In /etc/asterisk/zapata.conf change
signalling=em_w
to
signalling=featd
and restart asterisk
On Sat, 2004-07-03 at 17:56, Robert Jackson wrote:
I am trying to receive both CID and DNIS from the telco through a
non-pri T1. Currently I have the T1 setup and operational both outbound
and
add localnet=x.x.x.x/mask
in the [general] section.
where x.x.x.x/mask is the internal network mask
of your * box.
Matteo.
Il dom, 2004-07-04 alle 05:33, Damian Minkov ha scritto:
I have the following situation.
My Asterisk Box is behind firewall ( for example 10.1.1.2 ) I have
mapped
Hello everybody,
I am trying to setup asterisk to redirect international calls via a carrier
which uses a fixed price tel number. The scenario is
Dial 087..something (UK number)
Pause for answer at the other end
Send required telephone number 003..etc followed by #
What is the easiest way of
Mike Benoit wrote:
I'm curious to know if anyone else using SPA-2000's have the same
issues. I wonder if when calls are made from SPA-2000's to PSTN numbers
through Asterisk, asterisk is just amplifying the SPA-2000's own echo
somehow.
I've noticed that my SPA-2000 has very bad echoing at the
Have you tried 'D' option in dial.
exten = _395X.,1,Dial(SIP/[EMAIL PROTECTED],D(wwPINNOww${EXTEN:3}))
-Kannaiyan
- Original Message -
From: Vassilis Konstantinou [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 04, 2004 9:31 AM
Subject: [Asterisk-Users] Using call
I have already read explanation about that in some places but I don't have
still a clear image about the meaning of Penalty parameter inside of
queues.conf
What means that?
Thanks,
Isamar
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Took me some time to get around to check this.
Anyway for the benifit of everyone else. It worked after implementing your
suggesstion. Thanks for your help.
Umar.
p.s I will update the wiki with this information.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi,
I am trying to implement a dialplan in which the user is notified of a
missed call, if no voicemail is left.
Basically what I would like to achieve is something like ...
exten = _0207XXX,1,DIAL(SIP/${EXTEN},15)
exten = _0207XXX,2,HasNewVoicemail(${EXTEN:[EMAIL
Sorry, I've been on voip-info.org but I still can't get a clear definition
of what IAX trunking is. It says you need the timing from a zaptel device
(or ztdummy or zaprtc) to make it work, but nothing specific about what it
is or what it does. Maybe I'm looking in the wrong place.
Right now, I
I have a Cisco 7960 phone, I just updated to P0S3-07-1-00 image/firmware,
due to a ton of fixes from P0S3-06-3-00 which we were running. But now when
I call my phone using X-Lite, the second I answer, it reboots. I tried
upgrading to the latest X-Lite but nothing. So I then tried FireFly, and the
On 04/07/2004, at 6:32 PM, Trevor Peirce wrote:
Mike Benoit wrote:
I'm curious to know if anyone else using SPA-2000's have the same
issues. I wonder if when calls are made from SPA-2000's to PSTN
numbers
through Asterisk, asterisk is just amplifying the SPA-2000's own echo
somehow.
I've
On 04/07/2004, at 11:24 PM, Deon Rodden wrote:
Sorry, I've been on voip-info.org but I still can't get a clear
definition
of what IAX trunking is. It says you need the timing from a zaptel
device
(or ztdummy or zaprtc) to make it work, but nothing specific about
what it
is or what it does.
On 04/07/2004, at 11:26 PM, Deon Rodden wrote:
I have a Cisco 7960 phone, I just updated to P0S3-07-1-00
image/firmware,
due to a ton of fixes from P0S3-06-3-00 which we were running. But now
when
I call my phone using X-Lite, the second I answer, it reboots. I tried
upgrading to the latest
On Sun, 2004-07-04 at 08:24, Deon Rodden wrote:
Sorry, I've been on voip-info.org but I still can't get a clear definition
of what IAX trunking is. It says you need the timing from a zaptel device
(or ztdummy or zaprtc) to make it work, but nothing specific about what it
is or what it does.
Is this done automatically when using IAX2?
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 04, 2004 9:37 AM
Subject: Re: [Asterisk-Users] What is IAX Trunking?
On Sun, 2004-07-04 at 08:24, Deon Rodden wrote:
Sorry, I've been
I have a Cisco 7960 phone, I just updated to P0S3-07-1-00 image/firmware,
due to a ton of fixes from P0S3-06-3-00 which we were running. But now when
I call my phone using X-Lite, the second I answer, it reboots. I tried
upgrading to the latest X-Lite but nothing. So I then tried FireFly, and
Hi,
Can someone tell me how to register and enter in irc.freenode.net chat?
Thank You for your time,
Chris HARIGA
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Just to add some info to this, Hope it will help
I had a similar problem when first testing my * setup.
I was testing it with an active dual line phone line
(all four wires active) and for some reason the X100P
did not like that at all.
Easiest way is to make sure your line from the jack is
just
Is this done automatically when using IAX2?
You need to specify trunk=yes in the IAX config file.
Andrew
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It wasn't a corrupted load, tried this on 5 different phones. For whatever
reason, it's because I had canreinvite=yes on, and nat=no
The phones are on a 10.0.10.0/24 network and my workstation is on the
10.0.0.0/24 network. There is a firewall device linking the 2 subnets.
Either canreinvite=yes
If you are in windows
http://www.mirc.com/
if you are in linux, I use
http://www.xchat.org/
in fact it has a windows version too
all you need to do is download it and install it
once it's up and running all you need to do to enter a room is
/join #[room name]
On Sun, 2004-07-04 at 11:04
On Friday 02 July 2004 21:25, Soren Rathje wrote:
This is what keeps my (CVS-HEAD) server happy..
bindaddr = 192.168.0.200 ; Local interface
externip = 80.63.xxx.xxx ; Public IP address
localnet = 192.168.0.0/255.255.255.0 ; Local LAN, internal clients etc.
When did U join chat community last time???
Take a look... :)
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Mail
Sent: Sunday, July 04, 2004 11:43 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IRC
If you are in
Thanks,
I read that article. Seems like Trunking is a good thing, you said it really
helps at 4+ calls. Does anybody have any reasons for not using Trunking?
Any disadvantages?
- Original Message -
From: Andrew Yager [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 04, 2004
I have a Cisco 7960 phone, I just updated to P0S3-07-1-00
image/firmware, due to a ton of fixes from P0S3-06-3-00 which
we were running. But now when I call my phone using X-Lite,
the second I answer, it reboots.
last eve, i was using
xlite in australia - asterisk in states - 7960-7.1 in
From the archive on June 18:
Date: Fri, 18 Jun 2004 19:23:57 -0500
From: Brian K. West [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] #asterisk is +r now, meaning register your nick with nickserv
How do you register?
do this
You can /msg nickserv register password
You will need to /msg nickserv identify password
Before you can join #asterisk
Bkw
Ps this was covered in the mailing list archives not more than 2 weeks ago.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]
On Sunday 04 July 2004 12:08, Deon Rodden wrote:
I read that article. Seems like Trunking is a good thing, you said it
really helps at 4+ calls. Does anybody have any reasons for not using
Trunking? Any disadvantages?
IAX2 jitter buffer doesn't seem to like it.
I'm a bit on the pissed off side right now.
If you call Free World Dialup conf bridge.. 514 is the FWD number.
Ok last night I was trying to talk to a few people in the conf bridge
because they msged me saying they were there. Ok I call it says Please
Hold then says All circuits are busy... Ok
I'm a bit on the pissed off side right now.
If you call Free World Dialup conf bridge.. 514 is the FWD number.
Ok last night I was trying to talk to a few people in the conf bridge
because they msged me saying they were there. Ok I call it says Please
Hold then says All circuits are busy... Ok
brian wrote:
You can /msg nickserv register password
You will need to /msg nickserv identify password
Before you can join #asterisk
For the lamers: don't use password as your password, use something
semi original...mkay
Jeremy McNamara
___
This looks like a job for AGI...
I'd do something like
exten = _0207XXX,1,Dial(SIP/$EXTEN},15)
exten = _0207XXX,2AGI('missed-call-email.agi')
exten = _0207XXX,3,Voicemail(u${EXTEN:4})
exten = _0207XXX,4,Hangup
exten = _0207XXX,102,AGI('missed-call-email.agi')
...etc...
On Jul
Hai there,
Is it possible to save all the configuration of the
asterisk on the pgsql database? or just the cdr record?
best regards,
Freddy Setiawan
~SimpleWare Solusion~
While your frustration is understandable, FWD is a free service.
It's plausible they had a busy night or whatever, and they couldn't handle
that kind of traffic. Although I've never had a problem personally with the
conference rooms, I rarely use them.
If you really needed the conferencing
Hi, I am very interested in VOIP and telephony in general, although
admittedly, I don't know much about the theories and protocols behind it.
Having also an interest in Linux, I was really excited to come upon
Asterisk. I would really like to learn more about Asterisk and VOIP in
general and am
No the final traffic count was 3mbit's flooding my network that's
unacceptable with any service free or pay.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Deon Rodden
Sent: Sunday, July 04, 2004 1:49 PM
To: [EMAIL PROTECTED]
Also on a side note I have many many asterisk servers handy with meetme and
I'm not an idiot. I was calling the FWD conf bridge because some friends
where there and they msged me on AIM and told me to join. I'm just going to
stop using FWD.
bkw
-Original Message-
From: [EMAIL
hello I'm trying to figure out if anyone's accomplished putting someone on
hold with a hardphone that doesn't have a hold button or multiple lines. I'm
thinking transferring the caller to a specific extension or something...is
this possible? Has it been done?
thanks
hank
Hello Hank,
Would using parking not work for this? You can just have the timeout number
set extremely high, or set so it never times out and when you want to return
to the call - just dial the number that asterisk read to you. Read up on
parking, it may be what you want.
- Joshua Colp.
-
Why not try call parking...
http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20parking
Press # to initiate a transfer, then dial your parking extension. The
calling party will hear MOH and you will hear the extension they were
parked on. Simply dial that extension to pick them back
Remco Barende [EMAIL PROTECTED] wrote:
Cool, did you just use the standard ebuilds in portage (although the
'unstable' versions) ror did you build from cvs?
I have just received my hardware and want to build asterisk on a gentoo
box too :)
I built Asterisk using the latest CVS source. I
Shaun Dawson [EMAIL PROTECTED] wrote:
It sounds to me as if there's a problem on the line. If your X100P
was just not answering then the caller should hear a ring tone,
rather than a recorded message. Well, I don't know anything about
your telco, but that's what would happen here if I
I don't really go to the asterisk irc room that must I'm almost always
on in other rooms
-Stephen
On Sun, 2004-07-04 at 11:48 -0400, Chris HARIGA wrote:
When did U join chat community last time???
Take a look... :)
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
Sorry forgot about the register thing I did it awhile ago and forgot
about it
-Stephen
On Sun, 2004-07-04 at 13:13 -0400, Jeremy McNamara wrote:
brian wrote:
You can /msg nickserv register password
You will need to /msg nickserv identify password
Before you can join #asterisk
All
I've suddenly lost incoming audio on my FWD connection. It worked fine
up until Wed when all of the sudden my calls would complete but I
couldn't hear any audio (I could see the status of the call on the CLI
and could see that my call was using bandwidth on the ethernet switch
and router).
Weird problem.
We have 3 PRI's and 1 5 year old Channelized (Channel bank?) T1 (24 lines,
not pri, no caller id support). Incoming calls run into a Cisco, from there
it gets sent to the Main Asterisk server.
Now, when I have it go to an extension, and have |m at the end to play
music during the
I am happy using AGI, however the dialplan does not seem to work. What
should I expect the priority to jump to when the caller hangsup during
voicemail greeting playback.
Thanks
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chad Scott
Sent: 04 July
with an spa-3000, but
spa3k/line1 - asterisk - xten
i.e. a pure voip connection, from the states (spa3k) to
australia (xten), i heard vicious echo from the states
end. the cairns end heard no echo.
going to
spa3k/line1 - asterisk - sipphone.com - australian/pstn
gave no echo. [0]
randy
I was not thinking straigth I guess.
the behaviour is default, when the caller hangs up, the dial plan jumps to
exten h if there is one.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of usedcanon
Sent: 04 July 2004 22:17
To: [EMAIL PROTECTED]
Subject:
I was there as well, and someone came on, an proceeded to play music on
hold, the same 4 songs for HOURS, tying up a channel you could of used.
At 13:03 7/4/2004, you wrote:
I'm a bit on the pissed off side right now.
If you call Free World Dialup conf bridge.. 514 is the FWD number.
Ok last
www.voip-info.org !
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven M.
Sawczyn
Sent: 04 July 2004 19:53
To: Asterisk-Users
Subject: [Asterisk-Users] looking for newbie resources
Hi, I am very interested in VOIP and telephony in general, although
www.voipbox.de, EN lang blog for news etc
On Jul 4, 2004, at 8:53 PM, Steven M. Sawczyn wrote:
Hi, I am very interested in VOIP and telephony in general, although
admittedly, I don't know much about the theories and protocols behind
it.
___
For make outgoing call, i setup 0. However 0 is write in the cdr dst field.
Is there a way to remove it when asterisk send it to cdr_mysql ?
exten = _0X.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
I just want have in cdr dst = ${EXTEN:1}
This don't work :
exten = _0X.,1,SetVar(EXTEN=${EXTEN:1})
exten =
Hi this is just a heads up about an opportunity for commercial
Asterisk experts. I dont know if this even possible but dont see
why not and it is way beyond my capabilities so thought I would pass it out to
the list.
Ive been looking into Microsoft Live Communications
Server over the
On 04/07/2004 at 14:53 Steven M. Sawczyn wrote:
Hi, I am very interested in VOIP and telephony in general, although
admittedly, I don't know much about the theories and protocols behind it.
Having also an interest in Linux, I was really excited to come upon
Asterisk. I would really like to
I have a tdm400p 4 port fxo card which is not reliably creating the dtmf
dialed digits when making a call. I have placed a linemans handset in
monitor mode on the line and can hear that what the system reports it is
dialing is not what the card is actually dialing. This happens about
25-50% of
hello andy is your user guide updated?
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 04, 2004 5:24 PM
Subject: Re: [Asterisk-Users] looking for newbie resources
On 04/07/2004 at 14:53 Steven M. Sawczyn wrote:
Hi, I am very
no,
that mean use with gk.because that isn't bridge by *.
Does anyone knows another way can do that?
mack_jpn
On Thu, 24 Jun 2004 23:10:43 +0900 (JST)
Isamar Maia [EMAIL PROTECTED] wrote:
Nakano San,
Have you tried to make * only to route the connection and
they just talk
Hi all
How can I fix this problem?
Regards,
mack_jpn
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