Robin,
Thanks for the viddler.com suggestion! I'm uploading all of the ClueCon
videos to it right now.
John, so far I'd have to give viddler.com two thumbs up. I'm adding my stuff
here:
http://www.viddler.com/explore/cluecon
Your ClueCon presentation should show up some time on Friday. I've
Hi Folks,
I just wanted to share with you all some information about two
well-respected members of the OSS telephony community who will both be
speaking this year at ClueCon http://www.cluecon.com. Their topics are
relevant to Asterisk users so I felt compelled to let everyone know about
them.
Date: Fri, 21 Nov 2008 16:20:28 -0600
From: Terry Wilson [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Large Asterisk installarions (~10,
000
extensions), preferably at universities
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
If you would point me, i would gladly take a look at this patent list,
for now my searches were unsuccessful.
The ITU maintains a list of IPR (Intellectual Property Rights) claims
for various technologies. Check it out:
http://www.itu.int/ipr/IPRSearch.aspx?iprtype=PS
On the left-hand side
I wonder if they've got patents on various strains of Anthrax...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Wednesday, October 01, 2008 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
To those running call centers I have a question: what kinds of soft
phones, if any, do you use? I'm wondering what is out there that has
some hooks for custom applications or host system integration, etc.
OTOH, do you prefer a desk phone for any reason? If so, why?
Thanks for your thoughts,
Gives us legitimate telemarketers a bad damn name. :-)
Isn't legitimate telemarketers an oxymoron?
-MC
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You should expect that; in fact, that's what the 'TB' in 'TBCT' stands
for... for a time, there are two B-channels involved. TBCT is a method
of taking two existing already connected B-channels and linking them
together into the network, it is not a 'transfer' facility where you
provide a
Try GM Voices. $6.95 per prompt plus $175 studio setup fee. To make it
truly cost effective it might be worth it to find other users who need
prompts recorded and then you can split the setup fee. Even if you have
dozens or hundreds of prompts the fee is what is. I think they charge a
separate
Hmm...
You may be in one of those positions where there just isn't a great
solution because your environment has so many constraints. You might
want to check out the way freeswitch handles IVRs, dialplan hooks,
FAGI-ish connections, etc. It will still take some work, of course,
because there
Agreed. It looks like you've tried to tell the Avaya to be the network
side but it doesn't seem to be acting like the network. Do what Steve
suggested and see if you get a different result...
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
You'll probably need to turn on pri debugging for this span and then
capture the output from when you connect the T1 cable. That might yield
some clues, like whether or not any activity is happening on the
d-channel and if so, if there are any errors that might tell you what's
going on.
-MC
I can't speak to exactly what the alarm status stuff does if the port
you're looping expects to have a PRI plugged into it: I would expect
Green, but no actual traffic, but I could be wrong, I'm a bit new on
that front.
Just for the record, this is generally a correct statement. I can't
Ok, I''ll bite. The question is:
Do we want asterisk to contain a licensing engine ?
That depends on the implementation. Your questions, I'm sure, will be
discussed on the call tomorrow.
Such an engine would need to :
Hand out license tokens to proprietary modules linked to
asterisk
Gentlemen,
Dean Collins alerted me to this thread which I had skipped over.
(Thanks, Dean.) I thought I'd offer my viewpoint on the matter; please
take it for what it is - just another opinion, although I hope it is an
informed one. From my personal experience with buying software,
Gang,
I know some of you like to keep up-to-date on various VoIP-ish
happenings. Here's an interesting little article about FreeSWITCH that
also mentions Asterisk:
http://digg.com/software/Freeswitch_Poised_to_Shake_Up_the_Open_Source_V
oIP_Scene
The author guesstimates that Asterisk has
Question: is anyone planning on going to the Cluecon convention this
year? (www.cluecon.com http://www.cluecon.com/ ) I'm hoping to go
this year and I'm hoping to meet other OSS telephony users and
developers. BTW, Anthony Minessale said that there is a need for
Asterisk speakers, so if you're
John
You have raised few valid points. Thanks.
However, I will say that it is not asterisk but people/company
deploying
it. Generally speaking after deployment, and as long users are using
the system normally, no reboot is required.
And yes, running the whole thing from standard PC
Is there a minimum zaptel and libpri version for use with 1.6-beta1?
Thanks,
MC
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what is the difference between FreeSwitch and Asterisk ,
The main difference in functionality is that FreeSwitch is a
voip-switch
only.
Technically, FreeSWITCH is a soft-switch, or a modular media switching
library that can switch more than just voice. Also, technically, FS is
a library,
Is there a reason it resets? Aka does it serve any kind of purpose?
Just curious: what protocol variant (i.e. 4/5ESS, DMS, NI2, etc.) are
you using? Also, which carrier? Finally, have you turned on PRI
debugging to see if it is the telco that is requesting the restart? In
some cases the telco
Is there anyone out there who has tried to connect up an asterisk box
to
make and take calls through a NEC NEAX 2400 using Q.sig or anything
like
it? Can anyone tell me if it is possible?
Phil,
I've successfully connected my NEAX 2400 to Asterisk using line side and
trunk side T1's. I've
Gang,
I recall several months ago that there was a company that was giving
away a free 1-port T1 card, with some specific conditions. Do any of
you recall who that was? My Google searches are coming up empty and now
I'm wondering if I was hallucinating...
Thanks,
MC
http://www.pikatechnologies.com/
--
Kristian Kielhofner
Thanks, I guess I wasn't hallucinating!
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I would have thought an LGPL version wouldn't be out of the question.
I hope not! LGPL is perfect for library-ish FOSS. Releasing libraries
under standard GPL, while making Richard Stallman's heart go
pitter-patter, limits what they can do since they can only go into other
GPL projects.
If anybody thinks they have a magic spell that will calm down the
CDR's, I will not mind the information at all!
Murf,
I don't know if it's relevant or not, but I do know that at least one
legacy PBX vendor (NEC) has a 'solution' that helps with some of the
sillier CDR's that could get
I just got the 2nd edition Asterisk book from O'Reilly, and was
surprised
to find nothing in there about AEL, except a mention of extensions.ael
on
page 471.
This is too bad. A preliminary chapter, an intro into AEL, why it's
valuable, etc. would have been very welcome. Even an appendix of
You know, you don't have to wait for the 3rd edition... you could
always
write something yourself and post it on the web, or join the (mostly
dormant, unfortunately) Asterisk Documentation Project. :-)
-Jared
Well, I could, if I _could_! :) I was hoping to learn AEL from the new
book... my
I'm a complete newbie to Asterisk and have been reading through
documentation and sites for the last couple of hours trying to
understand
what to do to start my learning curve with Asterisk, and am very
confused.
It's a big world, so take a deep breath and don't worry about being
overwhelmed
I also think this is a
positive thing for the Asterisk community as well, as key pieces of
the
Switchvox system will be rolled into the open-source version of
Asterisk.
(I've personally heard of two or three things that the Switchvox team
has
done to improve Asterisk, and I'm sure there are
www.freeswitch.org http://www.freeswitch.org/
(still in early beta)
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Quitoriano
Sent: Friday, August 24, 2007 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Not sure what all the licensing in TrixBox is but if they dump the
open, can't we always just fork. I have not played with TrixBox in
some time but most of it was just a bunch of separate, valuable
projects
meshed together. I don't really see how they can close that.
Yes! That's one of the
Perhaps. I'm interested in knowing what this is all about. Hopefully
it's just Fonality trying to create a new revenue stream, kinda like
Digium did w/ ABE. I'd hate to see them dump the open part of their
community. That community is very valuable for beta testing, giving
feedback,
This freaked me out at first also, but it is totally normal, and is a
good way to let the telco know that your equipment is 'still alive and
kicking'...
-MC
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Nickerson
Sent: Wednesday, August 08, 2007 7:41 AM
To:
I think its a fair decision . 1.2 is very stable and they are not
closing it all together , security issues will still be fixed . They
need to concentrate more on 1.4 to make it bugfree .
Fair indeed. I would guess that a completely stable 1.2 w/ security
maintenance is acceptable to the
Mike,
First, what kind of T1 card(s) do you have? (Just curious.)
I've seen two different theories of operation, although I have
experience only with one, and that's not with Asterisk. One is a
passive tap, the other is a pass-through. I can't say that I know if
they work or how, but
Except that for some users 1.2.18 is NOT stable. I've had to roll
back
to 1.2.15 on my production servers in order to prevent core dumps at
least once per day. No, I am not willing to turn my production
servers
into testing servers to solve this. Doing so would make me a former
consultant
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Gommidh Riadh
Sent: Wednesday, May 23, 2007 3:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fax detection
Gommidh Riadh wrote:
For exemple
I call
.html
He seems to have the skills :)
On Thu, 3 May 2007, Jorge Mendoza wrote:
http://www.gl.com/laptopt1.html
Jorge
Michael Collins wrote:
Why? There used to be a saying 'usb is for mice, firewire is for
men',
though USB has grown a bit in bandwidth since then, it is still
Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not
very
well suited for a high sustained bandwidth. NOw T1/E1 is not that big,
I
suspect a lack of demand. Havng a E1 termintae in your laptop is quite
I can well understand the idea of having USB T1 adapters since that
way
you can colocate 1U Asterisk systems ;-) which at least doubles you
density in a rack...
Frank
I'm glad I asked the question! I was just thinking to myself that it
would be cool to have a USB T1 adapter so that I
http://www.gl.com/laptopt1.html
That's the first item I found when I did a Google search. It prompted
me to ask the question - is there something more generic than this? I
was quoted a price of US$8000 for this, which is way more than I'm
willing to pay for an item which would be used for
Maybe we could interest the guy thats building his own open telco
hardware:
http://www.rowetel.com/ucasterisk/pr1.html
He seems to have the skills :)
I'm working on it right now! :)
-MC
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How about PCMCIA and 2 T1/E1/J1 interfaces?
http://www.utelsystems.com/instruments/hardware/pist-2mp-pro.php
Nice, but less portable than a USB - most desktops and servers don't
have a PCMCIA slot. I'm thinking about the 'U' in USB. If I'm going to
have something be portable, why not make
Just curious: has anyone seen or heard about a USB-based T1/E1 interface
device? I've seen some serious T1/E1 testing equipment that is
USB-based, but I was wondering if there was something more generic, like
a Zaptel-ish T1/E1 that used USB instead of PCI/PCIx.
Thanks!
-MC
On Wednesday 11 April 2007 03:11:38 pm Brandon Kruse wrote:
I wrote a very extensive plugin for cacti to monitor asterisk.
It uses the manager interface to poll and get statistics for 1.4 and
1.2.
Let me know if you interested, ill post it, or email me directly.
-bkruse
I did
Salvatore Giudice wrote:
Take a look at this patent:
http://www.freepatentsonline.com/20060098624.html
Title: Using session initiation protocol
Document Type and Number:United States Patent 20060098624
Inventors:
Morgan, David P. (Lexington, MA, US)
Sullivan, Daniel B.
Jordan,
I don't know if you've down this step before, but my network admin sent
me these instructions a few months ago. It allows you to tell your
Exchange Server's SMTP to allow relays from specific domains, hosts, or
subnets. Hope it helps. (Works for Exch 2000 and 2003.)
-MC
1. Go to
span=1,0,0,esf,b8zs,crc4
This needs to be span=1,1,0,esf,b8zs
I'm not sure if the crc4 is necessary.
Doug
I concur with Doug. I have two PRI's in one system. My zaptel.conf
looks like this:
span=1,1,0,esf,b8zs # PRI line - LD Qwest (interstate)
bchan=1-23
dchan=24
Looks like user interface is not a concern - if they are thinking of
FTP
text files. In this case, a simple script to kick off some call files
should suffice. Won't take a week. (Search for call file.) But
having to
deal with answering machines is always tricky for any automation.
Yuan
I've never seen a PRI dchannel on a T1 on a timeslot other than the
24th. Are you sure that it's really on channel 23?
I think he meant channel 23 of channels 0~23, aka the 24th channel.
-MC
Matthew Fredrickson
On Mar 20, 2007, at 8:54 AM, Kanelbullar wrote:
Thanks for your answer,
I would suggest that we create a new wiki, make it solely for Asterisk
topics, as not to offend or replace voip-info. Build mirrors to
multiple sites and multiple domain names. This would give this
community a second resource with redundancy which is what I think ALL
of
us are looking for.
I am using the * auto-dial out feature but don't want to have to
specify
a channel (Zap/G2/) to connect to the extension.
Current file I use:
Channel: Zap/G2/12127778866 # I have to specify a specific
channel
MaxRetries: 1
RetryTime: 60
WaitTime: 30
#
# Assuming that your
Hi everybody,
i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of
TE110P
and also if you can tell me have to made a cable like that??
Modem Teleco ---Self CrosscableAsterisk
You might check this out for a quick reference:
Hello,
i've installed trixbox with TE110P TDM400B, but no led is ON in the
TE110P, i don't know why even if the 4 leds of My TDM are greens
any explaination
Thank You
No LEDs on TE110P (and similar cards) can sometimes mean that the Zaptel
driver isn't running. Can you run zttool and see
Yeah, it's hard to know what it would be filed under. However, if you
use zap trunks then you'll want to know about this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+Channels
BTW, see Dialing a Group for specifics on 'g' vs. 'G' as well as other
cool stuff.
-MC
/13/07, Michael Collins [EMAIL PROTECTED] wrote:
Yeah, it's hard to know what it would be filed under. However, if you
use zap trunks then you'll want to know about this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+Channels
BTW, see Dialing a Group for specifics on 'g' vs
Of course, you should take this with a grain of salt since I tried [EMAIL
PROTECTED]
(now TrixBox) for a total of 2 weeks before gutting it. Now, I just
use
my own GUI for everything from graphical setup to scripting.
There is nothing wrong with starting out with Trixbox. I still use
.
-Brian
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Collins
Sent: Monday, February 05, 2007 4:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Using Local Channels with Originate
I haven't quite gotten
I haven't quite gotten this working yet but I am going to update the
thread with what I have learned. Maybe this will help the next guy who
tries to figure this out...
The trick to using the DIALSTATUS seems to be to put it in the handler
for the h (hang-up extension).
[outdialer]
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Collins
Sent: Thursday, February 01, 2007 12:38 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial option G - Passing parameters?
Has anyone used the G
I have been having a very similar problem. Has anyone here gotten a
DIALSTATUS for calls started with originate?
I did some research and saw some posts that local channels are the
solution
to this problem. However, I could not find examples of how to use
local
channels with originate. I
Has anyone used the G option with the Dial app? I'm looking for a way
to control the called party leg. Specifically, I'd like to pass a few
variables to the called side for some call control. Here's a synopsis
of what I'm doing:
Make outbound call w/ AMI Originate action.
Called party answers
I've discovered that when dialing out using API's Originate action, a no
answer is considered a failed attempt, while a busy is considered a
successful attempt. The problem I'm having is that when I dial an
invalid number, say a disconnected number that gives a fast busy, my
CDRs are identical to
NoAnswer and Invalid phone number
On 2/1/07, Michael Collins [EMAIL PROTECTED] wrote:
Is there a way to distinguish between a no answer and an invalid? For
me, a 'failed' attempt is dialing an invalid number, and I'd like the
CDRs to reflect that. I'd like a no answer to be as 'successful' as a
busy
ESF
B8ZF
Inbound = EM Immediate
Outbound sig =Wink Start
Yield to Glare = Yes
In zaptel.conf, when having something like
span=5,0,0,cas,b8zs
and in zapata-channels something like
signalling=featb
try
em_w: E M Wink Start
Jerry is right - you need to set signaling in
Can Asterisk support vxml?
Can i work with Asterisk and vxml?
Is there any AGI framework that can use vxml?
It seems like support is still a bit limited, but evidently it is
available:
http://www.voip-info.org/wiki/view/VoiceXML
HtH,
MC
___
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Halliday
Sent: Wednesday, January 24, 2007 11:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] setting up AMD
I'm trying get this working. I've looked through the list, and can't
see how
] app_amd.c: Got hangup
Jan 24 18:53:36 DEBUG[31567] pbx.c: Extension s, priority 2 returned
normally even though call was hung up
Jan 24 18:53:36 DEBUG[31567] chan_sip.c: update_call_counter(6079362172)
- decrement call limit counter
On 1/24/07, Michael Collins mailto:[EMAIL PROTECTED]
[EMAIL
The correct way to determine the ending cause of a call is the
${HANGUPCAUSE} variable that Dial creats. Just to be sure, set
priindication=outofband in /etc/asterisk/zapata.conf. HANGUPCAUSE
should always be set.
HANGUPCAUSE is indeed always set. The question is, Set with what data?
The
original message
I am trying to automatically detect disconnected numbers when using the
outbound dialer I have written.
* Some numbers hang up immediately with a Cause Code 0 and no voice treatment
* Some numbers get voice treatment with a PROGRESS indication and an associated
Cause Code 0
One thing that is really confusing me at this point: if I want to
leave
an automated answer machine message, and amd tells me it's a machine,
how do I know when to start leaving the message ? Some intros are
long
(thanks for calling, me and mine are not here right now, please leave
a
message
I've got a curious one: all of a sudden my .call files and my manager
API 'Originate' actions are no longer producing a CSV file. The call
still generates just fine, and Master.csv is updated. However, I don't
get the usual CSV file in the form of xx.csv where xx=account
number.
I
I've got a curious one: all of a sudden my .call files and my manager
API 'Originate' actions are no longer producing a CSV file. The call
still generates just fine, and Master.csv is updated. However, I
don't
get the usual CSV file in the form of xx.csv where xx=account
number.
David is correct: there are several issues to resolve. Some common T1
settings in the USA are:
Framing: ESF
Line coding: B8ZS
These are very common settings.
Now you'll need to find out what the signaling type is. No point trying
to guess - they vary greatly. If you can find out how the
Also, anyone have suggestion on licensing? LGPL? FreeBSD?
One advantage of LGPL over GPL is that GPL is 'viral' whereas LGPL is not. For
a more in depth discussion please see:
http://www.ugcs.caltech.edu/manuals/devtool/autotoolset-0.11.4/toolsmanual_87.html
In short, if you want anyone to
I believe I am going to start out with some refurbished Dell Poweredge
servers. They have had a high success rate with a friend.
One word of caution: some have had various hardware issues getting
certain telephony cards to work with certain Dell PowerEdge servers. If
you aren't going to have
Question: I'm trying to put a double quote into the CDRUserField. What
I end up with is a pair of double quotes. Example:
exten = s,n,SetCDRUserField(data)
exten = s,n,AppendCDRUserField()
exten = s,n,AppendCDRUserField(moredata)
My record will look like this:
datamoredata
What I want is:
?
Michael Collins wrote:
Question: I'm trying to put a double quote into the CDRUserField.
What
I end up with is a pair of double quotes. Example:
exten = s,n,SetCDRUserField(data)
exten = s,n,AppendCDRUserField()
exten = s,n,AppendCDRUserField(moredata)
My record will look like
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Richard Lyman
Sent: Saturday, December 30, 2006 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialed Number missing from the CDR
you need to add
;this extension MUST be here for OriginateFailure triggers
exten = failed,1,Hangup
to your context used for *send too after connect*
Richard,
THANK YOU!! This makes a lot of sense - I don't know why I didn't catch
that before. I can add my SetCDRUserField stuff in the
I think the CDR generator of the Asterisk
needs change to record the complete information.
Agreed. However, there are still challenges here. First, you could use
the custom_csv to create your own CDR layout that includes the dialed
number, but you'd still need to come up with a way to get
The CDR, both the csv file and in MySQL does not contain the dialed
number (src) in case of a call placed using .call files.
Is this is Bug ? The cdr should have complete info, what ever the
source
or method of the call.
I have found this same problem and have not found a solution within
But apart from that: have you tried at least building that driver with
1.4.0 ?
Yep. The build process seems to work just fine. The ztcfg and zttool
stuff all acts normal. I copied tor2.c and tor2-hw.h from the custom
1.4.0-beta1 drivers (that work just fine with asterisk 1.4.0-beta1) and
Has anyone else installed the official 1.4.0 release? I have, and it
installed very easily. However, I don't have any of my usual command
line tools for monitoring and debugging zap channels and PRI lines:
asterisk1*CLI pri show span 1
No such command 'pri show' (type 'help' for help)
You must use zaptel 1.4 and libpri 1.4. Asterisk 1.4 specifically has
checks in the configure script to check for the unique stuff in those
versions and the associated channel driver (chan_zap) will not build
without it.
I think I found the issue. My Tor2 clone has a modified driver. The
Does anyone know what the time values in amd.conf are? Are they
seconds,
fractions of seconds, heartbeats, what?
Milliseconds.
;'initialSilence' is the maximum silence duration before the greeting
initial_silence = 25; Maximum silence duration before the
greeting.
It doesn't say
Firstly, in the setup you are envisaging, how do you distinguish
which
company the caller is calling from? Their extensions number?
The context
at which they enter the dialplan? Or something else?
Good questions, all of them. Unfortnately, I don't have answers to
them. I
wanted to take
I am configuring two cards in Trixbox. 1 TE110P T-1 card and one
TDM2400P
with 16 fxs ports (All 24 show up in zaptel.conf so the PRI channels
start
at 25). Can I use a channel range to separate the config for each
card, as
shown below, or do I have to enter configs for each channel?
Also
Has anyone done any fax machine detection on outbound calls? I've heard
of NV's fax detect app but I haven't seen any indications that it
supports outbound fax machine detection.
-MC
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Gang,
I'm wondering if anyone has run into this problem and found a solution.
When I use the manager interface to generate a call, I don't get very
much information in my CDR records when the dial status is BUSY, FAILED,
NOANSWER, etc. I am putting the dialed number into the CDR Userfield in
The manager interface expects Exten NOT Extension argument header.
Well honk my hooter!
I had been using 'Extension' but since I always used the 's' extension I
never noticed anything goofy until I tried a numeric extension. Thanks
for the heads up.
-MC
Question:
I'm using a .call file to make some test calls. The call file works
great. When I try the same thing with the manager 'originate' action I
get something weird - the originate action looks for the 's' extension
in my context, regardless of what I supply as the 'extension' argument.
The
Has anyone had experience with one or both of these cards? I'm in a
position where I might need to recommend one over the other. I've read
everything that I can find online, so now I'd like to hear of personal
experiences. Everything I read on both cards is 5 stars! Awesome! It
Rocks! They
I want to know how to get the uniqueid or a call started from asterisk
manager using Originate command.
Are you wanting the uniqueid for the call right after it is started,
i.e., while it is still in progress? What is in your Dial command?
-MC
___
There is no dial command, I'm sending originate action from asterisk
manager.
Oops, I didn't ask my question correctly. You're right, it isn't a
dial command. What I wanted to know was the contents of your
originate action, e.g.:
Channel= 'zap/g0/' . $dialed_num
(From one of my Perl
New Asterisk user, wondering if anybody has connected an Asterisk box
to
an NEC Aspire S? We're in the beginning processes of attempting this,
we'd like to have the Asterisk box connected as an extension off of
the
NEC box, wondering about the wiring and settings/programming needed to
get
On the NEC, digital stations (ip1na-12txh)
I am not familiar with the Aspire, but if it is even remotely like the
2400 then you might be able to get a jumpstart using my 2400 how-to:
http://www.voip-info.org/wiki/index.php?page=Asterisk+NEAX2400
It deals with getting a Tormenta2 clone talking
Great link. After I all you said I get this error loading the module
in
asterisk via load app_swift
The 'load' command is deprecated and will be removed in a future
release. Please use 'module load' instead.
[Nov 30 13:54:08] WARNING[7825]: loader.c:362 load_dynamic_module:
Error
Sometimes the data comes back separated by \r\n, and sometimes it's
separated by \n.
The whole thing is completely inconsistent, and trying to write any
kind
of API for it is -GHASTLY-
Doug,
What language(s) are you using? Just curious. I've been tinkering with
Perl, POE, and
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