There is no such device -- it's outside of the POE spec.
Class 3 devices are allowed to consume at max 15.4W. Most phones are class 3
devices. The math just doesn't work out. Even if you used the draft standard
for class 4 (~30W), you could still power max 2 devices at 15W/ea.
-Dave
On Thu, Jul
You probably have a cron job running that executes 'asterisk -rx'
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Monday, April 05, 2010 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subj
Hi All
I'm involved in discussions with my carrier right now and am wondering if
anyone has interconnected Asterisk to Metasphere via SIP?
Thanks
Dave
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
> and also to do LCR and Quality based routing of International calls?
I don't know what that means.
LCR = "Least Cost Routing"
Routing a call based on the quality or cost of a route (PSTN term vs SIP to
PSTN term vs SIP to SIP) is actually quite common.
--
Bumping a thread without adding anything useful is annoying. If you do
it again, I won't be helping.
Although I have gotten quite a chuckle from your posts, it's really going to
hurt when you fall from that high horse.
--
_
-
I would love to see any info on this as well. I see similar issues with meetme
bridges having locked channels. It's easy to set a timeout but a fix (maybe I'm
just doing something wrong?) would be better than a workaround.
-d
-Original Message-
From: asterisk-users-boun...@lists.digium.
Does anybody use the Morsecode app for anything interesting? I'm strangely
fascinated by this core piece of Asterisk functionality.
Duh! How are we going to spread the word about how to take those alien bastards
down if we don't keep morse code around!?!??!
http://www.imdb.com/title/tt011662
I've been waiting for this for years. Except that snom phones are crap -- I
would really like to see openvpn or ssh tunneling hacked into a Cisco phone...
But it's still awesome.
-dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists
I have upgraded the phones to the most recent firmware (POS3-08-11-00)
and * is "Version: 1:1.4.21.2~dfsg-3+lenny1" (debian).
That doesn't look like cisco firmware to me... Unless I'm mistaken. What
version are the phones on? (Settings => Status => Firmware Versions)
-Dave
--
___
However, if you're going to be doing
massive joins for reporting, you're better off using something else (or
running individual MySQL slaves, whose purpose is to run those complex queries
and doing nothing else). In a past life, our MySQL database ran circles
around Oracle, Informix, and DB2... u
many people around think mysql is not a good option for database, they
think mysql
is only suit for small business. but i want to have a try. i need to
convince them to use this.
This statement is absolute BS. Give me some factual, backed statements by
trained database professionals who don't
This is WAY OT but I had no idea what fnal.gov was, so I checked it out:
http://computing.fnal.gov/xms/Services/Getting_Services/Web_at_Fermilab/Professional_Home_Pages_at_Fermilab
And I quote "...professional information about themself..."
About themself? Really? Really?
That is all.
Cheers
Da
Admittedly I didn't read your SIP debug (on the mobile), but do you have
reinvite=no set for the extensions and SIP trunks (providers)?
This sounds on the surface like a classic case of the Mondays. Erm reinvites I
mean.
1. Incoming call from pstn/viop provider
2. Call is answered by a user
3.
Going foward, is there any way to programmatically inject DTMF tones into an
already-bridged channel?
Well, due to the lack of responses, either I missed something obvious or nobody
cares. I'm really hoping I didn't miss something obvious... :).
In any event, I got curious of my own old quest
'w' is really only supported on channels where digit-by-digit dialing is
the norm, which generally means analog trunks (or digital trunks using
CAS signaling).
Thanks Kevin, that's what I figured (though not quite so concisely)...
Going foward, is there any way to programmatically inject DTMF
This doesn't work?
Dial(SIP/*31#ww061234123412)
When I was browsing the sip debugs, it seemed that the 'w' was not being
honored for one reason or another. My thought at the time was maybe it didn't
work at all over SIP.
Does the w *just* work with dahdi or does it work over sip as well (assu
But then the other peer says:
-- Called *31#w06123456...@xs4all-out
-- SIP/xs4all-out-0234 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION'
Anyone an idea where i should look, or how i
And how will we ever re-write the 10+-year-old RFCs which no longer hold
relevance to modern email clients if nobody goes against the grain and does
what makes sense rather than what has been generally accepted?
-Dave
And to add on to this: aside from whether you think it is silly or not,
ther
I would have read your message but I couldn't find it amongst all of this
garbage...
:)
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne
Sent: Friday, January 08, 2010 11:10 AM
To: Asterisk U
I haven't had a good mailing list war in a while.
Yes, gmail DOES default to top posting, because bottom posting is silly (in
general, but especially for a client that hides quoted text (like gmail)). Top
posting is modern. And better. And doesn't make me scroll through 10 thousand
messages and
Gmail DOES process those headers...
>
>And a proper mail client will also parse the headers and provide unsubscribe
>information/buttons based on that
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing li
In that case, you're going to have to talk to your provider.
They SHOULD be able to easily send the DID with the call...
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Taylor
Sent: Tuesday, December 15, 2009 5:17 AM
To: Ast
This may belong on -biz, but does anyone have experience with a decent and
cheap IVR/prompt recording house?
Are decent and cheap mutually exclusive?
A nice *sounding* lady would be nice... you can keep any burly voice studios to
yourself :)
Thanks
Dave
___
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last "incoming" label defined in those trunks' contexts in
sip.conf.
My ITSP insists on insecure=very in the trunk context; is this the cause?
Your provider is probably sending the DID in the SIP header TO: field. Th
Just a guess, but the connection probably went from full to half duplex.
Full vs. Half duplex networking would NOT cause half duplex phone calls.
-Dave
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
A client has two offices in the Virgin Islands that MUST maintain data
connectivity, and there are no available "leased line" options to run
a P2P link between them.
Is there line of sight? I've been wanting to do a long-shot wifi link and my
company would give it a shot if you want :).
Do you
My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP packet
obtained from the CLI with SIP Debug On. Other than I stripped out the IPs
The variable you are seeking is ${SIP_HEADER(TO)}
I parse the SIP head
If you had 1gb of memory, a 200mb load with everything else would be pretty
taxing. Hope this is helpful.
What distro are you using?? If linux is using 800Mb of memory in an idle state
for anything other than file system caching, there's a problem...
-Dave
F
Cisco 7960 does not do BLF (at least not on the SIP firmware) but the
> 7961 might. It's a shame they haven't added such features, but there we
> go.)
It does with the skinny firmware :)
The skinny channel driver also comes with the 'random crash' feature ;-p. But
truth be told I only every tri
Cisco 7960 does not do BLF (at least not on the SIP firmware) but the
7961 might. It's a shame they haven't added such features, but there we
go.)
Are you sure about this? I believe the 79xx series on 8x SIP firmware loads
does BLF with SIP/TCP, just not SIP/UDP.
-Dave
I use two 'lines' though 'Line appearances' would be a better term, though
still confusing in my book.
One line for incoming, one line that auto-answers for paging.
Cisco really has so many line appearances on their phones to enable BLF using
SIP over TCP.
-Dave
From: asterisk-users-boun...@l
And? Noticed any significant performance advantage?
Massive increase in performance on mysql VMs with database sizes that exceed
memory size (file caching). Boot times on VMs (windows and linux) under 10
seconds.
There is no noticeable change in performance for normal operations on normal
VM
I recently implemented a vmware host using SSDs for the VM storage.
I wish you could see the grin on my face right now. It's so fast.
Remember thought that all SSDs are NOT created equal... Be careful what you buy.
> On a closely related note, has anyone built a normal (not embedded)
> system o
Thanks for the reply. I am not getting any output from the Asterisk CLI when I
place the call. The phone give busy signal as soon as I push the first digit
of the extension #. When I call the 7961 from another extension I get the
following on the CLI - that works fine.
If the phone gives a
Customers in Europe all have mobile phones, while senders in North America
rarely have them ( they have answering machines, though ).
What planet/year are you/your clients living on/in? I don't know anyone who
doesn't have a mobile. Maybe it's just that they call it a cell phone instead
of a
Not trying to be a smart-a$$, just hoping to find something a little smoother.
Is there a better way, or is help as useless as it is starting to appear?
If you're actually 'sitting' at the *nix console, use CTRL+PageUP to scroll
back up in the buffer.
If you ARE using the console directly,
There are some other methods to display content on the phone screen without
editing local configs. Check http://www.ciptec.co.uk/ - commercial site but
shows the way.
If you just want to display user info on the phone, why not use the idle url
feature:
http://www.personal.psu.edu/wcs131/blogs
Here is a link to a reboot script http://www.dave.vc/wordpress/?p=14
that uses your ability to "press keys" on the phone. You could apply
the same idea to "press" the correct buttons to change the background
without rebooting.
I can't find the script that I found to do this, but I'll keep lookin
What say you to the proposal that some approaches to seeking help are
so ridiculous they should not be tolerated?
I say give me a break.
Pre-judging people doesn't work on mailing lists given the inherent language
barriers, etc.
___
-- Bandwidth and
the IAX quality is at best 40-50% of a SIP connection.
How is this calculated?
Thanks
Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, October 20, 2009 4:46 PM
To: 'Asterisk Users Mailing List - Non-C
You probably want to set the option
CURLOPT_SSL_VERIFYPEER to FALSE.
Especially with chained certificates (cheapos from godaddy, etc), I have had
lots of trouble with CURL being able to validate a cert. That's probably
because I didn't tell it where the root certs were... but either way.
-Dave
My messages go through rather quickly (minutes).
Unless the lists.digium.com server is running on an Atari, it's probably NOT an
overload issue...
-Dave
Are there any plans to beef up the mailing list server so that messages
can get through with less of a delay?
_
terisk-users] Call File Channel
If you use a Local channel to dial it then it will fall under the same rules
Channel: Local/numbertod...@the-context-you-want
This gets a CDR produced, it does pay to check everything works the same
but it should be fine
Cheers Duncan
David Gibbons wrote:
>
>
context. The context can then
dial out as you write it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 4:10 PM
To: 'Asterisk Users Mailing List - Non
SIP/trunk_y/#2&ZAP/g1/#3,60)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [as
I know I'm missing something here (been a long day)...
How can I specify more than one channel in a call file?
I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1...
Thanks
Dave
___
-- Bandwidth and Colocation Provided by http://www.a
I fail to see how Obama has ANYTHING to do with this.
Danny, please DO elaborate so that I don't have to go on believing that you're
a fool.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nichol
I am using the phones quite successfully, though I have not tried non-English
menus.
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, August 12, 2009 12:33 AM
To: Asterisk Users Mailing List - Non-Commerci
is for the call to go to only one of the unused lines and
then fall straight through to voicemail after the timeout.
Anyone have some thoughts on getting it to work that way?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digiu
Yes each extension needs to be configured separately in the cisco CNF file.
I use a distinct extension on each phone (2 phones can't register to one
'extension' afaik) and ring them in order:
1,1,Dial(SIP/xx)
1,n,Dial(SIP/xx1)
1,n,Dial(SIP/xx2)
Or ring them at the same time:
1,1,Dial(SIP/xx&SIP
How about a shell script on the monitoring server:
#!/bin/sh
trunk=`ssh aster...@astbox "asterisk -r -x 'sip show registry' | grep USERNAME"`
state=`echo $trunk | awk '{print $4}'
if state is 'Registered', yay!
else, UHOH!
EOF
Based on that ssh/shell script framework (you'd obviously need hos
I was having the same problem with about half of my POTS lines.
I switched the polarity on the connections for those lines and the problem
disappeared.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C
This may be a stupid question, but IS THERE a message waiting against your PSTN
lines?
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, August 04, 2009 1:33 PM
To: asterisk-users@list
Mike,
1. Remove the 'line 2' entries completely from the SEPXX.XML file.
2. Change the 'Version' tag in the SEPXX.XML file. You need only change
one digit; I usually just increment the last digit.
(1.0.0.0-9).
3. Restart the phone (Settings -> **#**).
4. This should do it. If
ot retrieved from my 79x1 devices and I
> had
> to specify the firmware version in each SEP file. I am using 8-4-4S, but
> for you this would be something like this:
>
>
>
> SIP41.8-0-2SR1S
>
>
>
>
> And you shouldn't need the tlv file.
>
> -J
I've found that different types of TFTP servers return differing errors when a
file doesn't exist. You don't need the TLV file, but you do need a distro that
tells the phone it's not there correctly. I have not had ANY luck with windows
tftp servers, only linux.
-Dave
-Original Message
ists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug
2mb is small potatoes... unless you mean MegaBytes ins
from the PC LAN physically (2
separate switches), by design. So theres no web browsing etc on that 2mb
circuit.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'As
2mb is small potatoes... unless you mean MegaBytes instead of Megabits...
I am assuming you've already implemented QOS? That is likely the problem if the
intermittent quality issue is only on calls between internal and external
parties.
If someone tries to access the yahoo homepage while someon
Danny,
Just out of curiosity, can you elaborate? Anything in use for asterisk should
be in cache by the time it's needed for a SIP stream. And nothing related to a
SIP stream should ever be read directly from the disk...
Unless I'm mistaken.
Thanks
Dave
Since this is internal SIP, I'd proba
mediately by reply e-mail or telephone and then delete this message,
including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY
14225 USA.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Beh
Assuming you mean the firewall in front of the client, you don't need to open
any ports as long as the VPN client is tunneling all traffic to and from the
Asterisk server.
I set NAT=yes in the config file for the extensions behind a VPN.
-Dave
From: asterisk-users-boun...@lists.digium.com
[m
Cory,
Precisely what do you mean by 'Anything other than Callmanager will essentially
be a "hack"'?
I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP
image. They're not 'hacked', they're set up properly against the Cisco provided
SIP image and are rock-solid stable.
then delete this message,
including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY
14225 USA.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Tuesday, May 26, 2009 10
I could be wrong but I don't think the cat5 limit of 100 meters applies to any
analog signaling over that copper. I believe it only applies to Ethernet
signaling.
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Se
Tunnel samba or nfs through ssh, rather than using sshfs, then mount using once
of those more ubiquitous standards.
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent: Wednesday, May 13, 2009 1:09 PM
To: Asterisk
Redirect traffic with iptables like this:
Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to
NEW_PUBLIC_IP
I'm not sure if this will work for SIP. You may need the proxy to change info
in the sip messages between server and client.
--Dave
From: asterisk-users-boun...@lists.d
Of course, that's assuming your satellite is in geosynchronous orbit. If
It's in LEO...
Singer,
You are of course correct, low earth orbit will have lower latency. I was
assuming that this user would be using a stationary link on the ground, not a
portable sat phone or an aimable dish to make
...routing via satellite adds about a quarter second of latency to the path.
Is that too much?
Eric,
I believe that you are mistaken. Routing via satellite adds about a quarter
second of latency PER TRIP from earth to orbit. This is simply due to the
distance a satellite is from the ground
Yes, you can flash them back and forth as you require.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Shauger
Sent: Monday, May 04, 2009 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
And don't top post ;)
On 3 Apr 2009, at 14:38, David Gibbons wrote:
> The fact that you sent this again (what is that -- 3 times now?) AND
> with high importance, will likely cause
The fact that you sent this again (what is that -- 3 times now?) AND with high
importance, will likely cause people to ignore your messages rather than trying
to help you.
There are few things that annoy me more than messages sent with high importance
(same category of annoyance as messages wri
I had a similar situation a while ago and the fix was setting up
indications.conf:
http://www.voip-info.org/wiki-Asterisk+config+indications.conf
-Dave
I configured a SIP registration with my SIP provider (SIPCall).
Everything works fine except the ring tone for the caller. The caller
hears si
This is the outgoing callerid. If you have 1200 DIDs in a range, you probably
only need to outpulse 4 digits (they already know the first six). If you want
to be able to make your callerid anything that may or may not be one of your
DIDs, you probably want 7 or 10. I pick 10 no matter what for t
Harry,
Chill on the duplicate posts. Sometimes the listserv takes time to forward the
message.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Harry Vangberg
Sent: Thursday, March 26, 2009 3:25 PM
To:
I have several Dell boxes running onboard Broadcom and Intel NICs any haven't
had any issues. It's preposterous to make a blanket statement like that about
all Dell hardware.
Maybe you should re-compile your drivers. Or have prosupport come put a new
mobo in for you :).
-Dave
Not at all, jus
Is this a question?
Haha.
"Computer won't doesn't turn on. Got blck scrn."
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chuck Coleman
Sent: Wednesday, February 25, 2009 3:11 PM
To: asterisk-users@lists.digium.com
Subject: [asteris
You could use the XML browser on the cisco 79xx series.
--Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Chamberlain
Sent: Thursday, February 19, 2009 2:18 PM
To: Asterisk Users Mailing List - Non-C
We will be testing the ADT connection heavily this week. The modem
connections to my understanding are 2400 baud. Over G.711U and a T1 I
don't see why this wouldn't be as solid as a POTS line, but our tests will
tell!
We do *fax* in this way and it works like a charm. We can hit much more than
Certainly a sobering thought. Have others had to deal with this in PBX
replacement scenarios? Its a giant cost savings in this case - they are
dropping about 12 POTS lines in favor of utilizing (an underutilized) T1
trunk that was already in place.
Yes -- our alarm monitoring company considers
On a similar subject, I have been able to get a 7961 to switch to a SIP
firmware, has anyone had any luck with this?
Yes, I have several 7961s and 7971s running SIP, same firmware generation as
the 41s
--Dave
___
-- Bandwidth and Colocation Provided
I've got SIP load SIP41.8-4-1S running w/o problems in a stable environment.
I'll provide SEP.cnf.xml's if requested off-list.
--Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Friday,
Anyone use CIsco 1760 with Asterisk
No, but I'm using 7941G-GE and 7961G-GE in a deployment of ~80 phones. Did you
have a question about implementation or are you just curious?
--Dave
___
-- Bandwidth and Colocation Provided by http://www.api-d
Problem is that its crashing for seemingly no reason at all, no errors
on the console, no logs (that I can find), nothing in /var/lib/messages
- its puzzeling! Management is screaming like banshees, calls are
dropping like flies, and all hell is about to break loose if I can't
stop asterisk from c
- Non-Commercial Discussion'; David Gibbons
Subject: RE: [asterisk-users] Sending Calls via SIP trunk from two different IP
addresses from same Asterisk Machine
My provider has one IP and one port ONLY, I need to send for him the calls from
different IP's on the same Asterisk machine
If your provider has two different IP addresses at its endpoint, you could use
iproute2 (source based routing) with two local source addresses to make sure
that there is a one-to-one mapping of source address to destination address.
Then you could have two peer definitions and an address=declara
Which firmware load? We had all kinds of trouble with 8.4.x, after being stable
for a few months on 8.3.x. Going back to 8.3.x made all of the weirdness
disappear. While we're on the cisco note, I have script to remotely reboot the
SIP firmware load Ciscos and to provision the phones based on a
How pompous are we now?
What happened to the 'open source community'?
There's a give and take involved; you answer questions you know how to answer
in the hopes that someone with greater experience and knowledge of the software
will answer your questions.
Yikes.
-Original Message-
Fro
The higher you raise the barrier for entry to the mailing list, the more you
decrease the amount good the mailing list is actually capable of doing.
(barrier height is inversely related to how much help we can provide to the
people that need help the most)
I agree with you regarding the subject
One problem to overcome is that your competitors are:
1) Literate.
2) Post to the right mailing lists.
Meftah Tayeb wrote:
Ha ha ha ha.
So, you're saying you don't want the job?
LOL.
___
-- Bandwidth and Colocation Provided by http://www.api-digi
I'd be willing to bet *TWO* pennies that you're correct. I certainly was not
coming into the conversation as an expert, just stating what I'd read/heard of
their service... hence the "My understanding is that..." beginning to the
email. :-)
Fair enough. I get worked up when I hear the cable c
My understanding is that Charter 'telephone' doesn't use IP at all but
rather uses some additional frequency spectrum on their cable network.
Hence, the reason why faxing with their service is reliable unlike other
providers who are *actually* using VoIP.
I think what you're referring to is the
I'm confused as to why you think leaving a phone off the hook is better than
parking the call and hanging up the phone. The phone that's off the hook can't
receive any more calls after you've 'pulled' the one it was on the line with,
assuming you don't walk back to that phone and subsequently ha
Ken,
An empty conference call or a parking lot with MOHMP3 both come to mind.
--Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Thursday, January 08, 2009 4:15 PM
To: asterisk-users@
Last I checked, Lynch mobs don't shoot people.
I wonder if there would be interest in organizing a bounty for a lynching
mob, that would track down these !...@#$# silly excuses for human beings and
shoot them. If we all chipped in a few dollars I bet we could hire
someone.
--Dave
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
According to lists.digium.com/pipermail/asterisk-dev/2006-March/019516.html
the value is stored in ${PARKEDAT}
*grin*
I guess I deserved that.
Thanks for checking.
Dave
__
Hello,
When I execute parkandannounce() in the dialplan, is the extension that the
call is parked to stored in a variable? I would like to send it to an AGI
script but can't seem to figure out where the 'announcer' gets its information.
Thanks
Dave
_
How about a call queue using the roundrobin strategy?
http://www.voip-info.org/wiki/view/Asterisk+call+queues
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christophorus
Laube
Sent: Friday, November 14, 2008 11:29 AM
To: asterisk-users@lists.digi
on't be any spam.
David Gibbons wrote:
> I'm glad I'm not the only one who got that. I sent them a nasty response
> earlier this morning...
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
> Se
I'm glad I'm not the only one who got that. I sent them a nasty response
earlier this morning...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
Sent: Thursday, November 06, 2008 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Di
1 - 100 of 119 matches
Mail list logo
