It looks like the answer is yes.
http://crazytechthoughts.blogspot.ca/2011/12/call-external-program-from-mysql.html
From the page, here is code to execute a UDF library and call a shell.
Clearly there would be a heavy penalty to launching a shell so you would
want to carefully evaluate the
I have 2 FXO channels from which I want to route incoming calls to
different contexts in extensions.conf. I edited the context entries in
dahdi-channels.conf and created matching entries in extensions.conf.
One channel is routed to the new context as I want, but the other
channel is stuck
Don't forget that many routers treat the designated private address space
differently because it assumes the device is being implemented as a border
router. In this configuration they block most traffic unless you
specifically set rules to permit traffic to flow.
-dbc.
Has anyone seen something like this before. Randomly, on longish calls, the
local side of the call audio goes dead. Meaning remote caller can hear us
but we cannot hear the remote person?
Linux voip 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:10:25 EDT 2009 x86_64
x86_64 x86_64 GNU/Linux
Asterisk
On Mon, 27 Jul 2009, Jeff LaCoursiere wrote:
1) The latest 8.09 kamikaze no longer supports the Broadcom radios, so ...
Because of closed-source drivers the Broadcom chips only work on the 2.4
series kernels. OpenWRT does make a 2.4 kernel version _and_ a 2.6 kernel
version. Use the 2.4 and the
Yeah, have it running on several units. It's really quite simple now.
- Goto System - Packages
- Scroll down to Update Package List and wait a few seconds for that puppy
to refresh.
- You now should have a list of installed packages followed by a very long
list of available packages.
- Find the
/
transcoding?
On Fri, 24 Jul 2009, David Cook wrote:
Yeah, have it running on several units. It's really quite simple now.
- Goto System - Packages
- Scroll down to Update Package List and wait a few seconds for that
puppy to refresh.
- You now should have a list of installed packages followed
One of our client Bank has 900 employees working in different locations.
They need to record all internal and external calls. Can any body suggest
Call Recording Solution for this
requirement. We need to know the Hardware / Bandwidth and all
requirements and costing.
Few questions first
Steve,
Lots of good info! So if I put a T1 card in an Asterisk Server, and a T1
card in the Norstar
How does a user on the Norstar dial 221 and reach a voip only user
connected to asterisk via
ip only? That assumes as you mentioned new users are added as voip users in
the future?
Have the
Is there any reason why I should be experiencing such bad line
quality on inbound calls from PSTN? Call quality is perfect when
plugging in a regular analogue phone.
Do you have other phone lines you can try the A200 with? Have you
asked Sangoma support?
Ditto on Sangoma support - they
Two use-cases where autofill=no is desirable:
1) If it's important that you answer your callers in strict order (i.e.
in order to meet estimated wait time commitments etc).
Not always the case. Let's look at multiple queue assignment where agents
have skills (logged in) to multiple queues.
resist.
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Seriously, if you intend on proposing this to a customer it means you are
selling your professional services. If you are asking questions like this,
how successful do you expect your customer engagement to be?
Even if someone recommends the best phone for your particular application,
you will
Ahh. Differences with the 7961 software from that of the 7960's. Sorry, need
to research more.
- dbc.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen
Sent: September-26-07 12:29 AM
To: David Cook
Cc: [EMAIL PROTECTED]; asterisk-users
Gary, if you register multiple lines with the same SIP credentials the phone
will do rollover and take care of it. (2nd call comes in on L2, etc.)
- dbc.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen
Sent: September-25-07 6:37 PM
To:
Date: Fri, 31 Aug 2007 13:19:32 +0300
From: Dovid B
Subject: Re: [asterisk-users] phone as control interface (was 99
bottles of beer)
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type:
On 8/21/07, Steve Edwards [EMAIL PROTECTED] wrote:
To control the tv in this room, press 1. To control a tv in another
room, press 2. To control the outside lights, press 3. To control the
sprinklers, press 4, ...
Before this thread I already had a Firecracker on the server, a
Has anyone here ever used a Polycom IP 4000 Soundstation SIP
Conference Phone with asterisk? If so, how well does it work and how
does it sound?
Works fine and sounds good. It's a Polycom so it has horrible webUI. You
really should use config files for it instead. Remember with Polycom the
box
being your CSU/DSU/Firewall Router but for a small office this can
actually be a good thing.
Sangoma cards with their Wanpipe drivers can do this for you.
dbc.
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David Cook
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Quoting Tim Litwiller
to connect to the speaker system I either need to trigger a ring on a
analog line to the phone interface on our speaker system, it picks up
on
the first ring, or we can manully push a button that picks up the
line.
If we do the second we would have to have something in
Quoting [EMAIL PROTECTED]:
I have two options, T1 or 15 analog lines.
The question is, if I use TE100 with PRI , will I have same issues?
I would appreciate any comments and sample zaptel.conf and
zapata.conf
15 lines should be well beyond the cost justification point for a T1 and
you will
Quoting Stephen Bosch [EMAIL PROTECTED]:
I'm trying to decide which phones to experiment with. I have these
options:
- A combination of Polycom, Aastra and Snom
- Just Polycom
One the one hand, I'd like to keep things uniform, since it greatly
simplifies provisioning. On the other hand, I
Does anyone have a small, plain services.xml file for a cisco ip
phone,
preferably one that will work on a 7960?
I can't seem to get my xml right, and no matter what I send to the
phone
I keep getting parse errors.
Thanks
Shawn
CiscoIPPhoneMenu
TitleXML Portal/Title
PromptChoose from a
to dial out on.
dbc.
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Lito Lampitoc wrote:
thanks for enlightening. So you mean, if I have 3 lines when the
caller
dialled the first line and it was busy, the call will be diverted
to the
next two available lines in random?
I don't think it's random. I think its just sequential. If main
line
is busy,
From: Ricardo Carvalho [EMAIL PROTECTED]
Subject: [asterisk-users] How to match wild card inside a GoToIf?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't
).
David Cook
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eth0
185: 260531 IO-APIC-level Cyclom-Y
193: 25788929 IO-APIC-level aic7xxx
201: 30 IO-APIC-level aic7xxx
209: 2849304364 IO-APIC-level wanpipe1
NMI: 0
LOC: 2850775576
ERR: 0
MIS: 0
dbc.
--
David Cook
of RAM they have. You will be storing voicemail
in RAM unless you put it off-device like an NFS mount, etc. (Some
mfg/models have USB2 ports and you can put a USB stick on them and
basically forget about the problem).
--
David Cook (Canada
Vincent Delporte wrote:
Hi
Most of our customers have generic names like Hospital, so I need
to
rewrite their caller ID name by looking up the number in a database
on
the Asterisk server, and rewriting the name such as Reading
Hospital
so that we know who's calling.
Any idea if this can be
Michael Sampson wrote ..
Say I have agents using a softphone like eyebeam that has 6 lines.
They
log in to the queue. Say there are 3 agents in my queue. 3 calls come
in
and all three agents are on a call. Now a fourth call comes in. Is it
possible to have it setup so that the 4 call rings
with no problems.
Best regards
David Cook
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Yes it can be configured on the phone.
Settings - Call Pref - Speed Dial Lines - pick your button edit
Putting something here overrides the ability to use it as a line button
and changes the icon to a dial pad. (I have programmed my MoH extension
on Line 6 so I can listen to my mp3 catalog)
The 7960's have an envelope that appears in the display next to a line
which has voicemail. Also, the MWI light is a logical OR of all the
defined lines.
Is there a way to tell the phone NOT to display the MWI for certain
lines but retain the envelope for all? If you get enough VM on busy
isolate them with a solid state relay or
something. A C program to turn on/off is fairly trivial and run it from
AGI. I don't want to clutter the list with code but I can supply if
anyone needs it.
dbc.
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I am using FOP .27 and I have Zap IAX trunks. Although the IAX trunks
do show and appear registered (not dimmed) on the display, they show no
activity while in use. Any ideas??
Segments of op_buttons.cfg iax.conf are included:
op_buttons.cfg
[Zap/1]
Position=23
Label=Cook (Main)%0a(905)
Thanks Kevin! That's what is great about these forums. I never thought
of using gotoif() inside ... one of those Doh! moments.
I included your concept in my standard [dial-ld] context with
${EXTEN}:1:3=800, etc. rather than by 2's, (so it doesn't overlap with
8XX area codes) and select my
Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches sort
of like the SPA's can?
Tollfree numbers for example. I can have a line for each combination:
exten = _1800NXX, Dial,
exten = _1866NXX, Dial,
exten = _1877NXX, Dial,
exten =
I echo (pun intended) Rich's response. The Spa3k is ~ok~ but echo has
always been a problem for my home office. The A200D works flawlessly.
I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA
motherboard as an always on, low powered solution.
I have seen an A200D in a
From: Filip Dr?gowski [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Two phone numbers, one SIP provider
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-2; format=flowed
all at once.
Xorcom are just about to release BRI versions of their Asterisk specific
channel banks as well.
Best regards.
David Cook
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My cell vm goes to asterisk, not the carrier. Apparently MWI is turned
on/off with specially formatted SMS messages. Anyone know how to do this
on a Treo 600? Having the phone light from Asterisk would be HUGE ...
not to mention extremely cool.
dbc.
I've been working on this off and on for AGES. There are some SMS portal
sites that claim to be able to do this as well, but I have not managed to
find one.
I had found a company called bahamasystems which has an asterisk interface but
it's a service and it's expensive.
Another poster
?
txgain=0}
signalling=fxo_ks
group=2
mailbox=500
channel=3
rxgain=0
txgain=0
mailbox=
channel=4
rxgain=0
txgain=0
Thanks, dbc.
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Obviously if Asterisk keeps going down there is another problem to be
found. However, why not start it from /etc/inittab with respawn??? Else,
poll from cron or a script with ps ax | grep asterisk | grep -v grep |
wc -l to find out if it is running. dbc. Date: Thu, 2 Mar 2006 22:01:01
+0200
inbound CLI display (CLIP) and/or the ability to specify the
outbound number you are presenting as a CLI (CLOP/COLP depending on who you are
talking to) this needs to be specified as well. By default you get neither but
both are non-charegable upgrades (in our limited experience).
David Cook
may be able to help out.
I understand the Florz patch may help you
(http://zaphfc.florz.dyndns.org/) and this post to the list may give you
a start on how to best handle multiple HFC cards in the same box:
http://www.voip-info.org/wiki-Asterisk+zaphfc+install26
Regards
David Cook
Me thinks it is time for ISDN30e and a TE110P ;-).
David Cook
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Bagnall
Sent: 03 November 2005 10:57
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Multiple
. From
memory they cost around £100-150. I am going to revisit this as a solution to
our ever increasing PSTN-GSM call spend as soon as we have our Asterisk PBX in
place.
David Cook
JP Computer Services
Delivering Business Benefit
http://www.jpcompserv.co.uk
line should just be ignored.
right now I have a context set for dring2 cadence 0,0,0
exten = s, 1, wait(30
exten = s, 2, Hangup
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maintain a naming convention that makes
; sense.
; How do I associate this with the inbound itsp so the calls come into
; the s extension in a particular context so I can deal with the DID?
I simply don't see how I associate the inbound stream with my section
heading?
Thanks, dbc.
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David Cook
purpose computing I am a huge Dell fan on quality, performance
and price point but this disappoints me.
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). The box is a Dell PowerEdge 1400SC. Apparently the SC
means Simplified Configuration and limits options on IRQ's among other
things.
dbc.
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http://www.voip-info.org/tiki-index.php?page=Asterisk+Connect+2+servers
I have posted a doc on this to the wiki. Fist time poster. I couldn't
figure out how to escape square brackets and tables looked like I would
be there all day. Be nice :-)
dbc.
David Cook
- and if memory serves, this was
on a weekend.
I immediately set it up and have been using it since. I'd like to see an
export of the CDR so I can do something with it down the road if I
wish, but all-in-all, I'm quite happy.
dbc.
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in the Dist Ring Detection however for they
make cheap DID's for low volume like home offices, dedicated voicemail
numbers, etc.
David Cook
From: Jim Van Meggelen [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] A bit of a survey: What do do if
youneedmorethan4C.O. lines
To: 'Asterisk Users
program only **
*
* (c) David Cook, 1994
*
* Set signlal lines on serial port to turn on 5vdc
* signal. Used for solid-state relay (low current
* draw on RS232C port) to switch high voltage/high
* current load
.
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their
flagship product line on commodity Intel servers on Red Hat linux -- so
we are in good company.
David Cook
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of the other guys in Toronto interested in * put together a
meetup.com group. Please join in and we can see where to go from
there.
http://opensource.meetup.com/42/
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I'm Toronto (well Pickering). I think that could prove helpful.
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Quoting [EMAIL PROTECTED]:
Anyone in the Toronto area interested in getting together to share
notes
and swap war stories?
--
Jim Van Meggelen
[EMAIL PROTECTED
flexible config
options (FXO/FXS/EM, etc.) which adds significantly to the price.
Mediatrix is list price 650.USD and the 2 port MultiTech looks to be
900. USD list.
dbc.
--
David Cook
Quoting [EMAIL PROTECTED]:
I want to in remote locations were we need to have single or 2 PSTN
lines
download program code
to then they are autonomous which I don't think is what you are looking
for.
I use a) d) extensively here. If anyone wants the code or more info,
just ask.
David Cook
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?
Does RHEL 3 have a kernel for IO-APIC if appropriate or am I expected to
do a custom kernel build to get there from here?
dbc.
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enough to be a viable production
solution?
I presume this means that I can have it ignore other patters I don't
want it to pick up at this time (spouse factor) by only specifying
certain ring patterns to have a select setting.
Thanks!
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to that of maxsilence was a null message and to discard it.
David Cook
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, MCI and ATT offer this now that I am aware of.
Yes it is very expensive, but for multi-site high-availability services
like banks, airlines and insurance companies it pays off in spades.
dbc.
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We have five offices in Canada. Our main office is in Edmonton , with
branch offices all over the nation. I would like to place the
Asterisk
server in the Edmonton office and have it route calls to the branch
offices. I would also like to have each of the branch offices have a
local phone
in the CTI world that are completely _not_
related to programming skill. The wrong implementation simply won't
have a market.
dbc.
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I'd like a good plan for this too, however this problem seems to exist
only with analog FXO interfaces. If you have 12 lines, would it not
have been cost effective to go fractional T1 then the box would be
cleaner and the problem be averted?
Quoting [EMAIL PROTECTED]:
I have just installed *
will need to change the dringx line to be the three
digit code that shows in your console when that ring cadence arrives.
So you need to call the system with that number and record the values
you get on the console, then put them in zapata.conf as appropriate.
Enjoy.
dbc.
--
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=134,0,0
dring2=137,0,0
dring1context=internal2
dring2context=default
channel = 1
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is the (or a major contributing) factor, can
we not devise some interface circuit which will allow a variable rate
on the impedance so we can dial out the echo based on individual line
conditions?
dbc.
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using the local
processing power for this.
[paging]
; Overhead paging through the sound card
exten = 2900,1,Ringing
exten = 2900,2,Dial,console/dsp
exten = 2900,3,Hangup
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http
Quoting [EMAIL PROTECTED]:
From: Kanuri, Seshu [EMAIL PROTECTED]
Dave,
I am implementing this solution and would appreciate if you can send
me the doc at this email address - [EMAIL PROTECTED]
Thanks
Seshu Kanuri
Enough people have asked me for this that I will try and condense it for
/paste but my mail reader
didn't like the columns from the doc.)
If you want it drop me a line and I'll send you the file. (Should also
probably put it in the wiki :-)
dbc.
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range forwarded as well. Registration works, but no
audio. Obviously the RTP stuff is not happy with the forwarding.
dbc.
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Default)/Disable(0 -
12Hr)
Thanks, dbc.
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Quoting Rich Adamson
On Fri, 13 Aug 2004 06:34:26 -0400, David Cook
[EMAIL PROTECTED] wrote:
I upgraded my 7960 to sip v 6.3 and my display time has now
disappeared
from the top left corner.
Funny enough my phone has done the same thing. I figured it was
just
a configuration
);
wc-ring = 0;
}
}
Is changing the wc-ring = 1 to 0 an appropriate place to fix this for
outbound-only operation?
dbc.
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questions,
Thanks a lot in advance.
Matt Gibson
Unix Administrator
Experthost / NJ Tech Solutions
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I am presuming your Asterisk box is behind your ISP. You don't actually
need user/pw to send somebody email in the outside world, but your ISP
has prevented you from _directly_ sending email to anybody and make you
go through their SMTP server which forces you to authenticate with it
like a Mail
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
context = PSTN-in
channel = 1
Thanks, dbc.
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with an openline4. I would like nothing more than to stick with
Digium
hardware - this thread and obtaining a replacement card is my last kick at
the cat.
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for this. Is it just broadcasting looking for ntp?
The net of my problem is that it is 1 hour slow. I have ntp running on
my network and it has been told to respect daylight savings time. Is the
SPA omitting this feature?
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Server to extensions
5000-5999 (mapped by extension _5XXX) will get sent to receiving server
(my.receiving.server.ca) into the local context on the receiving server.
Performing the same configuration in the opposite direction will allow
cross-calls between Asterisk systems.
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David Cook
thick. Maybe the visual person in me needs to see a matrix.
Further, If I can get two boxes to talk together like this, what exactly
is the register for ... what does it actually do?
dbc.
Quoting Kevin Walsh [EMAIL PROTECTED]:
David Cook [EMAIL PROTECTED] wrote:
[mycontext]
exten =
_5XXX,1
Perfect! Thanks for the clarification. That's what my brain needed - on
both points.
dbc.
Quoting Kevin Walsh [EMAIL PROTECTED]:
If that's on your outgoing side then you'll also need type = peer
in there. The incoming side would have type = user.
Outgoing = peer, incoming = user. Friend
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