})
exten = 0293353510,1,Macro(call-local-sip,edwin,${EXTEN})
exten = 14140293353510,1,Macro(call-local-sip,edwin,${EXTEN:4})
Is there a way to do this in one line?
exten = _.93353510,1,Macro(call-local-sip,edwin,0293353510)
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/asterisk/. You can
also set callerid via extensions.conf by using apps available. Look at
show applications via the asterisk CLI.
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your problems.
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delete the .version file in the Asterisk source and rebuild/install
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- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 25, 2005 4:02 PM
Subject: Re: [Asterisk-Users] size and quality of audio clips
effecttheplayback??
Have you
is waiting for
a timeout instead of matching a real specific pattern.
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than well
stocked with the answer to your question. Did you bother looking there
first?
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into the kernel.
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to
fxsls=97-98
switchtype = 5ess
signalling = pri_cpe
group = 2
channel = 1-23
channel = 25-47
signalling = fxs_ls
context= internal
group = 3
channel = 49-50
change to
channel = 97-98
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, identifying the location your DTMF is being detected. If it where
all zap, then it is the DTMF routines in asterisk/zapata, but as you
didn't bother to expound what is going on, it could be SIP hardware
phones acting up on you.
More details please before you go batty.
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there is a possibility that there is
also a pattern match problem on the vonage SIP device
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get the
sale as it is a feel good action.
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that if you stick to the 2 cards
per machine that you would be up to 4 machines just for those 30 ports.
I don't know what the MGCP card costs for the ADIT. Compare it to the
cost of 1 or 2 T1 ports via Digium cards.
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On Fri, 2005-01-21 at 09:55 -0800, Asterisk List wrote:
On Thu, 20 Jan 2005 22:41:08 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
Shouldn't you contact your vendor for support and not a different
vendors support channel?
As far I know, although Digium hosts the asterisk-users
.
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mailing list like this. I turnip could do more research than you did.
Try http://www.asteriskpbx.org/index.php?menu=hardware
Try http://www.digium.com/
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seem
that Dana is taking the suggestions to heart and is changing for the
better.
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that does the codec translation for you will
probably make you more likely to consolidate too many interfaces into
one machine.
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On Thu, 2005-01-20 at 17:15 -0500, Dana Olson wrote:
On Thu, 20 Jan 2005 15:26:15 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
While it will probably be handled when you move out of outlook, please
wrap your lines at a reasonable length.
Please tell me that Gmail is fine
difficult this
might be?
Ahh so you have a client that wants to do some form of dictation. Yes
AGI as it stands can handle it. Look at the docs a bit closer. How do I
know it works, My company already has a dictation system written and n
production use using AGI.
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continue extension.conf processing. So
guess what, you have to answer the phone line with asterisk and then
dial whatever phone lines/extensions/voip phones.
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to make life with udev easier on poor FC3 users.
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to an external file.
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with the code made 100% available to you, you feel it
better to clutter this list about the API than for you to read the code?
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On Tue, 2005-01-18 at 08:08 +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Steven Critchfield [EMAIL PROTECTED] wrote:
On Tue, 2005-01-18 at 10:44 +1100, Howard Lowndes wrote:
Will Wait(n) still listen for DTMF input from the caller after there has
been a Background(some
started. Alsa devices will never scale to anything we want
to try to support and therefore I doubt any effort will be put forth on
it.
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?
Direction is mostly irrelevant. It is just 16 channels to be mapped.
Then it is just a matter of telling asterisk what to do with the 16
channels. So yes it can be done.
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worked with a full PRI before.
Thanks for any insight.
No trouble at all. Glad to hear you where able to convince the client to
use a T1 instead of analog.
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point me in the right direction it will be highly appreciated.
Care to give some logs or other information so we can think about
helping you?
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On Mon, 2005-01-17 at 12:56 +0100, Alberto MartÃnez wrote:
I got this error while compiling:
configure: error: termcap support not found
get termcap development package installed for your distribution.
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(telco)
For phones attached directly, you need FXS ports.
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 17, 2005 2:57 PM
Subject: Re: [Asterisk-Users] CAS
,1,Goto(somewhere-due-to-timeout)
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interupts
just fine. Granted it is missing some from time to time on a system not
even running asterisk yet.
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to use.
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way to get commercial SS7 is to have
it with commercial
asterisk.
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sure a simple search of the archive
would show a few threads of this exact nature in less than
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, January 14, 2005 2
(). Is there a way to
do this in *?
Hmm doesn't that example you mention sound an awful lot like an IVR
setup? Essentially, once you answer the line, just make sure there is an
extension that will send you to voicemailmain.
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through the PBX.
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is either normal
or an exception/deviation of normal.
On Thu, 13 Jan 2005 10:44:51 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Thu, 2005-01-13 at 16:08 +, Patrick Lidstone (Personal e-mail)
wrote:
I don't think Kelly's response is correct, at least for TDM FXO boards
by some kind of worm. Look for any other processes that decided to
fail around the same time. I know of a few people who are experiencing
unexplained ssh probes.
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filters, couldn't you just wait the 10-15 minutes it takes
for more messages from this list to come in?
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)
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 11, 2005 5:44 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] test-ignore
On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich
to give up on linux and asterisk now as you won't
spend the appropriate amount of time learning period.
On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich wrote:
This is a test, please disregard
Steven Critchfield [EMAIL PROTECTED]
Next time you post, make sure you turn off HTML.
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appropriate
set of settings.
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recompiled the kernel from
kernel.org sources?
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: arch/i386/kernel/asm-offsets.s: No such file or directory
UPD include/asm-i386/asm_offsets.h
mv: cannot stat `include/asm-i386/asm_offsets.h.tmp': No such file or
directory
make: *** [include/asm-i386/asm_offsets.h] Error 1
-Original Message-
From: Steven Critchfield [mailto
technology.
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[-1084595552]: loader.c:334 ast_load_resource:
chan_zap.so: load_module failed, returning -1
== Unregistered channel type 'Tor'
== Unregistered channel type 'Zap'
Jan 10 08:17:18 WARNING[-1084595552]: loader.c:429 load_modules: Loading
module chan_zap.so failed!
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, the
methods to accomplish your objectives will be much easier to
configure.
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On Thu, 2005-01-06 at 12:36 +0100, Tais M. Hansen wrote:
On Wednesday 05 January 2005 21:43, Steven Critchfield wrote:
While browsing the Wiki, I found this bit of information. Maybe it will
help you out some.
http://www.voip-info.org/wiki-Asterisk+TE410p+No+Interrupts
Sadly it didn't do
that creates
the .call files.
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it is in /usr/local/sbin/pppd.
Just to help out also, here was the line we used when testing and
actually brought it up with an ISDN Ascend Pipeline router calling into
asterisk.
;exten =
9022,1,ZapRas(debug|64000|noauth|netmask|255.255.255.0|192.168.10.1:192.168.10.2)
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locations. I'm not sure how to do the detect you need to route so
fire a script.
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the trouble here is that as soon
as the flash occurs, the new caller is the one going to be stuck in an
odd state and the previous PSTN caller is going to be in unrecoverable
limbo.
Just looks like you will be SOL on utilizing the PSTN 3 way calling.
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don't need analog lines for FAX. Follow the directions above for the
ADIT and you will be able to have analog ports to plug your fax machines
and route them out the T1.
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and the PSTN is non the wiser.
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wondering whether there is a change list or something. If you
tell me here I will update the Wiki ;-)
As someone else pointed out there is a -cvs mailing list to see changes
as they are committed and give you a bit of a heads up.
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many people who compound the problem when there are several in a thread
with all kinds of alternating font sizes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, January 05, 2005 1:28 AM
To: Asterisk Users Mailing
should be able to discuss
with them just replacing the Adtran box with your asterisk box.
Then you just need to make provisions for your fax machines to get
access to a phone line either via a SIP to FXS device or maybe one of
the TDM400 cards.
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. These are all the smaller
companies you see selling service.
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be
very grateful..
While browsing the Wiki, I found this bit of information. Maybe it will
help you out some.
http://www.voip-info.org/wiki-Asterisk+TE410p+No+Interrupts
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On Tue, 2005-01-04 at 10:08 -0500, Daryl G. Jurbala wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
Sent: Monday, January 03, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
, more importantly it encourages community building by keeping
conversations on topic and on list instead of going off list and not
contributing to the communal knowledge.
Of course, there are those who say we love to see people embarrassed or
hurt.
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if they do the same thing, your browser(?) is broken. Reply
should honor the reply-to and will send to just the list. The Reply-all
might go a step farther and grab the From header, but the result would
be different.
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out. I happened to buy a new cordless phone for her and
replaced a phone that was working elsewhere in her home. After putting
the new cordless phone in, the phone by her PC started ringing properly
and was able to be used without dropping the DSL.
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.
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On Tue, 2005-01-04 at 09:06 +1300, Richard Scobie wrote:
Steven Critchfield wrote:
Okay, link this to my rambling above and you would see that by thrashing
the disk, you are actually keeping the spindle spooled up and not
measuring the spool up draw. My guess is a spooled down machine
] Sphinx
Is anyone doing anything useful with Sphinx?
There is a small amount of info on the wiki to help with implementing
it but Im curious if anyone is actually using it.
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to reach them today.
It is possible that they choose to roll the Saturday holiday onto
Monday. Of course I think Mark is still out of the country as well.
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mozilla and allowed other
neat content to be displayed as well as links that where equivalent to
DTMF sequences.
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On Mon, 2005-01-03 at 14:07 -0600 Steven Critchfield wrote
And we come full circle to a comment I made before that TDM cards are
more for hobbyist than for real serious installs. A real install is
probably going to use a T1/E1 interface and bypasses all the troubles
listed in this thread.
So far
On Mon, 2005-01-03 at 15:25 -0500, Andrew Kohlsmith wrote:
On January 3, 2005 03:07 pm, Steven Critchfield wrote:
And we come full circle to a comment I made before that TDM cards are
more for hobbyist than for real serious installs. A real install is
probably going to use a T1/E1 interface
On Mon, 2005-01-03 at 15:06 -0600, Michael Graves wrote:
On Mon, 3 Jan 2005 15:25:29 -0500, Andrew Kohlsmith wrote:
On January 3, 2005 03:07 pm, Steven Critchfield wrote:
And we come full circle to a comment I made before that TDM cards are
more for hobbyist than for real serious installs
On Mon, 2005-01-03 at 23:32 +0200, Shoval Tomer wrote:
Hi all.
Can someone recommend a good UPS for using with an * machine that
provides some linux tested software to do managed shutdown in case of
power loss?
APC
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a
mention of a T1 circuit. To finish connecting the dots for you, I did
VoIP to the office and had 24 extension capability and about 17 actual
extensions on my system at home.
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Monday, January 03, 2005 2:59 PM
the multitude of
potential variables
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?
I'm calling Matt on Monday, and hopefully he'll RMA these cards.
I hope that everyone that has a life is out enjoying the New Year.
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On Sat, 2005-01-01 at 17:25 -0700, Michael Welter wrote:
Steven Critchfield wrote:
Those first 3 all sound like you have a problem with power supply and
consistency. You don't mention what modules you have in the cards, but I
bet you have FXS ports and have too light of a power supply
init_manager: Unable to
bind socket: Cannot assign requested address
Note that that is manager.c, means it is trying to open the manager
port. Had nothing to do with SIP.
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are approaching a year or more back and don't need
to be rotated for size yet. I will do it the next time I have to do
something with asterisk that time. My debug file is 418megs for over 1
year of logging.
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?
Of course not. FXO and FXS interfaces are for small deployments. We only
have T1 interfaces and IAX2 interfaces. PRI in, a channelized T1 using
16 channels out, and a few calls a day out to our remote system via
IAX2.
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Steven Critchfield [EMAIL PROTECTED
On Thu, 2004-12-30 at 08:29 -0700, Damon Estep wrote:
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Steven Critchfield
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Any analog FXO or FXS interfaces in that box?
Of course
On Thu, 2004-12-30 at 10:22 -0700, Michael Welter wrote:
Steven Critchfield wrote:
On Thu, 2004-12-30 at 08:29 -0700, Damon Estep wrote:
Only for small deployments? How do you interface with your fax machines?
analog alarm systems? pc modems?
I think most alarm companies continuously
sequestered away after 1 to 2 weeks.
Other option is to fake a call waiting and switch out to initiate
recording before joining back.
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Steven Critchfield [EMAIL PROTECTED]
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Core.
Look into the way your distro auto loads modules(covered recently on
this list). Then study the /etc/init.d files and the /etc/rc.[12345s]
directories.
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Steven Critchfield [EMAIL PROTECTED]
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. You are into higher level
protocols.
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+of+application+commands
pay attention to background
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Steven Critchfield [EMAIL PROTECTED]
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code just fine...
Thats highly debateable as you have a insane prerequisite of running
windows first. So you start with an insecure crap software as the base.
How can you have anything that is really worth the time of running when
it is first built on top of crap?
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. I still don't see much
of a benefit to it over using IAX2 and trunking.
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Steven Critchfield [EMAIL PROTECTED]
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electrically
difficult environment and on marginal(via) hardware.
Our enjoyment of the stability has prompted us to buy a TE410P and
another T100P for growth and redundancy.
I think some users need to have their assumptions and values checked to
see how far from reality they are.
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Steven
for Dialogic, you may not be able to
use any mysql functions due to license conflicts unless you bought a
commercial mysql license as well.
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Steven Critchfield [EMAIL PROTECTED]
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to:
span=1,0,0,esf,b8zs
Odd, you have told the card to ignore the timing based on that line.
Never had a red alarm, ever.
You have either miss typed or have been lucky.
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Steven Critchfield [EMAIL PROTECTED
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