On Mon, 2004-03-22 at 14:35, John Brown (CV) wrote:
I really don't want to burn a DID per WATTS line
so that I can route on the DID number.
Look at asterisk/doc/README.variables. Assuming you are using ZAP
interfaces look at ${DNID}.
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On Mon, 2004-03-22 at 15:22, Bill Hamlin wrote:
I didn't find anything like ldassume using google. Can you tell me more
about that?
It's in the RedHat 9 RELEASE NOTES.
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.
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On Mon, 2004-03-22 at 20:42, Simon Brown wrote:
I have my Asterisk server behind a Cisco firewall. I am trying to set up IAX
but I cannot work out which ports I need to open up on my firewall. I have
opened 4569, 5036, and 5060 but IAX calls will not proceed unless I turn off
all access
I have a Cisco 1750 with 2 FXO ports and 2 FXS ports. I have a
POTS line plugged into FXO port 0 and an Analog phone plugged
into FXS port 0.
I have the FXO port on the Cisco configured as PLAR OPX, which
means that when a call comes into the port the router does NOT
take the port off hook,
I have debugging on in Asterisk and sip debug.
How do I tell what username a SIP client is trying to use to
register with Asterisk as?
--Eric
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I changed 100 to 102 and 101 to 103 and the same thing happens.
On Thu, Mar 06, 2003 at 05:46:22PM -0600, Mark Spencer wrote:
exten = 2111,1,Dial(SIP/[EMAIL PROTECTED])
exten = 2111,2,Voicemail(u2111)
exten = 2111,3,Hangup
exten = 2111,100,Voicemail(b2111)
exten = 2111,101,Hangup
I like it. It's better than STUN. *grin*
On Thu, Mar 13, 2003 at 09:19:11AM -0600, Mark Spencer wrote:
What do you all think of renaming IAX2 as:
Telephony Authentication, Signalling, and Transport Exchange (TASTE)
TASTE is easy to remember and has a sort of ironic relation to SIP.
Is it
Thanks. I figured it was harmless (and seems to be harmless),
but I thought I'd report it anyway.
HOWEVER, even though we are sending a notification to the DTA310
that there are messages waiting, the message waiting light on
the DTA310 doesn't light up. I don't really care since I get my
Is there any way to require a caller to enter their customer
number when they call in AND have the info put in the CDR info?
Also is there a way do to the same for outbound calls?
--Eric
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What are the dtmf= options in sip.conf.
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I've used the AudioVox HP300-S for a little while. It's a
SIP/H.323 phone with power (power supply was included with mine)
and Ethernet ports on the back. On the front is an LCD display,
the dial-pad, and various buttons such as hold, transfer, etc. I
think it's about $140 or so. I've only
Try commeting out the username= and secret= and set the
host=dynamic. If that works you can try adding them back in.
On Fri, Mar 28, 2003 at 12:38:58PM -0500, Brian Capouch wrote:
[EMAIL PROTECTED] wrote: On Fri, 28 Mar 2003, Brian Capouch wrote:
I wonder what I'm doing wrong here, trying
You must first purchase a license from the G723 patent holders. A
license costs about US$10,000.
On Tue, 2003-05-27 at 01:52, [EMAIL PROTECTED] wrote:
hi!
From where do I get the source code for G.723 for asterisk. And how do I
compile it (is there any specialities other that make make
Does anyone have any suggestions on how to send different rings to a Zap
channel depending on if the call is from an internal extension or from
an outside line?
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I've never seen a Linksys router that was a proxy. They are all NAT
routers.
Dan Fernandez writes:
In the phone, if I set the outbound proxy to the linksys it doesn´t do
anything.
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As I understand it (and my understanding is obviously incorrect) the
switch = statement sells the Asterisk box to resolve (aka lookup)
extensions by querying the remote Asterisk server defined in the switch
= statement. The switch = statement is used to centralize dialplans.
I've not used the
ast1 and
ast2 and you'll get connected to ast2 Zap/1
Martin
On 13 Jun 2003, Eric Wieling wrote:
As I understand it (and my understanding is obviously incorrect) the
switch = statement sells the Asterisk box to resolve (aka lookup)
extensions by querying the remote Asterisk server
exten = _0.,1,StripMSD,1
exten = _.,2,DigitTimeout,10
exten = _.,3,ResponseTimeout,20
exten = _.,4,Dial,Modem/ttyI0:${EXTEN}
exten = _.,4,Dial,Modem/ttyI1:${EXTEN}
This is wrong. Should be:
exten = _.,1,StripMSD,1
exten = _.,2,DigitTimeout,10
exten = _.,3,ResponseTimeout,20
exten =
What's the proper way to specify the allowed codecs in iax.conf? It
doens't like allow=ilbc,gsm but if I put two allow= lines, one for ilbc
and one for gsm it seems to always to want to use gsm.
--Eric
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I'm starting to write an AGI script. I want to pass more than one
parameter to the script, but seem to be unable to.
extensions.conf:
exten = 85,1,AGI(/etc/asterisk/agi/args.agi,myarg1,myarg2)
args.agi:
#!/usr/bin/perl
print STDERR FNORD prog = $0\n;
print STDERR FNORD arg 1 =
I get the following message when I dial 74#. Does anyone have any ideas
on what might be going on? If I don't require numbers to be terminated
with # everything works as expected, but you have to wait for the digit
timeout, of course.
MESSEGE:
DEBUG[1150520624]: File pbx.c, Line 1683
Uh, what are we looking for other than better playback performance?
On Sat, 2003-06-28 at 17:48, Mark Spencer wrote:
I've made some major changes to the way Asterisk handles file formats.
I'd like feedback from people about any experience they have with these
changes. They *may* improve
I keep hearing Max of two T400P's in a machine. Personally I'd budget
for 2 machines and buy 1 machine and see if you can get 4 boards to work
in 1 machine, if not, you can always buy a second one. You will, of
course need 4 slots and 4 IRQ's that are not shared with ANYTHING to
support 4 cards.
I have a two remote PBXs. I use the switch = statement on each PBX to
point to the other PBX.
Now I want extensions on PBX-1 to dial extensions and PSTN numbers that
are local to PBX-2.
However, I ALSO want people to be able to dial into a Zap channel on
PBX-1 and be able to dial extensions on
If you have to set up different users for the different contexts what'
the usefulness of having a /context on the switch = statement?
On Wed, 2003-07-02 at 10:47, Steven Critchfield wrote:
On Wed, 2003-07-02 at 10:36, Eric Wieling wrote:
I have a two remote PBXs. I use the switch = statement
Which codec will be tried first?
On Thu, 2003-07-03 at 11:02, WipeOut . wrote:
in sip.conf...
disallow=all
allow=g723
allow=g729
etc..
or
allow=all
disallow=gsm
etc..
Is there a way in Asterisk configuration to force the
use of specific codecs only... for example:
That only works if you are using the G711 (ulaw/alaw) codecs. Other
codecs distort inband DTMF.
On Wed, 2003-07-30 at 15:26, Patrick wrote:
I have the same setup, and in the sip.conf file I set the dtmfmode=inband
for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
The SIP protocol is designed in a way that makes it tough to work with
NAT. The two SIP endpoints dynamically determine the ports to use for
the RTP (voice) data. Port 5060 is only used for control messages.
People have gotten SIP to work via a firewall (or iptables) but it's not
a trivial
This is a very useful patch. One of the biggest problems I've had with
another service (ComminuKate) that uses # as a control tone is that I
can't use any IVR system that needs # as a termination character.
On Sat, 2003-08-02 at 10:33, Iain Stevenson wrote:
Here's a patch that changes the
On Thu, 2003-08-07 at 17:10, Brian Capouch wrote:
The extension goes busy (from the perspective of someone else on the
PBX trying to call it) as soon as I plug it into the X100P, and when I
have asterisk pick it up I hear a faint busy signal.
You want to plug an ANALOG extension from your
for a 3 way conference, etc.) how is the port addressing managed?
Thanks,
William
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 06, 2003 3:57 PM
Subject: Re: [Asterisk-Users] Windows IAX soft phone
SIP negotiates
No. Asterisk only supports G723 in pass thru mode. Calling voicemail
would require Asterisk to transcode from whatever format the voicemail
audio files are in (usually GSM) to G723 and Asterisk can't do that.
Your best bet is to buy some G729 licenses from Digium and use that for
remote users.
SIP negotiates the port numbers it will use for voice for each call.
The call setup is done via port 5060 but the voice packets use
dynamically allocated ports and so firewalls don't know what ports to
deal with when doing NAT. IAX does NOT negotiate the port numbers for
the voice stream, it
I'm getting the following message when I start Asterisk:
WARNING[1024]: File codec_g729b.c, Line 413 (load_module): Unable to
initialize va stuff: -1
Did I mess up the registration key or is something else wrong?
--Eric
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On Wed, 2003-08-13 at 14:16, John Todd wrote:
I'm looking to implement some basic CLASS features, using my own
dialplans as well as those so thoughtfully contributed by
Is there any way on the iaxtel.com web site to see if my asterisk is
registering and what 700 number is associated with my iaxtel account? I
registered many months ago but never used it. My asterisk shows
registered, but I can't seem to receive any calls (callers get a the
user is not registered
software fax modem dsp
There doesn't seem to be many software fax modems out there, but I found
these few when doing a Google search a few months ago and bookmarked
them:
http://www.softmodem.org/
http://fabrice.bellard.free.fr/linmodem.html
On Tue, 2003-08-12 at 15:37, Mark Spencer wrote:
Couldn't agree more. The G.729 codec is so unDigium-like... don't buy
it is my recommendation.
I don't think anybody buys G.729 just to have it. They buy it because
they *have* to have it. And we sell it because they *have* to have it. I
If you can plug it into a regular analog phone line and have it work,
then it will work with Asterisk.
On Wed, 2003-08-13 at 10:42, Chris Hale wrote:
Anyone know if the ATT 964/954 series phones have any issues with
Asterisk? We have 5 phones and would like to reuse them if possible.
Any
These list messages might be useful:
http://lists.digium.com/pipermail/asterisk-users/2003-June/013918.html
http://lists.digium.com/pipermail/asterisk-users/2003-June/013946.html
On Tue, 2003-08-12 at 13:22, Steve Lane wrote:
I am trying to do the same thing you are doing. I am new to asterisk
Try adding:
exten = fax,1,Dial(blah)
Where Blah is the zap or SIP port your fax machine is connected to.
On Mon, 2003-08-11 at 15:26, Eduardo Goncalves wrote:
On Mon, 11 Aug 2003 15:15:08 -0500
Tilghman Lesher [EMAIL PROTECTED] wrote:
On Monday 11 August 2003 02:46 pm, Eduardo Goncalves
Did you try:
register = FWDnum@fwdnat.pulver.com:FWDpass@fwd.pulver.com
On Tue, 2003-08-12 at 13:24, Borut Senicar wrote:
Hi All,
Is there any way to connect (register, initiate and receive calls) with
Asterisk to FWD through NAT? Since I own my router port forwarding is not a
problem.
can?
The TDM400P takes one IRQ. The modules for the TDM400P do not take up
any additional IRQs
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In a related story, the IRS has recently ruled that the cost of Windows
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of the biggest reasons people use PRIs
is so they don't have this problem. I don't know what to suggest to
you, other than not to give up. This is a fixable issue with PRIs.
--
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In a related story, the IRS has recently ruled that the cost
.
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On Wed, 2004-06-02 at 07:41, Sergio Serrano wrote:
Hi all, I try to do next transfer:
A person contact with me, I would like transfer to other person
in next manner. I call to other person and when I say who wants talk
with him I hangup phones an call is redirect automatically to other
and support
supervised/consultative transfers (which is the term for what you want
to do)
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? Or is
it a better pots adapter for the wastebasket?
http://www.asteriskpbx.org/index.php?menu=hardware
http://www.google.com/search?hl=enie=UTF-8q=site%3Alists.digium.com+dialogic+driversbtnG=Google+Search
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In a related story
this feature.
They don't use Grandstream phones or they work around the issue if they
do use Grandstream phones. In the analog world everything is handled by
the PBX. In the SIP world most things are handled by the phone.
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In a related
it doesn't work.
show application dial on the Asterisk console. Pay attention to the
t and T options to enable # transfers.
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-
0x00460400, and 0x01a40400)
NatTimer: 0x0054000a
This is the standard config we use for ATA-186s using v2.16 firmware:
http://www.fnords.org/~eric/asterisk/ata-186.shtml
We are slowly migrating to the 3.1 firmware, but the settings are very
similar.
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On Thu, 2004-06-03 at 05:27, Tony Hoyle wrote:
Eric Wieling wrote:
Why are you even looking at VoIP? Analog ports and phones are pretty
cheap. They are not pretty, but they are cheap and all the smarts are
in the PBX.
Free calls to the US, basically, since the leased line is dirt cheap
. Regardless
the option was removed some time ago.
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.
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of interrupts being locked for too long
can be: graphics, frame buffer, IDE DMA. There might be other things
that cause this error, I don't know.
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:11 NOTICE[11276]: PRI got event: 6 on span 1
Jun 3 02:41:11 WARNING[12301]: Detected alarm on channel 1: Red Alarm
I answered this same question *today*:
From:
Eric Wieling [EMAIL PROTECTED]
Reply-To:
[EMAIL PROTECTED]
To:
[EMAIL PROTECTED]
Subject:
Re: [Asterisk-Users
.
Timothy R. McKee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Thursday, June 03, 2004 23:36
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] miserable time with Cisco ATA186
haven't tested the fix yet.
What IOS version contains the fix for QoS on Ethernet?
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,VoicemailMain()
exten = 3009,3,Hangup
exten = 3009,4,VoiceMail(u${RDNIS})
exten = 3009,5,Hangup
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On Fri, 2004-06-04 at 18:16, Maveric wrote:
What type of cisco phones? i'm using 7960's and i know they don't have a
to voice mail button. That annoys me.
Cisco 7905G is the one I had for a while.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS
On Mon, 2004-06-07 at 07:14, Duane wrote:
exten = _.,1,Dial(IAX2/username:[EMAIL PROTECTED]/number)
As far as I can tell Asterisk will not use iax.conf if you do it that
way. (specifically if you use @host rather than @iaxconfentry)
--Eric
--
Eric Wieling * BTEL Consulting * 504-899
The above will attempt to dial out your Zap interface first. If that
fails, it will dial out using username for the username and the
password, IP address info for the IAX2 peer will be grabbed out of the
iax2.conf entry that matches [iax-conf-entry].
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'
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it. :)
bkw
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Karl J. Vesterling wrote:
It would be nice to have an option where dialing two #'s would send a
single #.
I too have had difficulty with this.
My workaround is to use my cell when calling an IVR.
ANother workaround I have through of would be to use 99 for outgoing,
but with no transfer options
Remove the evil r option from the Dial line in Asterisk.
Mathieu Nantel wrote:
Good day,
I've got around to installing an X100P card in my computer to try out
asterisk. I noticed (and people who were testing with me also noticed) that
when dialing from my SIP soft phone to the PSTN, the ringer
]
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far - people who I've asked say No
http://www.google.com/search?hl=enlr=ie=UTF-8q=site%3Alists.digium.com+i4l+dtmf+patchbtnG=Search
--
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In a related story, the IRS has recently ruled that the cost of Windows
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These are the three cheap SIP phones that I've used.
Grandstream BT10x $65/street
Number only LCD
Zultys ZIP 2 $100/retail
No LCD
Uniden UIP 200 $120/retail
PoE, built-in switch
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently
. Not many people use Dialogic cards with Asterisk so
you'll not get much support from the Asterisk community. Contact Digium
if you want to buy the drivers for the card.
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost
that the phone is ringing
I am using asterisk Asterisk CVS-04/06/04-10:46:21 with T410
PRI card connected to a Nortel switch
Thanks
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
with my set up or is this a bug with Asterisk?
No. It's just the way Asterisk works.
--
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In a related story, the IRS has recently ruled that the cost of Windows
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-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday, 23 June 2004 0:47
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Busy message
On Mon, 2004-06-21 at 23:26, Simon Brown wrote:
When I dial a SIP phone which is specified in the sip.conf
.
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Also see:
http://lists.digium.com/pipermail/asterisk-dev/2004-June/004534.html
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]
www.moskaluk.com
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]
[mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling
Sent: Wednesday, June 23, 2004 9:59 PM
To: [EMAIL PROTECTED]
Subject: Re: FW: [Asterisk-Users] No dial tone after installation
I forgot to add noload = chan_alsa.so
On Wed, 2004-06-23 at 20:43, Eric Wieling wrote:
/etc/asterisk
option tells Asterisk to provide a
ringing tone to the caller REGARDLESS of what the caller should be
hearing. You'll notice that if you call a busy number you'll hear a
ring or two and then the busy signal.
--Eric
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In a related
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled
chan_iax2 has support for it, but I'm
not sure.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted
On Fri, 2004-06-25 at 09:24, Joseph wrote:
Does anyone know of a device that will take an SS7 link and convert it
to a PRI?
I think it's called an ILEC or CLEC. 8-)
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost
, but of course,
on * it is probably called something else.
Remove the r option from your Dial line.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss
cards? Maybe soon asterisk'll do it.
Jeremy Jones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Eric Wieling
Sent: Friday, June 25, 2004 9:28 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SS7 to Pri
On Fri
using 'soxmix'
utility which has to be installed on the system.
fs-2*CLI
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has
On Sat, 2004-06-26 at 15:27, Jeremy McNamara wrote:
I even manage a few different production systems with 5300s and they are
running absolutely perfectly on asterisk cvs -head with chan_h323.
Can you post the config from your 5300s?
--
Eric Wieling * BTEL Consulting * 504-899-1387
On Sat, 2004-06-26 at 16:32, Jeremy McNamara wrote:
Eric Wieling wrote:
Can you post the config from your 5300s?
They are customer owned gateways, but I can try.
Heck, post the Cisco configs, the chan_h323 config, and sample Dial
lines. The more info the better. 8
the same result.
BTW, if I want to modify my codecs in a sip context, it's not taking in
account by asterisk. Is'it normal behaviour?
Asterisk is trying to convert from ALAW to G723.1. Asterisk can't do
that. Don't use G723.1.
--
Eric Wieling * BTEL Consulting * 504-899-1387
as it came from the
install with some additions. I added longdistance to the
default context. Please help!
[default]
include = mainmenu
include = longdistance
exten = _9X.,1,Dial(ZAP/1/${EXTEN:1})
Try exten = _9X.,1,Dial(ZAP/1/ww${EXTEN:1})
--
Eric Wieling * BTEL
for Asterisk, but not the G723.1 codec.
This is covered over and over and over again in the mailing list
archives.
--Eric
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted
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