Hello,
I have some of my users that when they leave their voicemail it cuts
off mid message.
my voicemail.conf is below.(Asterisk 1.2.7.1)
i have the maxsilence set to 200, so i assume that is not the problem
not sure if i understand what the silencethreshold does and if that could help
James Arscott wrote:
I also tried just using s , this again did not work. I assumed the
‘Extension ‘’ in context’ part of my debug meant that the siemens is not
sending, or asterisk can’t work out, what extension is being sent If
that makes sense
It means that whatever context you
If you are asking if the server can handle it it
may, it may not. I highly doubt that a 266Mhz will be able to handle
it.
- Original Message -
From:
Walter Willis
To: [EMAIL PROTECTED] ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Saturday,
Stefan-Michael. Guenther (in-put GbR) wrote:
I have a fax server with an AVM Fritzcard that is connected to port number 4
of an EICON DIVA Server 4 BRI.
If the inbound is always going to be fax, set faxdetect=off in
capi.conf, so that it just runs the default.
Otherwise, add a fax
On Sat, 5 Aug 2006, Stefan-Michael. Guenther (in-put GbR) wrote:
Hello,
I have a fax server with an AVM Fritzcard that is connected to port number 4
of an EICON DIVA Server 4 BRI. As you can see from the following debug
messages, asterisk is accepting the incoming fax call on ISDN4 and
This Guide is offered as i know only to ITSP and large distributors not
to end-users.
You could find a User Guide for SPA 3102 at Linksys Website.
Regards
Marcos Rubino escribió:
Anybody have a recent copy of the Admin Guide (not the
user guide) for the SPA3000/3102? The only one I was
able
Hello,
The log doesn't show anything about the call is terminated.
Anyway, the message Fax tone detected, but no fax extension for is just a
notice. If you don't have an extension fax in your context, nothing else
is done. With newer chan_capi you can disable this with faxdetect=off.
Russell Bryant wrote:
On Sat, 2006-08-05 at 20:43 -0400, Roy Kidder wrote:
Is there some way I can better control the execution of playbacks so that
they take place as I expect them to?
Yes, your script needs to read a line of input from stdin to wait for
Asterisk to send back the result code
On Sun, 6 Aug 2006, Stefan-Michael. Guenther (in-put GbR) wrote:
Hello,
The log doesn't show anything about the call is terminated.
Anyway, the message Fax tone detected, but no fax extension for is just a
notice. If you don't have an extension fax in your context, nothing else
is done.
Hi,
Protocol error layer 1 (broken line or B-channel removed by signalling
protocol)
This is the cause of your problem! Your physical ISDN connection is
broken. Maybe your cross/NT connection is not setup correct.
okay, then one last question before I start testing all the options in the
Randy Paries wrote:
maxmessage=0
minmessage=3
Comment these two out.
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve neither Liberty nor Safety.
___
--Bandwidth and
On Sun, 6 Aug 2006, Stefan-Michael. Guenther (in-put GbR) wrote:
Protocol error layer 1 (broken line or B-channel removed by signalling
protocol)
This is the cause of your problem! Your physical ISDN connection is
broken. Maybe your cross/NT connection is not setup correct.
okay,
In fact I will handle G.729a. Most likely two calls. This hardware complement is very close to that of a Soekris Net4801, which I use running Astlinux & 2 G.729a licenses. It handles two calls at a time, but any more and the calls get choppy and stutter.
Michael
--Original Message Text---
Hi All
Is there a benefit to using a database to hold the extensions and sip
.conf information/configurations
or is using the standard Text file just as good and no benefit is received?
Also how does one go about converting the text .conf files to a
database, and the have asterisk read it
BTW, whether the streams pass through the server constantly, or not, depends upon how you set things up. If you allow reinvites for each end point and don't force Asterisk to monitor call progress then the streams will not pass through the server once the call is established.
If you plan on
The easiest thing is to use the asterisk data base to store variables. Here
is a short example that I have where people can call in and set if they want
to be available or not for emergency calls. Below they call in and they set
if they want to opt in or out. In the dial plan it checks to see
Title: Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Hi, I just realised I think I have missed a step
Asterisk is not matching the extension from the siemens because the siemens has not even sent one yet, it is still waiting for a dial tone. When I hit 9 on the siemens it does not
np, but in general its well worth the learn tho if you like python ;)
On 8/3/06, shawn bright [EMAIL PROTECTED] wrote:
Thanks for the reply Shidan,
i have looked at this package before, but was not exactly excited about
having to learn twisted, get twisted up and running, etc... just to get
Hey there, since the original post here, i have found more about twisted and what its all about, and i dig it. There is even a book from amazon about python twisted.There is a whole lot of stuff it can do for all other kinds of network stuff that i will eventually have to pick up to finish this
I am using asterisk svn 39892 branch 1.2 and on one box I am getting
a message saying previousreload of asterisk did not finish yet even
after several minutes. What could cause this? There are no log
entries whatsoever, loks like the reload is just hanging there.
Any assistance would be
Hi
I am new to asterisk, and learning as I plod along. Currently, I am
trying to work out how to create a ring group without using AMP.
I set my inbound line to ring multiple lines by using
Dial(SIP/101,SIP/102) but when I answered the call, the lines which
didn't answer became locked with no
Hi all,
Im looking for help with my AG-168V, an ATA based on PA-1688 chip now flashed on the latest SIP firmware version 1.53
Im not able to get it registered to any asterisk or SIP based servers. Even i tried reflashing it with IAX2 protocol still the problem prevails. Im running it on public
On Monday 07 August 2006 06:36, Chris Hembrow wrote:
I am new to asterisk, and learning as I plod along. Currently, I am
trying to work out how to create a ring group without using AMP.
You should check out the book - 'Asterisk: The Future of Telephony' -
published by O'Reilly it's available
ok, this seems like a workable solution. i will give it a shot tomorrow at work.thanksskOn 8/3/06, Benjamin Stocker
[EMAIL PROTECTED] wrote:2006/8/2, Andy Kuo
[EMAIL PROTECTED]: Hi, Can you give a quick example on how to query an EXTERNAL database?Create a AGI Script. It may take actual
Hello List -Asterisk Version: Asterisk SVN-branch-1.2-r38420MI'd like to create web GUI for basic internal outbound dialing.I came across TACI which, if I could get it to work, would fit the need.
My goal is to provide the user a form with the following:OriginateUsers# _Number they wish to
First my little Sunday story.
A client of mine with a big factory calls me up that he is trying to
call in to his place because the fire alarm went off. He is dialing
the extension I gave him that will call all the extensions (and worked
before) but after 2 rings he gets a message: The subscriber
Hey Chris, You could try our Firefox plugin to acheive the same thing in an easier way (with no programming), it'll just work with whatever website you throw at it. -- www.snapanumber.com
On 8/6/06, Christopher Aloi [EMAIL PROTECTED] wrote:
Hello List -Asterisk Version: Asterisk
I have rtcachefriends=yes in my sip.conf and it
seems that the only time that I have problems is when I have two phones behind
NAT on the same IP. Can anyone lead me in the right direction ? If I set them as
statick will that help ?
Thanks.
Dovid
- Original Message -
From:
Check that status of:
${RDNIS} and/or ${CALLERID(rdnis)}) in
/path/to/src/asterisk/docs/README.variables
C F wrote:
First my little Sunday story.
A client of mine with a big factory calls me up that he is trying to
call in to his place because the fire alarm went off. He is dialing
the
- Hello there,
How to emulate Music on Hold in a PHP AGI script? Ideally, I would like my
PHP script to play a predefined file to my callers while the script has to
spend time performing some internal calculations. Does anyone know how to
do this? Any suggestions?
Thanks in advance,
Leo
I forgot to add that This is what I got from the
console:
*CLI sip show
peersName/username
Host Dyn Nat
ACL Port Status
10330
(Unspecified) D N
0 UNKNOWN
10325
(Unspecified) D N
0 UNKNOWN
10310
(Unspecified) D N
0 UNKNOWN
10306/10306
69.114.216.204 D N
60060 OK (100
Hi James,
James wrote;
When I hit 9 on the
siemens it does not get a dial tone from asterisk, I assume this is
because I have not told
asterisk to give it one.
I might be wrong;
My question is, are you sure
your ISDN ( Asterisk span to Siemens ) is up logically?
ISDN is no tone
On Sun, 2006-08-06 at 06:58 -0400, Roy Kidder wrote:
I tried it again, reading a single line from stdin and got the 200
result=0 message. Is there potential for there to be other messages? i.d.
200 result=1 or 404 file not found? Also, is there always going to be
a single line from stdin, or
On Sun, 2006-08-06 at 22:31 -0400, Leo Burd wrote:
How to emulate Music on Hold in a PHP AGI script? Ideally, I would like my
PHP script to play a predefined file to my callers while the script has to
spend time performing some internal calculations. Does anyone know how to
do this? Any
I was thinking about it but didn't have a chance to test it. This is
an answer for my question marked as #2:
2. Is there any way to detect in the DP that an extension is called
from a redirect (any variables)?
After some testing it appears that the only time this variable is
populated is when
Hi list.
I have a customer request to deploy an HP rack server (ProLiant DL
series) as the base system for an Asterisk install. They also want to
use the Digium 24xxp card. I have heard that the Digium card is
oversized and does not fit in a normal size chassis. Does anyone know
if it
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