Hello all,
I have seen some chatter again about DTMF. I see most of the talk about DTMF
around not being able to get an external IVR to recognize digits, not a big
issue for me at this time but sill interesting. My issue though, is with
talk off on a zap channel. It seems to be getting worse
I have 2 servers currently running 64 bit SuSE 10.x on AMD Opteron processors
both of which are working very well.
John Millican
Senior Partner
Director of Technology
Sentinel Communications
PO Box 9
Wentworth, NH 03282
On Monday June 12 2006 1:39 pm, Kanishka Somaratne wrote:
Hey
Does
to 1.2.6 box and
again no audio between sip channels.
*CLI sip debug
SIP Debugging enabled
*CLI
-- SIP read from 192.168.1.200:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941
From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0
On Friday March 17 2006 8:07 am, Can2002 wrote:
I'd planning on install Asterisk on a hosted Linux box we're setting up.
The hosting provider that seems to offer the best deal can install
either Debian 3.1 or SUSE 9.x or 10.0 in either 32 or 64 bit editions
(running on AMD 64 bit).
My
On Friday January 13 2006 10:14 pm, James Harper wrote:
The best I can do so far (which appears to be a bit of a hack) is
(:0S0), which says to add a '0' to the start of the string and dial
immediately. This gives asterisk an extension dialled of '0', which
isn't the 's' that i'd hoped for,
don't feel that is the best way.
My 2 cents.
John Millican
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On Wednesday December 21 2005 1:32 pm, Colin Anderson wrote:
Um, not trying to be a smartass, but a simple 2 way splitter like the one
you get in the dollar store would do the trick nicely. Then you could just
plug in a POTS phone and turn the ringer off. Don't think it would
On Saturday November 26 2005 1:41 pm, John Millican wrote:
On Saturday November 26 2005 1:26 pm, John Millican wrote:
Hello all,
I have just upgraded to 1.2 from 1.0.9 and am receiving and placing calls
as expected. I have been trying to get atxfer working and am getting the
error
on hold on Zap/1-1
and then the channels are joined again as if nothing had happened.
I googled for the error message and searched voip-info.org but no results on
either.
Thank you for any help,
John Millican
On Saturday November 26 2005 1:26 pm, John Millican wrote:
Hello all,
I have just upgraded to 1.2 from 1.0.9 and am receiving and placing calls
as expected. I have been trying to get atxfer working and am getting the
error message:
WARNING[19541]: res_features.c:844 builtin_atxfer: Did
Hello all,
Okay when you are done laughing at the simplicity of this question could
someone show me please what I have wrong in the following statement?
GoToIf($[${numdial} != [1-9] ]?15:3);
What this is supposed to do is if numdial is not a single digit from 1 to 9
inclusive goto 15, if it is a
On Friday October 14 2005 8:26 pm, Samy Antoun wrote:
--- John Millican [EMAIL PROTECTED] wrote:
Hello all,
Okay when you are done laughing at the simplicity of
this question could
someone show me please what I have wrong in the
following statement?
GoToIf($[${numdial} != [1-9] ]?15:3
On Friday October 14 2005 8:57 pm, John Millican wrote:
On Friday October 14 2005 8:26 pm, Samy Antoun wrote:
--- John Millican [EMAIL PROTECTED] wrote:
Hello all,
Okay when you are done laughing at the simplicity of
this question could
someone show me please what I have wrong
allow=ulaw
register = username:[EMAIL PROTECTED]
[telasip]
type=peer
username=*
fromuser=*
authname=*
secret=*
host=gw3.telasip.com
context=default
dtmfmode=RFC2833
disallow=all
allow=ulaw
canreinvite=no
nat=no
Thanks in advance for any help
John Millican
Hi; I've been using Asterisk for a few months now, and I have run into
an interesting issue that I thought someone else in the community may
have run into:
I have an Asterisk install set up to receive helpdesk calls, route
them to several IAX extensions and an extension which is simply a
[snipping]
I get the following message on home:
Aug 8 18:31:03 WARNING[20598]: chan_iax2.c:6820 socket_read:
Call rejected by
69.xxx.xxx.xxx: No authority found
and get this message on away
Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read:
Rejected connect attempt
Hello all,
I know this has been covered on list but can not find the answer I need, lots
of references to no authority found, but none with an answer.
I have two * servers, one behind firewall with nat the other on a dmz with
nat. Both servers register with each other successfully.
home is
On Monday August 08 2005 7:06 pm, Carlos Chavez wrote:
On Mon, 2005-08-08 at 18:42 -0400, John Millican wrote:
Hello all,
I know this has been covered on list but can not find the answer I need,
lots of references to no authority found, but none with an answer.
I have two * servers, one
what I can't find is how to buy or get Swift? If I understand
correctly, swift is the actual program that makes the speech?
IIRC, you can download everything you need to make the thing talk,
including a voice like David. It works exactly like it will when you
buy a license except there
I have been reading about Cepstral, their voices and the Digium partner
agreement with them. I see where they sell the voices and the licenses for
them, but what I can't find is how to buy or get Swift? If I understand
correctly, swift is the actual program that makes the speech?
About once a day I have noticed a phantom incoming call with a caller
ID of [EMAIL PROTECTED]cut off. When I answer the call there is a
dial tone and the call is disconnected. Any clues?
David Koski
David and List,
I am having the same problem.
I have an * box at my house
I have my asterisk server up and running on OS X and now need to add the
capability to play a sound file asking for a 5 digit number, play another
message asking for a 2 digit number, pass these variables to a shell
script, and get the result. I have tried a number of different scenarios
but
About once a day I have noticed a phantom incoming call with a
caller ID of [EMAIL PROTECTED]cut off. When I answer the call
there is a dial tone and the call is disconnected. Any clues?
David Koski
David and List,
I am having the same problem.
I have an *
About once a day I have noticed a phantom incoming call with a
caller ID of [EMAIL PROTECTED]cut off. When I answer the call
there is a dial tone and the call is disconnected. Any clues?
David Koski
David and List,
I am having the same problem.
I
About once a day I have noticed a phantom incoming call with a caller ID of
[EMAIL PROTECTED]cut off. When I answer the call there is a dial tone
and the call is disconnected. Any clues?
David Koski
David and List,
I am having the same problem.
I have an * box at my house with 1 zap (pstn on
On Tuesday June 28 2005 4:56 am, Tom Fielding wrote:
Hi all,
Sorry for this elementary question (I'm a newbie).
I'm trying to write an agi script (test.agi) and run it when I call
in. However, I'm getting an error that says application agi isn't
being found. I've put test.agi into agi-bin
Mr. DiMartino,
how about you go to the qmail list and stay there so they can listen to your
whining and not us. This is a VERY helpful list. Yes there is the
occasional question that goes unanswered, but this is rare. Stop trolling,
go away, and grow up. Sometimes is is as important to know
Hello,
I have read about using redirect on a sip channel to get * to step out of the
voice path. Is this possible with ISDN or maybe a US T-1? I would like to
have the * box answer a call, do some niffty IVR/AGI stuff, and then redirect
that call back to the originating switch to be passed on
Dear all,
I am happy to tell you that I received a Digium's TDM20B card for my
Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I
need precise instructions to connect a phone to this card. please, assume
that I have a phone (a normal analouge phone connected to the
Hello all,
After much googling I have come to the conclusion that in asterisk land
DID(Direct Inward Dial) and DNIS(Dialed Number Identification Service) are
used rather interchangeably. If this is an incorrect assumption Please
correct me. Based on this assumption if I have everthing set up
I have several toll free numbers that get forwarded to a single local
number assigned to a trunkgroup. I've asked the telco to not forward
those toll free numbers but to assign them as DIDs to the trunkgroup, so
that I can differentiate via DNID.
They said that they can't do that. That
That was a policy we did not adopt, something about using the word
'unlimited' and then not wanting to fill it with a ton of qualifiers
like 'its unlimited unless you actually use then then we will limit you,
but if you never use it then ...' :)
We termed it unlimited interactive meaning
On Thursday May 19 2005 8:38 am, Allan Regenbaum wrote:
I have been trying for days to get an outbound connection to broadvoice
with no luck ..details below ... I have scoured all postings and seem to
get similar responses but none of these seem to help... any help is
appreciated ..
my
Hello All,
I have googled and wikied but must not be searching correctly.
Assuming the TE110P has same ability as old T100P to use some voice and some
data channels, lets say I have a TE110P set to accept voice on 10 channels
and pass the other 14 channels as data. Under this scenerio i am
Hello all,
I have asked this question of Broadvoice support and the following is their
responce:
John,
Unfortunately we are not able to fully support asterisk. We refer customers
to the Asterisk forums where users are quite well versed and some are
affiliated with BroadVoice.
The only thing
I can call in to and out of * from either number/account that i have.
The problem is i would like to answer with different prompts based on
which account/number the called dialed but broadvoice sends the call as
if it came from whichever account i register second.
Executing
Hardware - Pentium 1.4 Gig - 1 Meg ram - 1 FXO100 Card - Sipura 2000 -
Local Network Router SMC -Codec 711 - Asterisk @ home (lastest)
On average it take almost 10 - 13 Secs to make an outbound call to a local
number.
Is this a normal time ? Is there something that can be done to cut this
On Monday April 04 2005 3:58 pm, Kevin Kiely wrote:
T1 PRI
This brings up the question. What is the best service for concurrent
calls?
In the case where I have a small business I might have 10-15 people
needing
to call out and they could all be on at the same time.
-Scott
Even with a T-1
On Monday April 04 2005 5:14 pm, Brian McSpadden top posted:
I believe Kevin is meaning to buy a T-1 PRI from someone, like a phone
company of a CLEC.
On Apr 4, 2005 3:40 PM, John Millican [EMAIL PROTECTED] wrote:
On Monday April 04 2005 3:58 pm, Kevin Kiely wrote:
T1 PRI
it leave. Just like television, If
you don't like the show turn the channel. It is called maturity and doesn't
need a nanny or a moderator.
John Millican
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do not use ${EXTEN:1} in my
outbound dial and i always dial 10 digits.
John Millican
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On Saturday February 26 2005 4:45 pm, John Millican wrote:
On Saturday February 26 2005 4:30 pm, Chris Ford wrote:
I tried to call you number to see what I would get and you have a verizon
Voice messaging service.
if you called the 6037862111 that is a voicemail number tyhat i was calling
, same
error. tried all the other listed proxy's, no dial out.
I am totaly stumped. Am i not providing some helpfull info? If not tell me
what i am missing and i will get it. I am sure I have missed somethins but i
do not know what/
I greatly apreciate all the help so far.
John Millican
: [Asterisk-Users] Dial out through Broadvoice
Am i not providing some helpfull info? If not tell me
what i am missing and i will get it. I am sure I have missed
somethins but i
do not know what/ I greatly apreciate all the help so far.
John Millican
The service might just be down. I
This is working now, althouh i get chopped ringback , but once the call
path is set the audio is good.
Thanks again for all the help
John Millican
just a ps in /etc/hosts i have sip.broadvoice.com mapped to the ip of
proxy.lax.broadvoice.com
dtmfmode=inband
canreinvite=no
nat=yes
Thank you for any help.
John Millican
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On Saturday February 26 2005 4:30 pm, Chris Ford wrote:
I tried to call you number to see what I would get and you have a verizon
Voice messaging service.
if you called the 6037862111 that is a voicemail number tyhat i was calling to
test knowing it would not be busy and would not bother
On Monday January 24 2005 3:29 pm, John Middleton wrote:
Hi, I've looked at the Wiki for this, have seen the Swift.agi
details, but has anyone got a current script for Cepstral and an
example of integraton in * please?
I'm a * and linux newbie, so please be gentle ;-)
Thanks
John
I just
On Monday January 24 2005 3:29 pm, John Middleton wrote:
Hi, I've looked at the Wiki for this, have seen the Swift.agi
details, but has anyone got a current script for Cepstral and an
example of integraton in * please?
I'm a * and linux newbie, so please be gentle ;-)
Thanks
John
I just
Warren Burstein wrote:
I've noticed that some callers listen to our main menu and don't
press any keys.
snip
Remember Rotary Phones? They are still in use in some homes/areas
John M
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belive I have seen on the list where wcfxs has been changed to wctdm
this may be your problem?
John Millican
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have to
agree with me and we can both say so without fear of governmental reprisal.
Now lets get back to talking about the wonderful software that this list is
about, Please. I read this to learn about Asterisk, not GPL or BSD.
Thank you for your time,
John Millican
-Original Message-
From
Eric,
I was told by Bottom Line Tech that Linksys told them to pull all units and
stop all shipments unless there customer could prove they were and ISP,
which i am not so i can not, so no [EMAIL PROTECTED] for ME :-(
John Millican
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
are a ISP/CLEC and want
to order some of these.
Gary
Eric,
I was told by Bottom Line Tech that Linksys told them to pull all units
and
stop all shipments unless there customer could prove they were and ISP,
which i am not so i can not, so no [EMAIL PROTECTED] for ME :-(
John Millican
Well just did a search on bottom line and they do not have the PAP2-NA
listed anymore. They may still have them in stock if you call them though.
Sorry
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John
Millican
Sent: Wednesday, September 22, 2004 3
unless there customer could prove they were and ISP,
which i am not so i can not, so no [EMAIL PROTECTED] for ME :-(
John Millican
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Eric Merkel
Sent: Wednesday, September 22, 2004 10:07 AM
To: [EMAIL
help.
John Millican
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[EMAIL PROTECTED]
http
Use Read instead of background
CLI show application Read
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paul Penrod
Sent: Friday, September 17, 2004 6:53 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Background() command
Folks,
Apologies ahead of
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stephen R.
Besch
Sent: Wednesday, August 25, 2004 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: how to collect user entered digits
Ryan Courtnage wrote:
On Fri, Aug 20, 2004 at 01:19:54PM -0400, John Millican said:
I have been searching thru all docs that I can find on wiki and such but
can
not get an answer. I am trying to collect a date from user input in the
form of digits dialed from the phone to use in an agi script to do a
database
, google searches. Can anyone
point me to some other places that I can search through?
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John
Millican
Sent: Tuesday, August 24, 2004 4:13 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] how to collect user
'abandon-all-hope' (language 'en')
== Spawn extension (default, 657, 4) exited non-zero on 'Zap/1-1'
agi_request: callid.c
-- Hungup 'Zap/1-1'
What am I doing Wrong?
Thank you very much for any help
John Millican (a newbie obviously)
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Thanks will do that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Walt Reed
Sent: Friday, August 20, 2004 2:13 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] how to collect user entered digits
On Fri, Aug 20, 2004 at 01:19:54PM -0400, John
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