On March 13, 2005 09:57 am, Nigel Burgess wrote:
[door]
exten = s,1,Dial (SIP31,15)
exten = s,2,Playtones(dtmf)
However the call hangsup before trying to play the DTMF tone.
When a Dial happens, the dialplan stops until the call is
disconnected. See show application dial to see how you can send
Peter Svensson wrote:
On Sun, 13 Mar 2005, Robert Hajime Lanning wrote:
There are SMS sending gateways out there, but they are sending
only, no way to receive. This is fixed in the IM solution by
giving the system an account of its own.
Whatever gave you that idea? Most operators have an
Robert Hajime Lanning wrote:
quote who=C F
Well, as far as I know there is no such service in the USA. Take in
mind that SMS is not so popular in the states, email is, and every
cell phone in the US that I have seen that supports SMS, supports SMS
to email from the phone as well.
um, backwards.
Robert Hajime Lanning wrote:
quote who=Eric Wieling
Robert Hajime Lanning wrote:
um, backwards. E-Mail to SMS. I have not seen the other way
around.
Both Cingular and Verizon supports both.
I have not tried this, nor have I seen any documentation mentioning
it. Do you or anyone else have
Raoul Bönisch wrote:
Hello!
I'd like to Flash() a modem line (BRI) with Asterisk. It is a
passive ISDN-card connected to a hardware PBX. I use ISDN4Linux.
I recognised that unfortunately the Flash() application flashes
Zap devices only. Now I am wondering how I could flash Modem/ttyI0.
The source
Colin Anderson wrote:
I'm going to do a deployment of LocustWorld MeshBoxes in some of our remote
locations. Build 90 comes with Asterisk 1.0, and our plan is to use the
MeshBoxes as a WAP for non-Asterisk uses but also to add a 2nd NIC to deploy
Snom's in the remote location. This works fine (was
Brian McCrary wrote:
Hello,
I'm trying to run an ATA behind a NAT device, and am confused on exactly
what the qualify config option does, other than send NOTIFY packets.
Outbound calls work fine, but inbound calls do not go through. With
qualify=yes and nat=yes, my show sip peers looks like:
Roman Zhovtulya wrote:
Hello,
I wonder if I would have to sacrifice anything if I set NAT=yes for
all sip clients I have, regardless of whether they are behind the NAT or
not.
The idea is to have the setting that works regardless of whether the
user is behind the NAT or not, since I'm not sure
Raoul Bönisch wrote:
* Eric Wieling [EMAIL PROTECTED] [2005-03-14 16:56]:
Raoul Bönisch wrote:
Flash is an analog thing. It does not even apply to ISDN.
So how does the R key on my ISDN-telephone work then?
I suspect it sends an ISDN specific put call on hold or take call
off hold message
Ron Joffe wrote:
Hey folks
I have a new setup with a TDM400P for a pair of analog extensions and a few
SIP phones. We seem to be experiencing a bunch of Crackeling when talking
between the analog and SIP extensions.
Any ideas?
Yes. Check the suggestions given to the other guy that posted this
César Davi Ávila do Nascimento wrote:
Talk about skype is forbidden, but to be impolite is allowed...
Great list!
Skype does not interface with Asterisk in any way whatsoever. You
could just as well have asked if someone knows what RNA sequence 42 in
the turnip genome is for. About as many
[EMAIL PROTECTED] wrote:
Hello,
I upgraded my office from Asterisk 1.0.0 to Asterisk
CVS-HEAD-03/13/05-13:14:04 this weekend, and are now
experiencing some problems accessing voicemail. The web based interface
works fine, in addition to dialing 8500,
which is mapped to:
exten =
Kanuri, Seshu (Company IT) wrote:
I am using Asterisk 1.06 Stable.
When I dial my Mobile Number to check Voice Mail or my Bank Account
Phone Access Number, the IVR System on the other end asks me to enter
*2378 to transfer to an attendant.
But When I press *2378, Asterisk tells me that it cannot
Kanuri, Seshu (Company IT) wrote:
Thanks for the pointers. Here is my Features.conf where I have tried my
best to use Asterisk to give away control. I have enabled ## as the
combination key for Asterisk (in quick succession) to retain control,
but otherwise ignore the key presses.
I don't run
Neil A. Hillard wrote:
Using X-Lite to dial extension 400, I hear it ring and then get answered
and I hear about 0.1 of a second of the on hold music and then silence.
If I use the 'line 1' button to put the call on hold and then take it
off again I hear another 0.1 of a second of the music. This
David Zanetti wrote:
I've been beating my head a bit against the 1.0.6 Debian builds of
Asterisk, using an E100P (E1, single span) board.
In machines I've built in the past (back in 1.0.0 days), config I'm
using and that card and 1.0.0 driver combo worked fine.
ztcfg reports no problems:
Once you run Dial from an AGI script, you lose control of the call via
the AGI script.
Jean-Michel Hiver wrote:
(obviously if you do other magic in your dialplan this needs to be
adjusted. The important part is the 'g' flag to Dial (go on after
hangup), and the NoOp which echos the
Richard J. Sears wrote:
Hey Everyone,
I am using NuFone for 866 inbound service and I am trying to figure out
the callerid part of it. Any call into my * system just shows Toll Free
Call and will not give me the calling party's caller ID info.
Is this just something I have to live with using
Jason Becker wrote:
Daniel Burget wrote:
I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines
connected via TE405P. Everything works great, except MOH. I added an
exten with MusicOnHold(30), and it plays just fine. Conferences have
music when no one is in. I have SIP phones. When
John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten = s,1,SetVar(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,SetGlobalVar(EMERGENCY=1)
exten = s,n,SetVar(SET_EMERG_FLAG=1)
exten =
Thomas Andrews wrote:
If I comment out the following line in zapata.conf I would expect
asterisk to forget the cli information for that channel when I reload:
callerid=Uniden Dead (256) 428-6125
... but it doesn't; I have to restart asterisk for it to take effect.
The funny thing is that the
Anton Krall wrote:
What do you think?
CPU0
0: 16148159 XT-PIC timer
1: 4 XT-PIC keyboard
2: 0 XT-PIC cascade
5: 0 XT-PIC usb-uhci
8: 1 XT-PIC rtc
10: 161351663 XT-PIC usb-uhci,
Asterisk is not a SIP proxy.
Wei Su wrote:
We encouter a situation where we need to use SIP info to convey infomation
for one end point to another endpoint. I use asterisk to do the test and
find asterisk does not forward the SIP info to another endpoint, but act as
UAS and returns a 4xx error
C F wrote:
Now consider this (this works with the cisco 7960, even if you put a
7914 with it, it will still use all 20+ plus buttons this way, if CW
is disabled on the phone):
exten = 123,1,Dial(SIP/${EXTEN},30,tr)
exten = 123,2,Voicemail(u${EXTEN})
exten = 123,3,Playback(goodbye)
exten =
cmisip wrote:
No path to translate from SIP/fwdpulvercom-dd5a(2) to Phone/phone0(1)
I don't know why the above message is printing codec numnbers, rather
than names. *shrug*
show codecs will tell you what codec number are what codec name.
It appears that your Phone/phone0 is using G723.1. Looks
Tyler wrote:
I think you're looking for the 'ChanSpy' application that seems to have
inexplicably vanished from the asterisk CVS..
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20ChanSpy
If anyone has any info on this, let me know as I'm in a similar
situation.
As far as I know
Kevin P. Fleming wrote:
Matt Gibson wrote:
This is what I'm sending from my dhcpd server.
option ntp-servers 10.x.x.x;
option tftp-server-name ftp.x.x.x;
option time-offset -18000;
Keep in mind that using TFTP for a Polycom boot server is sub-optimal,
because you have to rename
Brian McCrary wrote:
Hello,
I'm wondering if anyone has benn able to successfully get g726-16
passthrouhg to work? I am wanting to use this codec instead of g729 as
I'm running out of DSPs using a high complexity codec on the Ciscos. I
would think it would work just as g729 does, which has been
It means the caller hung up in the middle of the voicemail app.
Anton Krall wrote:
So far, nobody has been able to tell us what this error means.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber
Sent: Lunes, 21 de Marzo de 2005 02:54 a.m.
To:
Alessandra Grasso wrote:
My objective is to estimate the performances of *
How much the trancoded can influence the performances?
Thanks,
show translation recalc 30
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
Sys Admin wrote:
I am setting up a new asterisk based call center. I just read:
http://www.voip-info.org/wiki-IAX+versus+SIP
After reading this and other google results for IAX vs SIP is there
any reason why i should use SIP anywhere !!
Because most equipment doesn't support IAX
--
Always do
Tom wrote:
This is what I have suspected all along is that the signaling and timing
constraints on the PRI are such that you basically need asterisk running as a
real-time process. The whole point of the thread (in my mind) is if there is
anyway to cause X to not run as such a real-time process
Hakem Taourchi wrote:
Hello,
Do you confirm there is a way to send information and update it while
the call is ongoing using the caller Id information ?
I strongly doubt this will work on anything except an analog phone. I
also strongly doubt that Asterisk supports this at all.
--
Always do
Daryll Strauss wrote:
Yep, I've seen it and from reading http://www.voxilla.com it's a
pretty common problem.
If you turn on debugging what you'll see is that the Sipura has
mistakenly detected a DTMF code in the audio stream and is relaying it
by repeating the signal (very loudly I might add)
So
Andrejus Stavickis wrote:
Hi,
On the iax2 show registry I only see an entry for my SixTel account,
no livevoip.
This is all I received from them on my account activation:
Example for your dial plan:
exten =
_1NXXNXX,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN})
exten =
Matt Loretitsch wrote:
Looking for some help any way I can. I've been closely following
digium's troubleshooting steps and seem to be okay there. I am
connecting, via PRI, to a Definity system. When I release the board on
the Definity side I get this in Asterisk:
*CLI Apr 7 10:17:23
Matt wrote:
I have a STUN server running on my Asterisk box which seems to work
for most of my SIP clients.. but some of them seem to require NAT=yes
turned on. If I go further and turn QUALIFY=yes to on, is there a
reason I need to keep running a STUN server? If so, what's the
difference?
I
Andy Hamilton wrote:
I imagine that you are using SIP, which has numerous issures with NAT.
Consider using IAX2; one of it's benefits is working with NAT, which I
gather is your problem.
Or he could just read the Wiki and the mailing list archives to see
the simple fixes for a lot of NAT related
snacktime wrote:
So far it seems that the major thing affecting voice quality on my *
box is codec translation. How much cpu is required to translate even
a single channel without getting static like sounds or other obvious
translation issues? I know this probably depends on the codecs
Ronald Wiplinger wrote:
Any idea?
-- SIP Seeding peers from Astdb: '3366' at
[EMAIL PROTECTED]:64440 for 3600
-- Saved useragent Sipcom/ATA2000-1.6.11 for peer 3366
-- SIP Seeding peers from Astdb: '886229421761' at
[EMAIL PROTECTED]:5060 for 3600
-- Saved useragent
Ugur GUNCER wrote:
How can play music when is clients phone ringing in dial progress.
Usually you read the documentation.
At the Asterisk CLI do a show applications to show you what Asterisk
apps are available. Also see musiconhold.conf.sample in the Asterisk
source directory (in the configs
Drew Einhorn wrote:
The ATA generates it's own dialtone, and waits for
the user to dial a number, before sending anything
to the * box. So one of the first examples in the
in the Brief Introduction to Dialplans from
Vol. 1 of the Asterisk Documentation Project.
[incoming]
exten =
izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can handle.
Anybody had his hands on this card or knows some details ?
Please God, if you can hear me, don't let them use a TigerJet chipet.
--
Jim Sturtevant wrote:
I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and
my * server.
My SPA is behind a NAT accessing a server which is also behind a NAT but SIP
and RTP ports are forwarded to it.
My SPA can successfully register. It can call another extension which
Jim Sturtevant wrote:
Thank you for your reply. There is a wealth of information on the wiki,
etc. I turned on RTP debug and the SPA is not sending it's public IP it is
sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere...
nat=yes makes Asterisk use the public IP that is
of the USA and want to relocate to Europe.
Eric Wieling
[EMAIL PROTECTED]
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo
Brian McSpadden wrote:
On Apr 9, 2005 5:03 PM, Bellows, Jared [EMAIL PROTECTED] wrote:
I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they charge incoming calls minutes as well? Is there the $0.02 connection fee for the incoming call as well?
That's the only thing they do
Andrew Kohlsmith wrote:
On April 9, 2005 02:13 pm, Eric Wieling wrote:
izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can
handle. Anybody had his hands on this card or knows some details ?
Please
Andrew Kohlsmith wrote:
On April 9, 2005 08:25 pm, Eric Wieling wrote:
Which specific Digium card does not use the TigerJet chip (as shown in
lspci)?
TE405P:
05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev
01)
I imagine the TE410 and TE110 are both also similarly
snacktime wrote:
On Apr 10, 2005 5:28 PM, Paul [EMAIL PROTECTED] wrote:
I have a Sipura SPA-841. I have configured my SIP.conf and extensions.conf
to allow the phone to dial an outside number via my Zap/2 PSTN. When I pick
up the handset I get a dialtone, however, when I press 9, the dialtone
Rich Adamson wrote:
On incoming SIP calls, the caller just gets silence instead of ringing
until * answers the channel. Is this a configuration issue on my
end?
Chris
Correction, this is true for both IAX and SIP incoming calls on my
system. I have IAX setup with teliax and SIP with livevoip.
Tony Hoyle wrote:
Eric Wieling wrote:
You configure the dialplan for your SIP device ON THE SIP DEVICE. DISA
is an ugly hack and should only be used to provide dialtone to devices
The OP's question is not answered by modifying the dialplan. He
specifically wanted to get a dialtone after
Paul wrote:
I have a Sipura SPA-841. I have configured my SIP.conf and extensions.conf
to allow the phone to dial an outside number via my Zap/2 PSTN. When I pick
up the handset I get a dialtone, however, when I press 9, the dialtone
stops. I assumed it would pause for a moment and give me another
John Breeden wrote:
Been there, done that - no joy :-)
It appears the modifier only excepts a numeric, anyone know if/how you
can feed it adecimal/hex for ascii #?
Rich Adamson wrote:
Is there anyway to append the '#' symbol to a dial string? -
hex/octal whatever? I'm surprised that I can't
[EMAIL PROTECTED] wrote:
Hi,
How can i implement VAD/DTX using zaptel with asterisk towards PSTN.
TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not
even a valid idea.
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David Masure wrote:
Hi,
I want to use the cdr to record the call log to my Microsoft SQL Server
using unixodbc and freetds
but when I compile, I've got this message
Does anyone have the same problem and/or know how to solve it ?
Update of /usr/cvsroot/asterisk/doc
In directory
You cannot disable call waiting on the polycoms. Therefore you need
to use SetGroup and CheckGroup to keep Asterisk from sending more than
one call to the same SIP peer at the same time. The polycom will
ALWAYS accept a second call on a line that's in use.
Wiley Siler wrote:
If you have two
tos=0xb8 will set the the packet to be DSCP EF (Cisco likes to use DSCP)
Rich Adamson wrote:
Does anyone know how setting the TOS bits in iax.conf corresponds to
the Cisco TOS types?
For example, if I set:
tos=0x04
in iax.conf, and on the Cisco, I use:
access-list 110 permit ip any any tos 4
I
Xu Wang wrote:
Hello
Our Asterisk works fine with 'real' IP. But when we change the domain to a
virtual IP, the audio stream probably goes to the 'real' IP. There is no
sound coming back. Asterisk log shows that it does not hang up.
Do you know what might be wrong?
Did you look at rtp.conf?
--
Underwood
Sent: Tuesday, April 12, 2005 9:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel cards
Steve Kann wrote:
Eric Wieling wrote:
[EMAIL PROTECTED] wrote:
Hi,
How can i implement VAD/DTX using zaptel
Digium support suggested today that I run CVS-HEAD zaptel with 1.0.x
CVS Asterisk. This seems totally wrong to me. Can others confirm?
--Eric
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
Asterisk-Users
Aaron Mathews wrote:
I'm having a problem with a new digium te110p card. I'm running it on a T1
with PRI signalling, and everything works fine *except* I get errors every
few minutes that look like the following:
Apr 11 23:23:04 WARNING[10251]: chan_zap.c:5993 zt_pri_error: PRI: Read on
40 failed:
Has anyone written up pretty voicemail user docs? I think voicemail
is so easy even my cat can use it. However, my users are complaining
about lack of docs for voicemail.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
No.
parijat wrote:
Pls could u be more elaborate as I am new to asterisk..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday, April 13, 2005 7:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Wiley Siler wrote:
As far as I can see, never gonna happen with an ATA.
ATA is your end point and has no exploitable features like that.
It just connects your analog phone to a digital network.
Meetme or Conference are probably your only bet in that case...
Mystery Glitch wrote:
In my [incoming] context I have something like this:
exten = 8885861575,1,Macro(vrforward,${EXTEN},8136361451)
And thie Macro contains this:
[macro-vrforward]
exten = s,1,GotoIF($[${CALLERIDNUM} = 954555]?40:2)
exten = s,2,SetGroup(${ARG1})
exten = s,3,CheckGroup(3)
exten
trixter http://www.0xdecafbad.com wrote:
I have done some further research, the first RTP packet is sent when
playback() is called. No others. The application is running, if I
press a key and goto a different item that would cause a new
playback()/background() 1 more RTP packet is sent.
To be
This message is to announce a bounty for the following:
If ztdummy is already loaded, generate an error to the console and
syslog when modules for Digium cards are loaded.
If a modules for a Digium card are already loaded, generate an error
to the console and syslog when ztdummy is loaded.
You
This is a bounty for a patch to app_hangup.c to generate an error when
Hangup is called from exten = h.
You should not call Hangup from exten = h.
The bounty is US$10 and will be paid via Paypal. The patch must be
accepted into CVS-HEAD before the bounty will be paid.
--Eric
--
Always do
Andrew Kohlsmith wrote:
On April 14, 2005 08:31 am, Eric Wieling wrote:
This is a bounty for a patch to app_hangup.c to generate an error when
Hangup is called from exten = h.
You should not call Hangup from exten = h.
I disagree; you should use Hangup() WHEREEVER you want to make absolutely sure
Andrew Kohlsmith wrote:
On April 14, 2005 09:42 am, Eric Wieling wrote:
exten = h will not be called unless the channel has ALREADY hung up.
I understand that, which is why I'm still suggesting a WARNING and not an
error.
Something like No need to execute Hangup from the h exten, line
No. Dial(Zap/1/) will dial out ONLY on channel 1 of the T-1.
Tim Connolly wrote:
Could this be caused by using dial commands like dial(ZAP/1/) instead
of using ZAP/g1/x I assumed if you have only one T1, the Zap/1 and
Zap/g1 were the same. Is this correct?
_
From:
I'm able to call park just fine, I can pick up a call just fine. but
if nobody picks up the call and Asterisk tries to send the call back
to te extension that parks it, it fails.
HELP!
001 -- Executing NoOp(SIP/0004f201e463-a-7650, EXTEN=3599
CONTEXT=toll-access) in new stack
002 --
Kanuri, Seshu (Company IT) wrote:
Does anyone know how Polycom 500s will be able to update their time.
My setup for a time sync with Public domain Time servers is not
successful.
We set the NTP server and timezone using ISC DHCPd.
option ntp-servers 172.16.7.1;
option
Adam Robins wrote:
When an outside callers hits my system, I play them a welcome message
and ask that they enter an extension. If the extension is invalid, it
tells them so, and asks them to try again. The relevant logic for this
is:
[extensions]
exten = _2XXX,Dial(SIP/${EXTEN})
;
exten =
Michael Crozier wrote:
On Monday 11 April 2005 12:04 pm, Michael Crozier wrote:
The zaptel drivers are proving quite unstable with this combination. If
I attempt to rmmod the zap drivers, the machine hangs and is unresponsive
to keyboard input, ping, or sysreq. Additionally, I attempted to
Rich Adamson wrote:
My specific issue has to do with ringing on my FXS ports.
A Northen Telecom Harmony phone (circa 1983) rings normally but when I connect my newer GE
2.4GHz cordless I never get more than 1/2 ring (it lights up and works fine... just can't get a
ring from it). Normally I'd
David Wilson wrote:
Hi guys,
How are you keeping ?
I have an analogue phone plugged into a Digium FXS Zap module on my TDM card.
The phone works well except that I cannot seem to transfer calls using the
flash key. I don't seem to get another dialtone as indicated in:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger
Sent: Friday, April 15, 2005 5:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] *8 nor *8# works for me!
I have put into each phone
Rich Adamson wrote:
My specific issue has to do with ringing on my FXS ports.
A Northen Telecom Harmony phone (circa 1983) rings normally but when I connect my newer GE
2.4GHz cordless I never get more than 1/2 ring (it lights up and works fine... just can't
get a
ring from it). Normally I'd
Andre Normandin wrote:
The same thing happened to me a few days ago.. Truthfully, I thought it was
just me, and a coincidence.. My DSL line went down, and astertisk refused
to work until it came back up...
I couldn't even dial out, nor would it receive calls on my 3 analog (X101P
card) lines
Gilad Ben-Yossef wrote:
Samudra E. Haque wrote:
hello, using Asterisk, is there any clever way to provide answer
supervision based upon the received audio only from the FXO interface
(from a public PSTN switch that does not have battery reversal, or CPC).
In zapata.conf use either
Mark Halverson wrote:
My local Telco uses B8ZSESF and does support PBX customizing ANIs on a per
call basis.
What I need to know is, can I use the SetCallerID command in extensions.conf
to transmit the DID# of the extension making the call with the TE410P or is
there a different one that does
Adi Linden wrote:
I can do the dial command like this to give me a 20 second timeout
exten = _9737,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20)
But this also means that after 20 seconds of ringing it goes on the next
dialpeer. I would like to be able to set the timeout Asterisk wait to
There was a bug with codecs for a very long time with Asterisk. In
[general] remove the bandwidth= line (all it does is allow specific
codecs) and disallow=all and allow= for eac codec you want.
Joseph wrote:
When I try to call iaxtel it goes to codec ADPCM even though I have
define in
Howard Lowndes wrote:
Will Wait(n) still listen for DTMF input from the caller after there has
been a Background(some-message) prompt, or do I need to use
Background(silence/n) to still listen for DTMF?
The WaitExten and Read applications won't work for you?
Michael Greb wrote:
On Mon, Jan 17, 2005 at 09:52:45PM -0700, Joseph wrote:
exten = s,1,Authenticate(X)
exten = s,2,DISA,no-password|local
Can someone explain to me what passcode is used for?
If I enter no-password I can make a call but if I enter any number
instead of word passcode it will
Olson, Dana wrote:
Are there any cards that work with * that do the VoIP-to-TDM processing
on the cards, with multiple interfaces?
The QuickNet Internet LineJack meets the description I believe, but it
only has a single FXS or FXO. Are there any cards that have more than
one FXS?
It's been
Andrew Kohlsmith wrote:
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
exten = 1,Dial(Zap/1)
should not generate ringback...
exten = 1,Dial(Zap/1,,r)
Michael Baird wrote:
It's only one guy who seems to attack each poster for not posting in a
manner of which he approves (there is one/two of these fellows on every
mailing list), don't let him ruin your day, this list is quite helpful
and many guys will give you a good answer without the extra
Olson, Dana wrote:
Actually, I do care, and I did search Google (albeit quickly) and I did look on
the hardware list as well as the VoIP wiki. Maybe one of the cards listed there
does what I need, but it wasn't listed like the QuickNet cards are. I thought
perhaps the feature list on the site
[EMAIL PROTECTED] wrote:
Hi, i cant make a three way call using grandstream phones (BT-100) and
asterisk using sip, is this supported or i need a zap interface?
The BT101 cannot to supervised transfers or 3-way calling.
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Paul Rodan wrote:
The BT100's do support conferencing, most SIP phones do. But how does your
Asterisk connect you to the PSTN? Through a Zap interface? If so, what kind;
or through a VoIP provider like BroadVoice, NuFone, LookieLoo, VoipJet,
VoicePulse?
You basically need to make sure your
MJ wrote:
I'm having problems using the following.
[sip]
exten = _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM})
[macro-test]
exten = s,1,Answer
exten = s,3,Flash
exten = s,3,Dial(SIP/${ARG2},30,t)
exten = s,4,Dial(SIP/${ARG1},30,t)
exten = s,t,Hangup
exten = s,i,Hangup
exten = s,h,Hangup
You
Kenneth Long wrote:
You really do not want to run Asterisk and X-Windows
on the same box.
That I understand... but this is not a production
machine. Loading is not an issue. I'm using icewm.
are there any other issues, besides loading, to not
run
x-windows at the same time?
Actually the issue
David Shaw wrote:
exten = 510,1,Dial(SIP/510,20)
exten = 510,2,Voicemail,510
exten = 8500,1,VoicemailMain
exten = _NXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _NXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
Me wrote:
Can someone give me a clue as to why I keep hearing DTMF type beeps on
my phone calls. It sounds exactly like someone on the other end is
pushing a key on their phone but they are not!
Has anyone ever heard of this before? It use to happen once in a while,
today it's been happening a
Paul Rodan wrote:
Using FireFly, all other codecs but G711 Ulaw is selected. But whenever
I place a call, I get:
Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband
DTMF is not supported on codec G.711 u-law. Use RFC2833
Umm, wtf? I thought Inband was ONLY supported
Mark Phillips wrote:
Getting nowhere with Digium support. Trying to tell me that their
engineers are working on it and that it could be months.
Ask if you can ship them the box so they can actually reproduce the
problem.
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Asterisk-Users mailing list
Daniel Nyström wrote:
Is it possible to turn off DTMF recognition (and all transfer services etc.)
pending on CallerID (or FXS channel)?
Some of the FXS channels I will setup soon, is going to work exactly like POTS.
It will be used by people not knowing their within Asterisk.
They will be pretty
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