Eric Wieling wrote:
The word Dialing... and Calling...
As in Dialing 911, please wait...
and as in Calling 911, please wait...
oooh boy wouldn't I be frustrated if I heard that instead of a ring when
I dialed 911? what else is it gonna tell me?
Philipp Kempgen wrote:
Mojo with Horan Company, LLC schrieb:
Eric Wieling wrote:
The word Dialing... and Calling...
As in Dialing 911, please wait...
and as in Calling 911, please wait...
oooh boy wouldn't I be frustrated if I heard that instead of a ring when
I
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HORAN COMPANY, LLC
403 Lincoln Street, Suite 210
Sitka, AK 99835
(907) 747-
(907) 747-7417 - Fax
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Raúl Gómez C. wrote:
Another silly question,
In the first Digium link posted before there is a line that said *The G.729
codec works with all Digium cards*, but this license will work with a
Sangoma Remora Card??? Or do I need to buy it from Sangoma??? (I don't know
if the are selling G729
Raúl Gómez C. wrote:
LOL!!! Thanks Mojo!
On Sat, Apr 19, 2008 at 12:07 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
The codec in use for a specific channel doesn't even care if that
channel exists over zapata analog or digital cards, sip channels, iax[2]
channels, smoke
until eaten.
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Sitka, AK 99835
(907) 747-
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[EMAIL PROTECTED]
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a conversation was going on
between two people in the room in distinct locations :D
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HORAN COMPANY, LLC
403 Lincoln Street, Suite 210
Sitka, AK 99835
(907) 747-
(907) 747-7417 - Fax
[EMAIL PROTECTED]
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not be
quite right.
Moj
--
*Mojo Wentworth*
HORAN COMPANY, LLC
403 Lincoln Street, Suite 210
Sitka, AK 99835
(907) 747-
(907) 747-7417 - Fax
[EMAIL PROTECTED]
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Raúl Gómez C. wrote:
Hi list,
snip
I think this is a very common scenario so, how are you doing to handle this
situation???
What if you were to set an account code to the extension that is
requesting the long-distance call?
So person at extension 111 requests a long distance call to
J. Oquendo wrote:
Its fine and dandy, but the problem is you're still getting 5 packets.
You're still saturated period. No QoS in the world outside of your
provider and more bandwidth can alleviate that. Your provider is not
going to care what you do once its passed to the CPE. So look at it
J. Oquendo wrote:
it does, when someone can realistically point this out please let me
know so I can switch from a DS3 to T1 and save money.
Use the T1 for voice and get a DSL modem for your data use? :)
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Nestor A. Diaz wrote:
1. I use a queue with just on sip device, one call at a time, however
and without reason just after some couple of hours the sip device show
in use and then no calls are transfered from the queue to the sip
device, i do a sip show inuse and this is the result:asterisk
one,two,or three|1)
exten = s,n,Goto(mainmanu,${pressedbutton},1)
exten = 1,1,blah
exten = 2,1,blah
exten = 3,1,blah
exten = i,1,NoOP(${pressedbutton})
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HORAN COMPANY, LLC
403 Lincoln Street, Suite 210
Sitka, AK 99835
(907) 747-
(907) 747-7417 - Fax
[EMAIL PROTECTED
Mojo with Horan Company, LLC wrote:
[mainmanu]
exten = s,1,Answer()
exten = s,n,Playback(Press 1, 2, or 3)
exten = s,n,Read(pressedbutton|Press one,two,or three|1)
exten = s,n,Goto(mainmanu,${pressedbutton},1)
Oops,
shouldn't have that second priority in there. Because Read is playing
://lists.digium.com/mailman/listinfo/asterisk-users
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Sitka, AK 99835
(907) 747-
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[EMAIL PROTECTED]
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Moj
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(907) 747
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HORAN COMPANY, LLC
403
Yeah, Asterisk I think would be more than capable of doing that. It'll
need some work to glue it all together. A lot of this would be written
as an AGI script, and PHP or so for the webpage part of it.
Sounds fun!
blackwater dev wrote:
We currently have an application used by the trucking
faraz wrote:
FOP is quite clunky!
Also the flash is almost un-usable with a large number of extensions
Would love to see something in PHP/Ajax which could be lightweight and
fast.
Last version of FOP I downloaded had a DHTML client in addition to the
fat Flash client, I'm pretty happy
On 3/25 Justin Newman wrote a message to the list mentioning his
SystemAnnounce application that broadcasts audio to all active channels,
I suspect his code would be easy to modify to broadcast to a single
channel...
Moj
John Hass wrote:
I have a voicemail application that users can listen
sean darcy wrote:
Kevin P. Fleming wrote:
Mojo with Horan Company, LLC wrote:
P.S. If you can't dial seven digit numbers in your area, but you miss
it, you can restore that behavior if you feel like selecting a default
area code:
exten = _NXX,1,Dial(Zap/1/907${EXTEN},,TWK
Kevin P. Fleming wrote:
Mojo with Horan Company, LLC wrote:
P.S. If you can't dial seven digit numbers in your area, but you miss
it, you can restore that behavior if you feel like selecting a default
area code:
exten = _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
Here, if I dial
try dumping your wav file in there :) unavail, greet,
and busy.
Moj
Mark Quitoriano wrote:
On Sat, Mar 29, 2008 at 7:26 AM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
You could save it to your asterisk voicemail directory, which is often
something like:
/var/spool/asterisk
Olivier wrote:
And what about SIP support ?
Should it be removed in 1.6 or 1.8 ?
Where have you been? SIP's been deprecated since 1.2.
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To
Alejandro Cabrera Obed wrote:
Can Asterisk control the RTP open ports the voip clients use ??? Or the
RTP open ports depend on the voip clients ???
It depends on the VoIP clients.
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Steve Davies wrote:
Could you point me at some reference material for how this differs
from KS, and what compatibility issues this might cause with other
equipment? Has anyone tried this in the UK? Would BT even understand
the request for ground-start signalling?
KS (Kewl Start) simply
Steve Edwards wrote:
4) How do YOU find an Iaxy on your network?
I was most easily able to find them by watching my DHCP server logs.
You're right about the -b switch to ping, that's required.
Moj
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Paul Whitby wrote:
Hello
Newbie question here: I have a box running Ubuntu Linux 7.10 gutsy
gibbon, and have a single Digium TDM410E card, with 1 FXO module
fitted and connected to my landline. I have it answering the landline,
directing to SIP phones, diverting to voicemail etc - and
Doug Lytle wrote:
John Meksavan wrote:
level high and still, the same problem. I tried to increase the rxgain
to 12.2 in the zapata.conf file and it had no affect
You'd want to fiddle with the txgain(Transmit)
Doug
He might actually want to deal with rxgain, because it
Sync the clocks on your asterisk boxen using NTP or whatever, and then
'touch' the call files into the future so each asterisk waits before
processing it...? Might get them closer.
Another option is get all three boxes into the same meetme room, waiting
a few seconds for them to be ready if
Mark Quitoriano wrote:
Hi,
I have a wav file recording that i want to use on my voicemail, how
can i set this up?
thanks!
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martin f krafft wrote:
What's going on here? From all I can tell, the clients do the right
thing, each selecting the first codec offered by asterisk (which
they support), but asterisk is going a bit lala here, isn't it
I think Brent's on to it there -- as he suggested, get your allow= and
Sean Dennis wrote:
bilal ghayyad wrote:
Hi All;
I have been chocked just when I saw some posts talking
about how much the IAXy is bad :) -
So I would like to ask, did any one try it later and
wether it is good or not? I am asking this because I
need to use it as it is NAT Transparent
bilal ghayyad wrote:
Hi All;
I need to buy one IAXy device, but I discovered that
it supports only g711 and ADPCM codec, so I was wonder
that it does not support g729 or GSM?!
Anyway, what is that ADPCM and how much it consumes
bandwitdh? Also, asterisk support such codec? What its
name
rid of echo
you must cancel echo.
Mojo with Horan Company, LLC wrote:
Sean Dennis wrote:
bilal ghayyad wrote:
Hi All;
I have been chocked just when I saw some posts talking
about how much the IAXy is bad :) -
So I would like to ask, did any one try it later
Aadilkhan Maniyar wrote:
Hi All,
I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17
and using it to make SIP calls.
I have a configuration of Asterisk which serves the users in a
particular domain, say internal.com
I would like to make a SIP call from [EMAIL PROTECTED]
Guido Hecken wrote:
-Ursprüngliche Nachricht-
Von: Mojo with Horan Company, LLC [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 25. März 2008 23:23
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Asterisk parking hold
Steve Edwards wrote:
On Tue, 25 Mar 2008, Justin Newman wrote:
Does anyone have use for a broadcast/annouce app?
I wrote SystemAnnounce which will play a specified file to all active
channels (in an UP or bridged state). This was originally to tell users
to get off the system, but
Guido Hecken wrote:
Hi,
anyone out there with the same problems and a possible solution to the
following?
The functions callparking and hold use the same transferdigittimeout in
features.conf.
While I think 3 to 5 seconds are enough to let the user find their keys on
the phone,
the
I think what you want is:
originate LOCAL_CHANNEL application dial REMOTE_CHANNEL
some examples:
originate SIP/112 application dial Local/[EMAIL PROTECTED]
originate SIP/112 application dial Local/[EMAIL PROTECTED] ;Echo
Chamber exten
originate SIP/112 application dial ZAP/g1/18005551212
Distinctive Ringing might be available from your telecom provider.
mark morreny wrote:
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to receive calls, but I am not able to
capture the destination number through the ZAP channel
Vincent wrote:
Hello
I run AGI scripts from extensions.conf to save data into an SQLite
database file, but this file must also be accessible in read-write
mode by PHP scripts served by Lighttpd.
As far as I can tell, Asterisk runs by default as root:wheel. I don't
know if AGI scripts
Lee, John (Sydney) wrote:
I am working on a menu to accept input from a caller like as follows:
Exten = 100,1,Answer()
Exten = 100,n,Playback(LONG-MESSAGE)
Exten = 100,n,Read(OPTION,,2)
...
When I tested it, I noticed if I start pressing a key before the
Playback() is finished, the input
Vincent wrote:
On Mon, 24 Mar 2008 11:05:32 -0800, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
?php
$u = posix_getpwuid(posix_getuid());
$g = posix_getgrgid(posix_getgid());
echo This script is running as .$u['name'].:.$g['name'];
?
1. Here's
access to temp.sqlite, you could do what you need to do, /unless/ sqlite
tries to create a temporary file and mv it over the top of temp.sqlite,
as this would require write access in the directory.
Vincent wrote:
On Mon, 24 Mar 2008 12:09:00 -0800, Mojo with Horan Company, LLC
[EMAIL
Rob Hillis wrote:
Distinctive ring is still not going to provide the line that was
called in the ${EXTEN} variable, so you're still stuck with dialplan
trickery to figure out which number was rung.
Mojo with Horan Company, LLC wrote:
Distinctive Ringing might be available from your
Ricardo B. wrote:
Hi all, new to the list and this is probably a basic question and
couldn't find anything clear googling around but I don't know how to
handle calls to sip extensions not defined on sip.conf while using
pattern matching. On my example I have sip extensions 10, 11, 12, and
Zoa wrote:
Mojo with Horan Company, LLC wrote:
Aren't all the frames in asterisk 20ms long, no exceptions?
Isn't ilbc the exception ?
Even though the ilbc codec likes multiples of 50 for its frame size (Is
this right?), I was under the impression that asterisk broke everything
really seen one break yet though. VxWorks is what runs
satellites and junk ;-)
Thanks,
Steve Totaro
On Wed, Mar 19, 2008 at 7:18 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
Steve Totaro wrote:
Anyways, as to the four FXO system, I would not think twice to steer
Mian M Asif wrote:
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
exten = s,n,Set(OLD_EXTEN=${EXTEN})
Then later, just use ${OLD_EXTEN}
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[EMAIL PROTECTED] wrote:
I am planning to write a module to find if a Special Information was detected
or not.
Can anyone please help me to figure out the below fields?
1. The Frequency of a frame
2. Length of frame in milliseconds
Aren't all the frames in asterisk 20ms long, no
An off-the-shelf 5+ year old MSI MS-6378X-L motherboard, 1.6GHz AMD, 512
RAM, 10 extensions, no more than three concurrent calls:
[EMAIL PROTECTED] ~]$ uptime
11:31:45 up 103 days, 1:00, 2 users, load average: 0.00, 0.00, 0.00
But:
[EMAIL PROTECTED] ~]$ sudo asterisk -rx 'core show uptime'
I'm just a user :) we do real estate appraisals, and I found the time
to roll my own (so to speak) pbx. We're on 1.4.4, TDM card with four
FXOs. Honestly, you'll find it's easy to toss some zaptel and asterisk
tarballs onto a system and compile them. You'll probably learn a lot
along the
Steve Totaro wrote:
Anyways, as to the four FXO system, I would not think twice to steer
that customer to the 3Com V3000.
Interesting :) When I (the tech guy) leave this office, they just
*could* be asking me what to do when it breaks? lol :)
___
He could mean SIP or IAX
Al Baker wrote:
Quote
This code is pre-Asterisk 1.0... It processes quite a few calls daily, I
have about 1,800 DID numbers pointed at it,
Are you SURE on that figure. Since you cold have at MOST 4 T1's coming into
that box, 1,800 DIDs pointing to it sems like
It must accept attachments, or we wouldn't get all these HTML messages,
right? I think that's how HTML messages get through is attached :)
Definitely not sure though.
On another note, I heard a rumor a while back that messages over 40k
might be held for moderation?
Moj
Drew Gibson wrote:
I agree, seems odd you didn't have a [peername] section for your
softphone in your sip.conf.
aren't 404 errors a likely symptom of this? :)
Mojo
Steve Totaro wrote:
Pete,
You are connecting via a SIP softphone correct? Where is that in your
sip.conf?
On Mon, Mar 17, 2008 at 11:42 AM,
Like this?
exten = _XNPANXX,1,Dial(Zap/g1/9${EXTEN}|20)
Notice it matches 18005551212 and it dials 918005551212. (The 9 before the
${EXTEN})
Moj
Joshua Kinard wrote:
Hey all,
Working slowly on getting the myriad number of parts to my fax system plan
together, and one of the pieces
http://vitelity.net has 800# DIDs for $0.50/month plus usage (which is
like $0.02/min I think)This price has been very bearable for me to
just experiment with -- I can ask anyone I want to call me to test my
services and they don't have to worry about toll charges
Moj
Mike wrote:
hey
Are you using buttons on your phone to effect the transfer, or are you
using codes defined in features.conf?
Moj
Ian wrote:
Hi,
Mojo with Horan Company, LLC said the following on 20-Feb-08 09:31 PM:
Is it AFTER you have parked a call? Meaning, for example, you transfer
an incoming call
? Our polycom transfer buttons have always just worked,
but my users, for some reason, all felt more comfortable using DTMF
keypresses... dunno why :)
So we all press ## to do a blind transfer now, or ** to auto-park to
first parking space.
Moj
Mojo with Horan Company, LLC wrote:
Are you using
That can be found in the monstrous admin guide for the phone, seemly in
Section 3.1.7 in my ancient version 1.5.0 document. It shows me that on
the 501, that button is 9 instead of 23.
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip301.html
There's a link to the
Delete extensions.ael too, unless you're using AEL instead of the dialplan
Mindaugas Kezys wrote:
We do:
in modules.conf:
noload = pbx_ael.so
noload = pbx_dundi.so
noload = res_config_pgsql.so
noload = res_smdi.so
in extensions.conf delete every context [default], [demo], whatever
in
Tzafrir Cohen wrote:
Delete extensions.ael too, unless you're using AEL instead of the dialplan
extensions.ael is harmless on its own.
It seemed that the default extensions.ael created some demo contexts and
extensions that might befuddle a new user, I could be wrong
Is it AFTER you have parked a call? Meaning, for example, you transfer
an incoming call to 700. No problem. Later, when it's picked up from
701, can it NOT be transferred again?
Moj
Ian wrote:
Hi All
Sorry to be a bother again but seems like I just cant get away from
the problems.
How about a computer with a copy of asterisk at each end?
You'd need good network connectivity between them. A recent post by
Gordon Henderson states that GSM calls can take up to 32K/sec with IP
overhead, less probably if they are trunked into an IAX connection. For
landline quality, Gordon
No problem, hope it gets you where you need to be :)
Moj
Anton Krall wrote:
This is a good start, thx Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: martes, 19 de febrero de 2008 01:35 p.m.
To: Asterisk
Sure, run 10 concurrently and see how it sounds. Scale up by a factor
of 10 until it sounds crappy then start scaling down. shrug At least
I think that's what Atis meant.
Moj
Tzafrir Cohen wrote:
On Wed, Feb 20, 2008 at 11:51:55PM +0200, Atis Lezdins wrote:
Test of audio quality is
Like 15 lines of php and html?
?php
$fn = /etc/asterisk/extensions.conf;
if ($_REQUEST['action'] == write $_REQUEST['contents'] != )
{
rename($fn, $fn...date(U));
$fp = fopen($fn, wt);
fwrite($fp, $_REQUEST['contents']);
fclose($fp);
}
?
form
Jim Duda wrote:
== Spawn extension (incoming-dial, fax, 0) exited non-zero on 'Zap/4-1'
Yes, I DO think that's a little odd. It should be priority 1, shouldn't it.
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There are some tdm400 cards on ebay, http://search.ebay.com/tdm400
Moj
Giorgio Incantalupo wrote:
Hi,
Digium stopped to produce TDM400P and the new TDM410 is too new to find
it in our shops. The only alternative available is a fully-compatible
Openvox product...but is it really
Will Set(MONITOR_FILENAME=/blahblah/filename) work for you?
Moj
Jaap Winius wrote:
Hi list,
The default file name format for touch monitor (automon) recordings is:
auto-${EPOCH}-caller-calee
It's possible to use the ${TOUCH_MONITOR} variable to change the
'caller-calee' part, but
bilal ghayyad wrote:
[channels]
rxgain=15.0
txgain=15.0
Wow! Is this necessary? Is this something you took from a sample
config somewhere, or numbers that you arrived at through trial and
error? They seem a bit high in my experience, *but* I've never been to
Egypt before, and I sure
Tilghman Lesher wrote:
On Monday 11 February 2008 11:55, Mojo with Horan Company, LLC wrote:
William F. Acker WB2FLW +1-303-722-7209 wrote:
Thanks for mentioning contexts. All of us are in the default
context. So I started playing around with the options pertaining
Tilghman Lesher wrote:
On Monday 11 February 2008 11:55, Mojo with Horan Company, LLC wrote:
William F. Acker WB2FLW +1-303-722-7209 wrote:
Thanks for mentioning contexts. All of us are in the default
context. So I started playing around with the options pertaining
William F. Acker WB2FLW +1-303-722-7209 wrote:
Thanks for mentioning contexts. All of us are in the default
context. So I started playing around with the options pertaining to
contexts. I found that if I uncommented searchcontexts=yes, I could send
from inside. The explanation
Don't forget to 1000,1,Answer the call
Moj
John Von Essen wrote:
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
addition being by SIP phone, and my incoming voip. Last
Soumya Kat wrote:
Hi,
I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8
system. Asterisk works fine for me and I can log into Asterisk-GUI and
monitor asterisk.
What I would like to know is how to get information such as SIP users,
number of SIP connections and
After Andrew's suggestion, if that isn't the problem, spend some more
time on OSLEC to be darn sure it's operating properly -- that thing
works like a champ for my crappy lines!
Moj
Brent Davidson wrote:
We're deploying an asterisk-based phone system at all of our branch
offices in an
Jaap Winius wrote:
* Why can't I delete any voicemail messages?
(Response: Message undeleted.)
* Why can't I listen to the messages in the Old folder?
* Why can't I use the advanced options?
(Response: I'm sorry, I did not understand your response.)
* How
rachid wrote:
Hello,
I have some problems to use G722, when my client sent an invite request
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.
I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I activate It??
Thanks.
Rachid.
It's
Mojo with Horan Company, LLC wrote:
So just tell your client to not /ask/ for 8kHz audio.
As Kevin just pointed out, apparently you do NOT have to tell your
client to ask for 8kHz audio. May I ask what client you are using?
Moj
___
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Have you swapped the phones between the FXS ports to see if the phone rings?
Moj
Shane Wegner wrote:
Hello all,
I have two handsets connected to FXS ports on a TDM400P,
both GE models but one rings and the other does not. The
phone models are not identical. The phone which doesn't
ring
Thomas Kenyon wrote:
The server that I will need to get this running on has an 82801EB/ER
(ICH5/ICH5R) AC'97 sound controller (and no expansion space left to put
another card in).
Just a suggestion, don't forget there are USB audio devices available
that work with linux, you may have an
randulo wrote:
On Feb 4, 2008 9:34 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
In my recollection, [EMAIL PROTECTED] worked when I tried it, without sip
or a colon. xxx could be anything at all. I noted this behavior back
in 2006:
http://lists.digium.com/pipermail
randulo wrote:
I have an IP 500 and I have tried everything I can think of to call a
SIP number like this :[EMAIL PROTECTED] without the call trying to go
through the registered servers. I even added an emergency server and
number in the sip.cfg. Dialing the number manually or in the directory
A COMPLETE shot in the dark, but:
Tomasz Zieleniewski wrote:
[Feb 4 09:33:09] == Parsing
'/home/asterisk/asterisk/1.4/pbx/etc/asterisk/manager.conf': [Feb 4
09:33:09] Found
If this is where you've got everything installed, i.e. with a base of
/home/asterisk/asterisk/1.4/pbx/, maybe:
what about astdb? is that too much of a global variable?
moj
Arjan Kroon | Mobillion wrote:
Hi,
I’m using videocalling on asterisk 1.4.10.
When I setup the videocall with exten =
n,1,h324m_gw([EMAIL PROTECTED]), I loose the variable DNID
(${CALLERID(dnid)})
Before the videocall is set
/different
orig/dest channel types
you'll get the events in the reverse order... Nothing much but: i)
you'll have to
track them either way and ii) it reveals that the AMI events
aren't 100% clean!!!
:/
--
exvito
On Feb 1, 2008 12:08 AM, Mojo with Horan Company, LLC
[EMAIL
The snippet is asterisk telling you I'm just letting you know that the
correct caller id for Channel: SIP/103-098500d8 is CallerID: 103
This is absolutely correct, it's just not a piece of information you
expected to be receiving at that point.
You probably also received a packet like that
My polycoms all have dtmfmode=rfc2833 and they work fine on both
asterisk's IVRs and external ones brought to me from the PSTN:
[120]
type=friend
context=internalaugmented
secret=a_secret
host=dynamic
*dtmfmode=rfc2833*
Moj
Jarga Jallow wrote:
Hi,
I am having trouble making a selection
Steve Edwards wrote:
Or, as a quick dirty...
DATE=$(date +%F-%H-%M-%S)
COUNT=$(sudo /usr/sbin/asterisk -r -x sip show channels | wc -l)
echo $DATE $COUNT /tmp/channel-counts
in a shell script executed every second in cron.
every *second* from cron? how the
Steve Edwards wrote:
in a shell script executed every second in cron.
every *second* from cron? how the heck would I you do that? sub-minute
accuracy from cron is something I don't know how to do.
Sheese -- that's what I get by trying to type without putting down the
crack
What does 'make menuselect' let you choose? Under #3, Channel Driveers,
does chan_alsa have XXX through it so you can't select it? does
chan_oss have XXX? This would indicate to you that the pieces of alsa or
oss asterisk would need are not installed properly.
Moj
Gilberto Nunes wrote:
Hi
Some phones have the auto-answer ability. So your phone could have two
extensions, one for normal use and one for auto-answer use. Redirect or
Originate, as you were, to the auto-answer extension on the phone. So
the phone would already put itself offhook, and asterisk would continue
and
It is when you type 'make install' that these directories get created.
'make linux26' IS obsolete as another poster mentioned.
broadband Voice wrote:
I successfully obtained the Asterisk code and extracted them into
/usr/src. When I make and install asterisk, zaptel, libpri etc. Are
they
Jerry Jones wrote:
Yours should work if you wait long enough for t to timeout.
I think your digit map needs a T on the end of it if you want to allow
timeouts for that match.
___
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I would GUESS that if this line is removed, asterisk is settling on slin
codec for the channel and does not try to negotiate anything better?
Hence it will work without it.
Mojo
Bhrugu Mehta wrote:
hi, all
I have test echo application for just fun.
I can'nt understand why this is used
Doug Lytle wrote:
Michael Munger wrote:
only connects me to a dial tone and says Enter More Digits.
It actually says this?
I would say then it's not the phone, but your phone system's
programming. The Polycoms don't verbally say anything, at least not the
ones I deal
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