Tim Pushor wrote:
I am still unable to initiate a call transfer with the keypresses
defined in features.conf in a couple month old version of asterisk from
CVS HEAD.
Before I go ripping things apart, I was really wondering if this is by
design, or should it work on all my devices? I have an
Kib Eki wrote:
Hi,
what do i have to configure to get a busy tone when dialing out over
ISDN channel with my Polycom 500 IP?
Try priindication = inband in /etc/asterisk/zapata.con
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Jerry wrote:
The digitmap is in your telephone. Used to terminate dialing and send
the dialed string to *.
Grandstream BT phones don't have a digitmap feature.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
Greg Boehnlein wrote:
I've got Polycom and Cisco phones and quite frankly, I prefer the Polycom
Soundpoint IP-500 and 600 to my Cisco's now. All things being equal
between the phones, the following are why I prefer the Polycoms:
1. Better speakerphone than the Cisco 7960s. Despite the fact that
Grandstream does not support a dialplan. It is supposed to support
Early Dial, but didn't work. I've been told that recent firmware
fixes the early dial bug. I doubt that Early Dial is the solution.
The solution is to buy a good IP Phone. Polycom and SIPura both
support continue dialtone
Chris wrote:
You need this before wcfxs
/sbin/modprobe zaptel
*sigh*
zaptel will automatically load when the card driver loads.
modporbe will also run ztcfg after loading the card driver because (if
you ran make install) /etc/modules.conf tells it to do so.
--
Always do right. This will gratify
Ron Wellsted wrote:
I have to agree, the Cisco 7960 is probably the best (I have yet to try
a 7970/71). Cisco are a pain to deal with (they only want to deal with
large value customers/distributors) and the phone do have some small
quirks/bugs but they are the best in functionality and build
Paul wrote:
Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P
cards in the system. One now has it's own interrupt and the other is sharing
one with the soundcard. I tested outbound calls on both cards, still have
the damn static. I am so sick of this. Is anyone else using
Steven Langley wrote:
I tried putting in nat=yes in the sip.conf file, and asterisk then rewrites
the sip message with the IP of the Nat and the external port. It still
works, but only if there is a constant flow of rtp traffic. If there is a
break in the traffic, then the connection is lost.
Pavel Siderov wrote:
Is it possible turn on/off VAD (silence suspression) w/ Asterisk?
Asterisk does not support VAD so it doesn't make sense to be able to
disable it.
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Anton Krall wrote:
Guys.
Ive read on the wiki that a common problem with nat is that you can only
have 1 sip phone behind, how do you get around this issue? Having a sip
enabled router behind the nat like the GS 488 489 or 486? Or how have you
done it without having any kind of linux box (SER or
Rizwan Chaudhry wrote:
Hey
I want to implement billing in Asterisk for a calling card type application.
My scenario is like this: PSTN = Asterisk =(IAX)= Asterisk = PSTN.
I used the ${ANSWEREDTIME} and ${DIALEDTIME} variables but
${ANSWEREDTIME} always gives a value even if the call is not
Damian Funnell wrote:
When I asked them for further information
on how to improve this they replied:
** Extract begins **
SCSI RAID can cause the problem. If disabling hyper threading does not
resolve your problem my next suggest would be to revert to a PATA IDE
hard drive solution configured
kurt x wrote:
I have the following extension (7700) that can dial out with the below config.
exten = _1nxxnxx/7700,1,Dial(SIP/[EMAIL PROTECTED])
exten = _1nxxnxx/7700,2,Hangup
If I change it to
exten = _1nxxnxx/77XX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _1nxxnxx/77XX,2,Hangup
It
Brian Watters wrote:
We have a Dell 1550 server and find that when attempting to start the server
with any one of the four new Digium Wildcard X100P OEM FXO PCI cards the
server will not even power up much less boot, upon removing the PCI card it
will boot no issues, Placing any other PCI card in
So the only thing you have not done is tried the cards in a different
system with a different motherboard. It is WELL KNOWN that the cards
will not work well if they are shareing interrupts with another device.
Ian Pattison wrote:
I don't know how everyone else is doing but my woes are
Matt Schwartz wrote:
Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to
install the MeetMe application. I don't think it installed with the
standard 'make install' command. If not, how do I accomplish this?
MeetMe requires Zaptel. If you do not have Zaptel installed,
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
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
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...
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?
--
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
Steve Mann wrote:
I think it is i you want, s is the start for a context, meaning anything
coming in through that context will start there, i is invalid, and fires
if an invalid extension is keyed in that context.
s is run when a call comes in and Asterisk does not know the dialed
number. It
Race Vanderdecken wrote:
This might seem really dumb but tack enough silence onto the back of
your file to make it five minutes long. Then the message play for 5
minutes and repeats.
Race The Tyrant Vanderdecken
This was a dumb idea.
-Original Message-
From: [EMAIL PROTECTED]
Angel Diaz wrote:
Mikael,
Well, to be more specific, I'm using ISDN PRI.
30B+D.
- Original Message -
From: Angel Diaz [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, April 11, 2005 3:55 PM
Subject: Re: [Asterisk-Users] Play Sound File Without Answer Channel
I'm using
The ONLY way to MAYBE play an announcement DURING a call is by using the
stuff put in for calling cards. See show application dial
Chris wrote:
That won't work on outgoing calls, will it?
Regard,
Chris
- Original Message -
From: Eric Wieling aka ManxPower [EMAIL PROTECTED
Angel Diaz wrote:
I want to use the Voicemail app and before that, I would like to play
an audio file but not billable in the Switch side. Than, to do so, I have to
be able to no send the Answer message during the play of the file. Then
after finish the file, I'w xecute the Voicemail app.
Tim Connolly wrote:
Well crapola... cvs-head works with Digium's te110xp, but not cvs stable.
Looks like there's a huge difference:
Stable=-rw--- 1 root root 248572 Jun 9 2004 chan_zap.c
Head =-rw--- 1 root root 326585 Apr 6 14:17 chan_zap.c
I run a te110p with 1.0.x CVS stable all
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
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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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
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
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
CuPoTKa wrote:
Hello!
Does anybody tried to work with IAX2 (client side - softphones) behind a
NATs that always increment ports?
At asterisk CLI I see:
-- Registered '12345' (AUTHENTICATED) at a.b.c.d:22269
-- Registered '12345' (AUTHENTICATED) at a.b.c.d:22289
-- Registered '12345'
Min Hwan Chang wrote:
Evening,
I'm having problems with a Polycom IP300 giving me a Stopping
Retransmission Found:102. It gives this error about every 30
seconds.
After searching the Help list, I went ahead and set Disallow=all and
allow=ulaw. This still doesn't seem to help.
Is this problem
Richard Dutton wrote:
I've seen from the Asterisk Hardware list that the Dialogic D/300JCT-1E1 and
D/600JCT-2E1 cards are supported by Asterisk, can anyone tell me if the
D/300SC-1E1 and D/600SC-2E1 cards are as a client has quite a few of these
particular model and would like to use them in an
J. Arnaud wrote:
Hi,
I am using the dial out feature
(/var/spool/asterisk/outgoing) but when I look in
CDRs,
calls that reached a all circuits are busy now,
please call later are considered as ANSWERED.
Is it the expected behavior? It there a way to change
that?
If you have analog calls are
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
Sergio wrote:
Telnetting the phone I see a good amount of free memory space.
subscribe/nority is just a firmware implementation.
I think it's just a market choice. They wanna sell their new phones with
that feature on.
What new phones do that have?
___
Scott wrote:
Is it possible to run more than one Asterisk PBX on a single server?
I don't think there would be a hardware restriction using modern gear
but is there limitations on installs etc? I know it would be trivial
to make multiple databases for AMP and likely use different ports for
the
Josiah Bryan wrote:
On Tuesday 05 April 2005 1:24 pm, Dov Bigio wrote:
Hello all,
I am looking for a list of all available sound files for asterisk and a
transcription of their content, so that I can have someone translate them
into portuguese.
I vaguely remeber reading some file in my server
Kris Boutilier wrote:
I have a PRI connection between Asterisk and a PBX. The connection passes through a
hardware echo canceller which includes some monitoring facilities. Occasionally the
T1 has gone yellow for short periods (2 seconds) and when this occurs Asterisk
seems to immediately tear
Tobias Jönsson wrote:
On Thu, 31 Mar 2005, Peter Svensson wrote:
It would not be very hard to add both features to libpri. Libpri
already has a function to decode and dump the time/date information.
If I remember correctly the time/date IE should be added to the SETUP
messages. I have been
Maik Hassel wrote:
Hello everybody,
I am having trouble setting up a SIP/analog phone gateway. The SIP
phones are working, just the Zaptel card doesn't seem to work. I am
using the zaptel TDM400P with one FXO module on the last bank (should be
channel 4 I suppose).
When I try to dial out
Glenn Powers wrote:
I keep getting this error every five minutes:
Apr 4 13:35:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread:
Request to schedule in the past?!?!
Apr 4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread:
Request to schedule in the past?!?!
Apr 4 13:35:01
Steve Mann wrote:
From what I have read, you made a small mistake, if you are not using Digium
hardware, but want to use MeetMe of Music on Hold, you still require a
timing source, regardless of kernel.
A Zaptel Timer has not been required for MoH for at least a year.
Randy Paries wrote:
Thanks for the info
OK my first questions
I have edited my zaptel.conf
fxsks=1-2
loadzone = us
defaultzone=us
I have two X100P cards installed
When I run /sbin/ztcfg
ZT_CHANCONFIG failed on channel 2: Invalid argument (22)
Did you forget that
Actually about 80k-82k when you take into account UDP and RTP overhead
and assume you are using SIP. Single IAX2 call may be a little less.
multiple IAX2 calls using trunking will be a lot less.
In fact, this question is answered on
http://www.digium.com/index.php?menu=documentation
Bernie wrote:
can that number be reduced? I'm looking at a system that would be
deployed to remote offices over fairly limited bandwidth links and need
to find a way of balancing quality vs. bandwidth constraints.
Yes. Read up on the various codecs and how much bandwidth they use.
John Goerzen wrote:
Hi,
I recently encountered an odd situation: the network cable to my
SPA-841 got unplugged while it was in the midst of a call. I got it
re-plugged in about 30 seconds, and the phone rebooted. The phone
showed no evidence of the previous call in progress and worked like
Richard J. Sears wrote:
Hey Everyone -
I am having a problem that is keeping me awake at night.ok, so maybe
not keeping me awake, but it is frustrating. :-)
I am running Asterisk 1.0.7 on Gentoo (2.6.10-gentoo-r6) on an Intel
700Mhz box with 512MB of RAM.
The system is very light, with maybe
Courtney Couch wrote:
If I were to buy 20 did's how do I know within asterisk which number was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
On Apr 3, 2005 8:56 PM, Ian Hailey [EMAIL PROTECTED] wrote:
Hello all,
I was hoping to be able to call a mobile and if it is un-reachable for
whatever reason (e.g. switched off) then I was expecting an unobtainable
response that would be detected in Asterisk. It seems that the operator
(Virgin in
Mark Halverson wrote:
exten = _1NXXNXX,1,SetGroup(${CALLERIDNUM})
Try using ${ACCOUNTCODE} and make sure the account code is unique to
each phone.
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Rod Bacon wrote:
This is quite interesting.
I tested calls to 2 mobiles that I knew were off, and not diverted to
voicemail. 1 with Telstra, the other with vodafone (I'm in Australia).
Via ISDN, both calls were shown as unanswered by asterisk. When the
calls went to voicemail, the call was
Eric Wieling aka ManxPower wrote:
Does anyone know what I need to do to use patlooptest? I have what I
think is a T-1 loopback plug in the card (1-port, TE110P), but I still
see a red alarm. Is this normal? I don't even know where to start for
this.
From Digium Support:
You will need
Kib Eki wrote:
Thanks, problem solved, I found somethind in this mailing list! Wrong
extensions.conf entry.
extensions.conf:
exten = 0237482,1,Dial,CAPI/@301:0237482,5,tr
?? But, what does ,5,tr mean ??
5 tells Asterisk to hang up if the call is not answered in 5 seconds.
t tells Asterisk to use
Brian Litzinger wrote:
iax.conf:
[general]
bandwidth=high
allow=all
jitterbuffer=no
tos=low
register = 1234567:[EMAIL PROTECTED]
[livevoip]
type=friend
secret=1234567890
deny=0.0.0.0/0.0.0.0
permit=217.160.244.186/255.255.255.0
context=from-livevoip
sip.conf:
I have dtmfmode=inband for both
Brian Litzinger wrote:
On Fri, Apr 01, 2005 at 12:12:57PM -0600, Eric Wieling aka ManxPower wrote:
Brian Litzinger wrote:
iax.conf:
[general]
bandwidth=high
allow=all
jitterbuffer=no
tos=low
register = 1234567:[EMAIL PROTECTED]
[livevoip]
type=friend
secret=1234567890
deny=0.0.0.0/0.0.0.0
permit
Brandon Patterson wrote:
Level 3 does DTMF inband DTMF. Period.
If he's using IAX he's not talking directly to Level 3.
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Jerry wrote:
On Mar 31, 2005, at 8:01 AM, Zoa wrote:
cpu load on te4xxp cards is very low, and now that they have echo
cancellers as add-ons cards, it will be even lower.
I can't speak on hardware compatibility as i never tried a sangoma card.
(But i can say that in the last year i've never had an
Scott Bussinger wrote:
forums). If only we could get people to quit posting in HTML email, life
would be grand. :)
Mozilla has an option to view ALL messages as text. I use that. I
suppose I should not. People that post in HMTL should not get my help.
Maybe I can use procmail to send an
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