Call pickup (defaults to *8) does not work for IAX2 channels.
troxlinux wrote:
works very well , features.conf
2008/5/1 Jose P. Espinal [EMAIL PROTECTED]:
Hello List,
Does anyone here have call pickup (with *8 ) working ok on Asterisk
version 1.4.19.1 ?
--
Consulting for
The word Dialing... and Calling...
As in Dialing 911, please wait...
and as in Calling 911, please wait...
Tilghman Lesher wrote:
We're about to do another batch of sounds, and I see by my word count that we
have some extra time left over. So, suggestions will be entertained for
additional
Have you tried something like this?:
exten = +12345,1,Noop(He died of ennui!)
Rod Bacon wrote:
This is probably a very simple question, but I can't for the life of me work
it out. I'm trying to use Asterisk as a PTSN gateway to OCS (and believe I
have all the SIP issues sorted), but OCS
Unless you meant to match the literal +X., I think you meant to say:
exten = _+X.,1,Answer
(notice the leading underscore -- which indicates this is a pattern match)
Paul Hales wrote:
I did some dialplan work with numbers starting with + (outlook) and from
memory things like
exten =
An amazing change from the old days when you could only get firmware
from a Polycom authorized distributer.
Jonathan C. Bailey wrote:
Polycom is affiliated with the project in some way.. They also have an
official Polycom moderated vendor forum.
-Jon
- Original Message -
From:
Many people think ZapRAS is for modem dialin. None of the RAS stuff
support modems, as far as I know. The RAS stuff in Asterisk is for
networking via ISDN, rather than modem.
Tzafrir Cohen wrote:
On Tue, Apr 29, 2008 at 02:36:36PM +0200, Tobias Wolf wrote:
Hi all,
i have found the two
remove callprogress=yes and busydetect=yes
lotusscript wrote:
Been using the Snom 360 and 190 for a while and decided to try the Cisco
7960. The problem I'm seeing is the call terminates between 2:34 and
3:00 minutes. This only happens when using Zap channels. Internal
calls work fine. No
Arthur wrote:
I am facing the problem that
zaptel is not going online when booting,
most people run it from /etc/rc.local ...
I thought most people ran it from /etc/rc.d/init.d/zaptel
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-- Bandwidth and Colocation Provided by
Try removing the quotes from the Caller*ID info.
Steve Davies wrote:
2008/4/24 Ken Williams [EMAIL PROTECTED]:
Came upon a problem today that I thought I'd see if it's by design, if I'm
missing an option somewhere, or if my fix is the way to fix it.
We setup a remote location with a server,
allow=g723.1 or allow=g723 (I don't remember which).
aby azid wrote:
Hi,
I have softphone with a g723 codec, my question is how do i set it as Pass
thru in Asterisk?
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network
For ABE support you really should contact Digium. BTW, there is no such
thing as a sip trunk. It's a sip peer or connection or account.
Noah Miller wrote:
Hi Jared -
For the first time, I'm setting up SIP trunking between two asterisk
boxes. The calls themselves work fine, but I'm
In 1.2 it is documented in /path/to/src/asterisk/doc/README.variables,
in 1.4 the file is called /path/to/src/asterisk/doc/channelvariables.txt
The doc directory is the only official source of documentation for
Asterisk that I am aware of. Read it.
[EMAIL PROTECTED] wrote:
Dinesh Nair пишет:
No, it is not the same thing. An IAX2 Trunk is a version of IAX2 that
puts audio from multiple calls between the same two servers into a
single UDP packet. Fewer packets need to be sent so you use the
bandwidth much more efficiency because you don't have the packet header
overhead.
SIP does
it in
the /etc/asterisk directory. I reloaded asterisk and still the message
appear when i sent call to Quintum. Am I doing it right?
cheers,
Aby Azid
On Sun, Apr 20, 2008 at 11:54 PM, Eric Wieling [EMAIL PROTECTED] wrote:
Use the indications.conf.sample that comes with the Asterisk source
Make sure you have a valid /etc/asterisk/indications.conf
aby azid wrote:
Hi,
this is my first ever post, would appreciate if anyone can explain it to me
this status message:
*[Apr 20 19:12:31] WARNING[759]: chan_sip.c:3966 sip_indicate: Don't know
how to indicate condition 9
[Apr 20
Use the indications.conf.sample that comes with the Asterisk source.
aby azid wrote:
Thank you for replying,
How would i know, whether i have the valid indicitions.conf ?
On Sun, Apr 20, 2008 at 8:47 PM, Eric Wieling [EMAIL PROTECTED] wrote:
Make sure you have a valid /etc/asterisk
What country are you in?? Yes, it is common for cell phones to
disconnect the call if they receive CONGESTION, but not BUSY.
Horwich IT Services (Godwin Stewart) wrote:
It *is* standard procedure for a cellphone to terminate a call immediately
it discovers that the called number is busy. It
The PSTN only allows ulaw or alaw (depending on your location). You
CANNOT send calls in any other codec over a PSTN line. Generally, if
you want to use G729 then you must buy a G729 license (with a few
exceptions).
Jeremy Mann wrote:
But I want my polycom to attempt g729 on SIPPEER-SIPPEER
.
Is there any dialplan logic that can coerce the transaction to be ulaw only?
Setting something in the SIP header perhaps?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Tuesday, April 15, 2008 8:05 AM
To: Asterisk Users Mailing List
That would work just spiffy if you are calling another SIP device, but
by the time the call gets to that point in the dialplan the codec of the
originating device has already been chosen and set in stone.
Tilghman Lesher wrote:
On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote:
But I want
/156-083514c0, ) in new
stack
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Tuesday, April 15, 2008 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Codec
That would work
This is a very common issue with Asterisk. There is no good fix, but if
you make sure ALL IP addresses of the server are listed in /etc/hosts on
the server it may help.
Marius Muja wrote:
It is using a local DNS server.
On Fri, Apr 11, 2008 at 10:43 AM, Steven Kurylo [EMAIL PROTECTED]
Any time you have this kind of hard lockup with a Digium card you should
run, not walk to the nearest phone and call them.
broadband Voice wrote:
I installed the Digium T1 card on Dell Poweredge 2950 and the system crashed
several times, we got a Kernel Panic and first though it was the OS so
core show application retrydial
You need to do a core show applications and look at what apps are
included with Asterisk.
Tony Mountifield wrote:
Has anyone here implemented Ring back when free in Asterisk?
The way it works in the UK is as follows:
1. A calls B. B is engaged (busy).
2.
A w in the D() string will wait .5 second. Example:
Dial(Zap/g1/5551212,,D(ww668))
If you are dialing out of an FXO or FXS signaled port, you can add w
to the dial string to wait .5 second. Example: Dial(Zap/g1/ww5551212)
Pete Kay wrote:
Is there anyway to have Asterisk to wait for 1 second
Yes, some kernels don't work with ztdummy. This is discussed over and
over and over again on this mailing list. Check the archives.
Tzafrir Cohen wrote:
On Wed, Apr 02, 2008 at 02:16:56PM -0400, Jerry Geis wrote:
On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:
/ I have no card
Rizwan Hisham wrote:
Hi,
Does anyone know the purpose of /n attached at the end of the dial
command below
Dial(Local/[EMAIL PROTECTED]/n Local/[EMAIL PROTECTED]/n)
Yes, and you will too when you read localchannel.txt in your Asterisk
source code docs directory.
--
Consulting for
This is a reasonably common problem. ztdummy uses the Linux kernel Real
Time Clock (RTC) and something is wrong with it. The solution is to
recompile your kernel, you should search the mailing list archives.
Prepend site:lists.digium.com to your Google search to limit your
search to the
responses inline
bilal ghayyad wrote:
So I would like to ask, did any one try it later and
wether it is good or not? I am asking this because I
need to use it as it is NAT Transparent (as I read
also, and I did not try it to see how much it is
transparent).
Thousands and thousands and
You will never get latency on a network low enough for echo to be
perceived as sidetone (like on analog). If you want to get rid of echo
you must cancel echo.
Mojo with Horan Company, LLC wrote:
Sean Dennis wrote:
bilal ghayyad wrote:
Hi All;
I have been chocked just when I saw some
Inband only works with the ulaw and alaw codecs.
David Nedved wrote:
Hi All,
Anyone have any idea what could cause incoming calls on a SIP channel
to no longer be able to use DTMF? DTMF on incoming calls on zaptel and
on local SIP softphones and ATAs all work fine. Nothing gets
Looks to me like you changed the ethernet controller. The G729 copy
protection is based on the MAC of the interfaces in the system.
Guilherme Loch Waltrick Góes wrote:
I'm trying to use the Digium suplied G.729 Codec, I have ran the register
utility, and got my licenses written to
Watkins, Bradley wrote:
Being able to pass variables around between systems is by *definition*
channel-specific, since the channel driver is the module responsible for
speaking a given protocol. Besdies, SIP already has (and has had for a
long time) a method for doing this (SIP headers). So
The only messages I have EVER seen Digium remove from the mailing list
archives are discussions about this unlicensed codec.
Martin wrote:
Download an appropriate binary from
[url removed]
and just drop into /usr/lib/asterisk/modules/
add allow=g723 to your sip.conf as necessary and restart
exten = i is for IVRs.
You would need a wildcard catchall extension like the one below.
Unfortunately you are doing the classic newbie mistake of thinking you
can have a simple dialplan by making the SIP user/account ID be the same
as the extension. Eventually you will realize this is a bad
As this is the mailing list, not the archive, that's not surprising.
Jaswinder Singh wrote:
That's strange , i am able to see the *url* in Martin's reply .
On Sun, Mar 23, 2008 at 6:14 PM, Eric Wieling [EMAIL PROTECTED] wrote:
The only messages I have EVER seen Digium remove from
Without knowing the line type, card model, etc, I doubt anyone can help
you. FXO signaled ports do not support receiving the dialed number.
mark morreny wrote:
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to receive calls,
Mian M Asif wrote:
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,...and these user can call to each
other. Now i want to configure separate voice mail box for
They get UTC/GMT from the NTP server. It is up to the firmware on the
phone to convert that date/time into the local time. No, it is not up
to Asterisk, it is up to the phone firmware.
Chris Carey wrote:
They get the time from their NTP server
On Mon, Mar 10, 2008 at 11:59 AM, Don Smith
Lower the rxgain and txgain on your Zap channels.
bilal ghayyad wrote:
Hi Brent;
I have been suffering from this problem since about 2
monthes and until now still did not resolved 100%.
First of all, I need to tell u that mostly u have a
problem that the first digit is duplicated, for
is already -2.8 and if I drop my txgain below +4 callers complain that
they can't hear the users on the Sip phones inside the offices.
Thanks,
Brent
Eric Wieling wrote:
Lower the rxgain and txgain on your Zap channels.
bilal ghayyad wrote:
Hi Brent;
I have been suffering from
I would strongly recommend ESF/B8ZS. If you have a RED alarm that means
the device does not see a line connected to it -- check cabling.
Mark Best wrote:
Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system
to work with *?
I'm not getting inbound or outbound calls to
Yes, they are listed in the Admin Manual for the Polycoms
Peter Hessler wrote:
Is there a way to crank the volume on Polycom speaker phones? The 430
and 4000 that I have are quieter than expected. The volume on the
device is turned all the way up, but are there firmware options that
can
?
TELCO---T1---MITEL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Tuesday, March 04, 2008 12:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mitel SX-200 + *
I would strongly
Andres Jimenez wrote:
On Sat, Mar 1, 2008 at 6:00 PM, Lacy Moore [EMAIL PROTECTED] wrote:
Not sure on #1, but #2 is not possible on SIP.
Busy Lamp Field IS available on SIP.
If it is not, I cannot imagine how my GXP-2000 does it.
I think the poster was specifically referring to Cisco's
Use most anything except FXO signaled ports. PRI, BRI, SIP, IAX2, etc.
aymen warfalli wrote:
How to calculate the PSTN call durations through zaptel ,where in the CDR it
gives the time durations started when the zaptel answerd + PSTN dialing
time + ringing time even thoug the destinations
phone1.cfg:
call.callsPerLineKey=1
Raúl Gómez C. wrote:
Michael,
I haven't used nor configured a Polycom phone, but you should check in
/etc/asterisk/sip.conf the call-limit param of the phone's config.
On Thu, Feb 28, 2008 at 12:31 PM, Michael Munger
[EMAIL PROTECTED] wrote:
We
Could this be ECFO?
Echo Canceler Freak Out, this happens when the rxgain is too high and
the echo canceler freaks out. Some users describe it as screeching,
feedback, static, or other useless terms. If users report static
on a system where there cannot be static (all digital, PRI, SIP,
My guess is a mismatch between Asterisk, Zaptel, and libPRI. Make sure
you are running the latest of each.
Tzafrir Cohen wrote:
On Mon, Feb 25, 2008 at 03:27:07PM -0500, arkda wrote:
Hi,
I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and
I've ran into an issue. After
I must have started reading this thread after you reported that you
actually had an AUDIO problem rather than a RINGBACK problem.
The issue you experienced is a common one. Someday I hope Digium fixes
that bug/design flaw.
Fons van der Beek wrote:
Jared YES
That seems to be the problem!
This problem would happen if you did not have /etc/asterisk/indications.conf
Fons van der Beek wrote:
I tried that, its gives me the same problem.
Kevin P. Fleming schreef:
Fons van der Beek wrote:
Because i want a ringing signal while people are in a waiting queue i've
created a wav
205 gives ringing
Eric Wieling schreef:
This problem would happen if you did not have
/etc/asterisk/indications.conf
Fons van der Beek wrote:
I tried that, its gives me the same problem.
Kevin P. Fleming schreef:
Fons van der Beek wrote:
Because i want a ringing
Replying to my own post. Asterisk uses indications.conf when it has to
provide tones AFTER the line is answered. You might get a message on
the console like Unable to handle indication 15 or something like that.
Eric Wieling wrote:
Don't answer the line. Also try using the US indications
:
NOT answering did the trick!
Tnx a lot! now it works like it should work!
Eric Wieling schreef:
Replying to my own post. Asterisk uses indications.conf when it has
to provide tones AFTER the line is answered. You might get a message
on the console like Unable to handle indication 15
No that will not work. You would want three exten = lines, one for
each area code.
Michael Munger wrote:
Will this work to match any number from the 770,404, or 678 area codes?
_[404|770|678]NXX
If this won't work, is there a pattern that will do this?
Yours,
That will match the following as well
770
700
740
400
470
670
600
604
608
etc.
You example says:
The first digit can be 7 or 4 or 6. The 2nd digit can be 7 or 0 or 4.
The 3rd digit can be 0 or 4 or 8.
Mike Trest - Personal wrote:
[746][704][048]
[At 01:21 PM 2/21/2008, you wrote:
On
If everything else fails, check to see if 1) DSL is available and 2) if you
can cancel within 15 days and not get a cencelation fee. Then order DSL for
that line. The telco will HAVE to fix any really significant issues on the
line before getting DSL to work on the line.
Obviously, try to get
Giorgio Incantalupo wrote:
Hi,
is there anybody who knows what this warning means??
WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type
I would bet that [10] doesn't have a type=
___
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Date: 11/14/2003 03:43 AM
Fix your system clock!
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
Mauro Zanin wrote:
Try different entry in this parameter.
In Italy mobiles start with 3, while public services with 1 and normal user
numbers with 0.
Using pridialplan=none, every number different from 0 was resulting in
termination code 1, normally used for number never seen on the network.
This is an interesting document about VoIP and Echo.
http://www.cisco.com/univercd/cc/td/doc/cisintwk/intsolns/voipsol/ea_isd.htm
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Mason Loring Bliss wrote:
Hey, all. I have the following, and
ignorepat = 9
; Testing - access to telco1/FXO
; XXX
exten = _9.,1,Dial(SIP/outboundfxo/${EXTEN:1},20)
exten = _9.,2,Hangup
Unfortunately, once I hit 9 on a connected phone, I do *not* get a dial
tone back.
ignorepat does not
VaibhaV Sharma wrote:
Hello,
In my quest to figure out the source of the random echo on our shiny new
asterisk install, I have been using ztmonitor on the TDM400p channels
for the good part of today.
I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to
them (last 2 channels
Steve Maroney wrote:
ignorepat only works for analong phones connected to FXS modules.
It also works for the IAXy and might work for MGCP and SCCP devices,
since dialtone is controled by the PBX for those protocols.
___
--Bandwidth and Colocation
Eric Bishop wrote:
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed
Bruce Ferrell wrote:
Then, on a commercial turn up (back when I did these, it was Western
Union and/or MCI), the tech at the other end would again dialup the
milliwatt, report the value measured over the loop and the pad(s)
re-adjusted to match the values for the loss in a document provided.
before dialing using HEAD from
2/24/05.
Anyone have different experiences, and if so, with what flavor of Asterisk?
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use
Innocent Evil wrote:
Hello,
Would you please suggest me, where can I buy g723.1 liscence in cheap.
I might need a liscence for 10-50 channels.
You can't.
Here is the licensing priceing info for G723.1 direct from the patent
holder's web site (it's not cheap):
John Novack wrote:
Started a new thread as my problem is somewhat different than the OP.
Seems his somewhat different problem doesn't work as advertised either.
Eric Wieling wrote:
I don't know what the problem is, but this is what I use and it works
on my analog FXO port.
exten
Joseph wrote:
On Tue, 2005-08-23 at 12:22 -0500, Eric Wieling aka ManxPower wrote:
I don't know what the problem is, but this is what I use and it works
on
my analog FXO port.
exten = _9NXXNXX,1,Dial(${PSTN}/w${EXTEN:1})
I've been trying to duplicate your pattern but I get, Invalid
or if the extension is busy.
An alternative would be to forward the call to another extension and/or
voice mail.
You looked at the features.conf.sample file?
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called
to the console in a way that will allow
me to grep or otherwise filter the text so I can focus on something in
particular?
You mean like the info in /var/log/asterisk which is configured via
/etc/asterisk/logger.conf ?
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate
Matthew Boehm wrote:
Eric Wieling aka ManxPower wrote:
You mean like the info in /var/log/asterisk which is configured via
/etc/asterisk/logger.conf ?
Damn. If I change any logging, that's going to require an asterisk
restart isn't it?
voip-1*CLI logger reload
== Parsing '/etc/asterisk
Brian May wrote:
On Wed, Aug 24, 2005 at 09:04:53AM -0500, Eric Wieling aka ManxPower wrote:
You looked at the features.conf.sample file?
Yes.
I don't see how that helps, at least in my version.
There is a parameter to change the timeout time, but I don't want to
change the time, I just
Steve Gladden wrote:
I'm looking for some help in how to keep asterisk from doing this.
If we loose Internet or routing to our upstream provider even for only a
few short minutes asterisk quickly gives up never tries again.
I have to do a manual reload to get it to register with my
sip
May 2004.
Does it mean that the *-1.0.8 doesn't have it?
The option D: doesn't seem to work for me:
exten = _51,1,Dial(SIP/[EMAIL PROTECTED],30,D(218))
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called
created automatically and have been for a very long time.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial
John Novack wrote:
Eric Wieling aka ManxPower wrote:
John Novack wrote:
Or, in the example below, wait before dialing?
exten = s,1,Dial(ZAP/g1/${ARG1},360) ; ARG1 is the number to be dialed
If you are using analog ports, yes. Dial(Zap/g1/ww15551212).
exten = s,1,Dial(ZAP/g1/ww
Colin Anderson wrote:
Recompile zaptel with
- MMX enabled
- Enable the AGGRESSIVE_SUPPRESSOR with MARK2
Excellent suggestion, I had forgotten about that. Note to those that try:
Enabling MMX in Zaptel will bugger up SpanDSP, your faxes won't recieve
correctly. Why? Dunno. Just my
Sherwood McGowan wrote:
I haven't been able to find an answerand got no response whatsoever to
my previous questions concerning it.
Has anyone found a fix for the remote connections to the CLI causing
crashes? Also, is there a known limit?
I have a huge need for using asterisk -rx in
John Novack wrote:
William Suffill wrote:
I'd suggest Dial(trunk/1800555,30,D(1wwww2)
That will cause it to dial that DMTF string on connect and w causes a
pause. I haven't tested it just referenced
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
and
John Novack wrote:
Or, in the example below, wait before dialing?
exten = s,1,Dial(ZAP/g1/${ARG1},360) ; ARG1 is the number to be dialed
If you are using analog ports, yes. Dial(Zap/g1/ww15551212). If using
digital ports then no, you can't have a delay before dialing the number,
but
-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Eric Wieling aka ManxPower
-Sent: Monday, August 22, 2005 3:15 PM
-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Subject: Re: [Asterisk-Users] asterisk -rx (or remote
-connections in general)
-
-Sherwood McGowan
Sherwood McGowan wrote:
Maybe I am, I don't doubt it.
But why does asterisk deadlock then when about 5 or 6 scripts hang while
getting output from *?
I don't know. The AMI (Asterisk Manager Interface), accessed via a TCP
connection to port 5038, has some known problems with many
= _8NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
See the notes in my .sig about the r option to dial.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care
Kamran Ahmad wrote:
i m getting follwing messages in asterisk-1.0.9 what
is the reason calls are not going out. can u pls tel
me how to solve this
Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new:
Unable to allocate socket: Too many open files
Aug 20 13:06:09 WARNING[7706]: channel.c:311
in post 1.0.x it is called wctdm. It's
pretty easy to see what card goes with which driver. The info is in the
README in the zaptel source directory.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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[EMAIL PROTECTED
[EMAIL PROTECTED] wrote:
Hello everybody. Recently I've been trying to limit the duration of some
calls for a simple application I'm writing. Unfortunately all of the
documented methods are failing and I'm not sure what else to try. Here
are some samples of what I've done:
I believe this
Sean Rima wrote:
Eric Wieling aka ManxPower wrote:
Sean Rima wrote:
Does anyone have any experience of these, I have been offered one and am
thinking of adding sticking it onto the back of my Asterisk box and just
ignore the WAN port if possible, It would be to stick my exisiting
phones onto
Falck Kenneth wrote:
Thanks, I was misguided by
http://www.voip-info.org/wiki-Asterisk+Variables which didn't mention
this.
Yeah. Nobody ever seems to mention on the Wiki when a specific feature
became available.
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Il Neofita wrote:
I put these lines on features.conf in asterisk CVS-v1-0-08/16/05
[featuremap]
blindxfer= ##
automon = *1
atxfer = *2
You need to use CVS-HEAD for those features. You are using 1.0.x CVS
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Rollin Weeks wrote:
Has anyone tried attaching calling a Dialogic D/41JTC-LS (analog) device on
another system from an asterisk system with TDM10B?
Calling to asterisk from the outside, asterisk correctly dials the internal
line and makes the connection to the Dialogic system. A few seconds
Brian Deep wrote:
[from-sip]
exten = _9.,1,Dial(SIP/[EMAIL PROTECTED])
exten = 201,1,Dial(SIP/201)
exten = 202,1,Dial(SIP/202)
try exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
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or too soft it can
cause errors in receiving Caller*ID Info.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial
Todd Weiser wrote:
We recently changed our asterisk system to begin using G.729a as the
primary codec. We have a Cisco 1700-series router which connects to the
PSTN via FXO ports, along with Cisco 7940 SIP phones. Everything is
working great, except... When an inbound caller calls into our
Pudenz, Duane wrote:
We are testing our Asterisk server prior to deployment. The server has
a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and
one PRI for local calls.
We are using sipp from two different stations routing a test number out
the LD lines and another test
disconnect sense, or
do I have to look at the key system (hybrid PBX) that it's attached to.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
Sean Rima wrote:
Does anyone have any experience of these, I have been offered one and am
thinking of adding sticking it onto the back of my Asterisk box and just
ignore the WAN port if possible, It would be to stick my exisiting
phones onto the asterisk box
No, you would ignore the LAN port.
Alan Bunch wrote:
Anybody using hardware echo cancelers a PRI at a time ?
What did you use ? How much did they cost ? Where'd you get em.
Tell us about them !
I'm playing around with the Tellabs echo canceler. Nothing to report yet.
___
-users
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
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