcomed
with open arms. First, you will find almost no
documentation on it's features. Second, if you try to
ask questions, you will be flamed and pointed to
worthless how-tos and 'the wiki'. These worthless
documents can only be useful for
First of all regarding my social ettiquette, it has nothing to do with my
lack of it but more to do with my philosophy of chew them up, spit them
out regardless of the venue they choose. I'll play on field, "You lay it,
I'll play it." As far as your thought of a generalized howto, it happens
Also to add a bit to what I said earlier, the reason I thought RedHat /
Fedora was a good howto was the mere fact that RedHat / Fedora has been
known to present it's own distinct installation problems because of
packages / pachage dependencies.
AJ
___
I wanted to give you some guidance on the configuration of the phone
Here is sniplet of configuration Aastra 390 and 480 Phones...
In an ADSI script for the 1st Slot:
;
; Asterisk default ADSI script
;
;
; Begin with the preamble requirements
;
DESCRIPTION "Asterisk PBX" ; Name of
Thanks 'gcc', that's exactly where I'm at now, with the exception of the
helpful comments on how to clear the services - thanks for those. I already
have those codes, and have used them to download (the somewhat
disappointing) 'asterisk.adsi' sample into both slots.
The problem now is that Comedia
I've tried it several times, and your ADSI clearing tip didn't check out on
my phone ... in particular:
Hit options
Hit Mute or Flash
When I hit options, then the mute button (I don't have a flash button) I'm
still left in the options screen.
-d
--
Darren Nickerson
Senior Sales & Support Engine
- Original Message -
From: "Lance Arbuckle" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, December 31, 2003 1:54 PM
Subject: [Asterisk-Users] after hours - is this logic ok ?
>
> Ok, first off, Asterisk is the coolest piece of software I have EVER had
> the pleasure of usin
- Original Message -
From: "Lance Arbuckle" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, December 31, 2003 1:54 PM
Subject: [Asterisk-Users] after hours - is this logic ok ?
>
> Ok, first off, Asterisk is the coolest piece of software I have EVER had
> the pleasure of usin
Andrew Thompson wrote:
>
> - Original Message -
> From: "Lance Arbuckle" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, December 31, 2003 1:54 PM
> Subject: [Asterisk-Users] after hours - is this logic ok ?
>
> >
> > Ok, first off, Asterisk is the coolest piece of soft
On Wed, 31 Dec 2003 21:19:10 +0200
"Stephen Karrington" <[EMAIL PROTECTED]> wrote:
> We needed the client browser to be open all the time for dynamic data to
> load without the page refreshing. After looking at all of our options we
> decided on programming it ourselves using flash rather than ja
I've never had early dial working, however, I resolved my multiple digit
issue by simply putting both the GS phones and asterisk in INFO mode.
This worked on both 10.0.3.81 firmware on the budgetone and the ATA286,
as well as 10.0.4.30 firmware. I'm not saying I don't believe you, but
doubelcheck
Well, since everyone else is top-quoting on this message, so will I :P
I'm no veteran either. As a matter of fact, I have had ZERO prior
knowledge to the telcom industry or more than 'user level' experience
with telecommunications in general. I decided that I wanted to expand
my knowledge, and a
> Steven Critchfield wrote:
>
> > On Tue, 2003-12-30 at 23:47, Greg Boehnlein wrote:
> > 3. I am also responsible for delivering inbound faxes to the DID numbers
> > via Email. I.E. customer has a document faxed to them and they get it in
> > Email as a tiff. I'm considering using Hylfax with a Mul
Hi Darren (and anyone interested in this issue),
Just an FYI that the factory reset procedure that wipes out the programming in these
phones is different for the 390 than the 480 phone. If any of you need this procedure,
or information on ADSI programming or the 390/480 phones in general for th
83 matches
Mail list logo
