ck to get around
> what seems like a clear limitation.
>
> I'll keep looking for a more elegant solution over the next couple of days,
> and give this a go if nothing "cleaner" turns up. Thanks for suggesting it!
>
>
> ----- Original Message
> From: &q
For my wife I recently set up a cron schedule that, every ten minutes,
greps the output of "show voicemail users" for a new message waiting.
Upon finding one, it dumps a call file into asterisk's outgoing
directory that rings the house phone and, when one is picked up, it
connects the user to
Just to be clear, I thought that dialtone provision didn't require the
power cable, just generating ring voltages? Can anyone say?
Moj
Anthony Messina wrote:
> On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote:
>
>> Hi:
>> I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on m
For me as well, if I append a #, asterisk tries to match it to an
extension and I get congestion indication when it fails to do that. This
would be nice though!
(TDM card too)
Moj
Giuffredi wrote:
>
> If I press # I get “incorrect number” as asterisk passes to the telco
> the numbers and the
Anthony, Robert here is mentioning a SIP phone, but I didn't see you
specify what kind of phone you have. Is this accurate, or is it a
Zaptel FXS port?
Moj
Robert Lister wrote:
> On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote:
>
>
>> well i'm looking for the feature that t
lving this issue.
>
> Best Regards,
> John
>
>
>> From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
>> Reply-To: Asterisk Users Mailing List - Non-Commercial
>> Discussion
>> To: Asterisk Users Mailing List - Non-Commercial
>&g
While I can't say this won't work the way you have it, I CAN say it's
not the way mine is set up and it's not a way I've SEEN it ever set up.
Could it just be complaining that you've got nothing on the right side
of the => for mailbox 200?
Or could it be complaining that you don't have anything
(With regard to your final question)
As far as I can tell, EAGI is AGI with the extra file descriptor devoted
to the linear pcm audio stream. As such, I would assume, but have never
tested, that in order to send DTMF OUT from your AGI app, you would need
to use the AGI functions, i.e. "EXEC SE
> CISCO_PROMPT then it will not work we CISCO_PROMPT is
> not UNIX variable)?
>
> Regards
> Bilal
> --- "Mojo with Horan & Company, LLC"
> <[EMAIL PROTECTED]> wrote:
>
>> You seemed to be unclear about unix variables:
>> >> The questi
From memory, your zaptel and zapata files look ok. signalling for an
FXO module would be FXS, and vice versa. As far as I can tell, you're
ok there.
Now, it's the FXO card that plugs into the phone line. The FXS card
gets a phone hooked up to it. Dialing the phone would be
Dial(Zap/1...
I'm not sure what features/variables you can use, or where to find
information about that, but what this basically means is you can change
your CLI prompt by this:
export ASTERISK_PROMPT="new prompt >"
then, what you access the CLI, instead of:
hostname*CLI>
you get
new prompt >
Moj
bilal gh
yeah, 'enough' adds back the gray area that the black-and-white 'atomic'
obscures... :P
Moj
Philipp Kempgen wrote:
> Mojo with Horan & Company, LLC wrote:
>
>> set your own mutex using astdb? It may just be atomic enough for you to
>> get by.
Is your dynamic page returning a newline after, like "SIP/12345-1\n"?
Moj
Mike wrote:
> Hi,
>
> Here is my first step (call it a proof of concept) in using the "hint"
> priority with dynamic values.
>
> Background - this works
> exten => 12345,hint,SIP/12345-1
>
> To make this a little dyn
I remember now, it totaled the *times* of cdr entries after grouping
them by account code.
Mojo
Mojo with Horan & Company, LLC wrote:
> Free and inexpensive aren't quite the same. I don't follow the -biz
> list because I don't want to hear plugs from "You at
Free and inexpensive aren't quite the same. I don't follow the -biz
list because I don't want to hear plugs from "You at Evariste" about
this stuff.
I sent a PHP snippet to the list maybe a year and a half ago, search
something like
"site:lists.digium.com Mojo csv"
or
"site:lists.digium.com
to clarify what I'm talking about:
I'm referring to the soundpoint ip admin guide for version 1.5 for
example. The wording is in section 4.6.1.15, or page 113. The
key *numbers* referred to, however, are found in section 3.1.7,
beginning on page 21.
Moj
Mojo with Horan &
I'm not sure of the correct wording in ipmid.cfg or sip.cfg, but I think
you'd be most successful using the block. A probably wrong
example might be:
for a soundpoint 50x and 60x.
or
for a soundpoint 30x
But it at least might get you pointed in the right direction. If Null
isn't what you w
set your own mutex using astdb? It may just be atomic enough for you to
get by.
Nicholas Blasgen wrote:
> I've got 4 SIP phone lines with a call-limit of 2 for each. I've
> written a handy macro to allow my users to dial a phone number and the
> macro will figure out the next available line t
But the OP stated this was SIP <-> SIP calls -- As Dan mentioned, check
environmental issues -- hard walls, poor handset quality, noisy desks,
volume levels too high?
Eric "ManxPower" Wieling wrote:
> by Cisco As for Echo Canceling, that is the job of the device that
> does VoIP/PSTN gatew
Can you verify the correctness of that URL? 404 from here.
Thanks!
Jared Smith wrote:
> On Tue, 2007-07-31 at 10:16 -0400, Matt wrote:
>> Does anyone know the algorithm that Asterisk uses to figure out when
>> you'll be speaking with an agent?I've heard it say such bizarre
>> things as '2 min
n
an easy fix ;) No offense intended.
Bruce McAlister wrote:
> Mojo with Horan & Company, LLC wrote:
>> Sorry that this is unrelated but, Bruce, do you double-click to send
>> your messages? Just curious.
>>
>
> Sorry that this is unrelated but, Mojo with Hora
Sorry that this is unrelated but, Bruce, do you double-click to send
your messages? Just curious.
Bruce McAlister wrote:
> Darryl Dunkin wrote:
>> Correct, if you have multiple licenses in there (say a single storage
>> location for a cluster of servers), it won't load.
>>
>> If you've tried oth
I would think if one didn't want caller ID they wouldn't pay the phone
company the extra couple bucks for it.. but just coincidence maybe :)
Paulo Garcia wrote:
> Hi everybody.
>
> I'm in a discussion and someone ask me in which situation we should use
> the zapata.conf usecallerid set to no.
But only when dialing bob or charlie. Only the second line, the
'include' line, is for call parking. The others are NOT for call
parking and are unrelated -- They are just for dialing charlie and bob
directly.
Jared Smith wrote:
> On Sat, 2007-07-14 at 13:20 -0700, bilal ghayyad wrote:
>> [i
After the Dial application completes, the variable ${DIALSTATUS} will
contain something like BUSY or NOANSWER or CONGESTION (CONGESTION here
means like no free phone lines or no route to destination for example)
Then, immediately after the Dial line is the Goto line
Goto(s-${DIALSTATUS},1)
th
try to transfer current call to parking spot, i.e. exten 700, then deal
with new incoming call, then go back to parking space to pick up old
caller when you're free. Just set the parking extension timeout to
something long so they don't fall out right away.
Moj
Joe acquisto wrote:
On 7/1
Is your incoming context using chanisavail, while your internal-dialing
context is not, and just sends the call, without checking?
Mojo
Michael Wareman wrote:
> Hi,
>
> I have (to me) an interesting problem.
>
> There are 3 physical extensions, 11, 12 and 13. All hang off Sipura
> adapters.
>
About two years ago, using chan_sccp driver, I was pretty amazed by the
workability. There were a few random reboots in the phone, and in my
understanding, this was the result of the channel driver because the
phone is more like a puppet than a real phone.
I would only expect it works better n
(Using things like fgets and strcmp and substr and sprintf and such.)
I've found that reading everything asterisk's got for me in one go and
then using explode function with '\r\n' delimiter seemed to work well
for me.
Anthony Francis wrote:
> Arun Kumar wrote:
>> Hi
>>
>> this is my code for *
No idea if this is where your problems are coming from, but change:
exten => s,n,Set(TIMEOUT(response=5))
to
exten => s,n,Set(TIMEOUT(response)=5)
(the parenthesis moved a bit)
Peder @ NetworkOblivion wrote:
> I am using the Find-me/Follow-me example below with screening:
>
> http://www.v
But those are not REGEX expressions, those are asterisk dialplan
pattern-matching expressions. great for the X in:
exten => _X.,1,blah
but not for use with REGEX() function.
I think it would be close to what Michael said, but like this:
1{0,1}[2-9][0-9]{2}[2-9][0-9]{6}
Michael, your expressio
I think he means "will he be able to tell what passwords were used by
looking at the cdr, for planning purposes, and who used the passwords?"
Bilal, using the CDR you could, of course, know who dialed international
numbers. The password, however, will not make it back there unless you
make sur
bilal ghayyad wrote:
> Also how can I add digits to the numbers like adding
> 00 in the beginning, so the dialed number becoming:
NUMBER=11336784888
exten => 1000,n,Set(NUMBER=00${NUMBER})
> 0011336784888 or adding digits (like 99) in the end of
> the dialed number, so it will become: 00113367848
you might use
sip show peer peername
to see what a peer will allow or
show channel channelname
(channel name as retrieved from show channels)
to determine what a current conversation is using
Moj
bilal ghayyad wrote:
> Hi List;
>
> Where I determine the codec to be used for the SIP
> Trunk
Ryan Goldberg wrote:
> Alternatively, the first line could be:
>
> exten => 101,1,Dial(SIP/${EXTEN}&Zap/4/12185551212,30,tpm)
>
> which would dial both the desk and the cell at the same time...
I've tried doing things like this. What got me was that SIP technology
allows for the phone not a
To determine versions:
for zaptel, use
ztcfg -v
For asterisk, in CLI:
core show version
Moj
bilal ghayyad wrote:
> By the way: How can I know the asterisk and zaptel
> version extactly that I compiled them? In other words,
> asterisk 1.4 and zaptel 1.4 ?
_
I was going to mention that too, even though I wasn't sure it was a
problem. I have spaces around my '=>' but nowhere else in the line.
Stephen Bosch wrote:
> Asif Raza wrote:
>> hi,
>> i am using Asterisk 1.4. and unable to get Voice Mail below is my config
>>
>> extensions.conf
>> exten => 50,
or copy /usr/src/asterisk-1.4.6/configs/http.conf.sample /etc/asterisk
and edit, because make samples I believe wipes out existing configs
Russell Bryant wrote:
> hugolivude wrote:
>> I just installed Asterisk 1.4.6. I didn't see http.conf in
>> /etc/asterisk. Is there a seperate install for AJ
Is it taking a while for _your_ messages to post to the list, or do you
mean messages from the mailing list software take days to get to you?
Moj
Lenz wrote:
> Hello list,
> I am getting the list with days of delay, take for example this message:
>
> Received: from unknown (HELO lists.digiu
Sounds like your filename.pl script should be in a cgi-bin directory
rather than in a document directory?
How exactly are you doing this from asterisk? Is this for a
microbrowser in a desk phone?
Moj
Robert A. Rawlinson wrote:
> I have Apache2 set up and running on a system I only use for te
Yes, you can only send calls to peers, not receive them, so no context=
needed.
Moj
bilal ghayyad wrote:
> Hi Noah;
>
> The reason that I am asking wether I need to determine
> the context is what I read in the documentation (about
> configuring outbound IAX connections), it did not
> mention th
I see three parts to this if I was doing it.
1) set up an extension that, when dialed, requests a huge pin number.
upon successfull pin number entry, it 'touch'es a file on the server to
update its modification time
[internal]
; could be extension to update heartbeat, asks for pin next
exten
From what you provided, I'm not sure that 'firewall is disabled' will
help you. Your firewall probably needs to be configured to forward some
ports to the asterisk box's internal IP address. I usually do the
following:
For SSH connections to the box to manage it:
forward a.b.d.c's external p
just an idea, but maybe qmail, samba, and bind have a smaller memory
footprint than an in-use asterisk? can you take the hardware offline
long enough for a memtest?
Moj
Luki wrote:
> It's no unusual seeing uptime for say
> qmail, samba or bind of 200+ days.
___
Cheers
>
> Alex
>
>
> On Tue, 2007-06-12 at 09:21 -0800, Mojo with Horan & Company, LLC wrote:
>> theoretically, with canreinvite=yes, it's phone <-> phone. with
>> canreinvite=no, it's phone <-> asterisk <-> phone. BUT there are a
I think what Jared recommended, looking at the sip messaging, will help
you here. He means to type "sip debug" in the asterisk CLI and look for
hints that SDP is being specified in the conversation. If it IS being
specified, then check into NAT/firewall issues, as he recommended also.
Mojo
s
Use the "dialplan show" CLI command ("show dialplan" in 1.2) to show
you exactly what asterisk has picked up, and scan it for aforementioned
leaks.
Rizwan Hisham wrote:
> Then i think u should use Atis's idea of using transfer_context
> variable...you should set it inside your dialplan and
Configure the channels with the proper disallow= and allow= lines, and
Asterisk should figure the rest out.
I could be making drastic assumptions about your situation, but it seems
like this:
-sip.conf
[userX]
...
context=internal
disallow=all
allow=gsm
allow=ulaw
...
[fax]
...
disallow=al
For real? I thought _ was to tell asterisk it was time for some pattern
matching:
; exact extension, exact cid
exten => 5000/19256002182,1,Answer
; any extension beginning with 5, from specific cid only
exten => _5./19256002182,1,Answer
; match exactly extension 5000, but anyone calling from
;
In his defense, when I first posted to the list, I wish I had found some
instructions somewhere that said "messages might take at least a half
hour to reach the list, so _don't_ double post!" Hopefully this upgrade
to the list software will reduce some of the strain on its server.
William Moor
ution isn't pretty, but it works. :)
Rob
Mojo with Horan & Company, LLC wrote:
I think you mean 60x not 50x. The polycom 501s don't have the
microbrowser.
Rob Schall wrote:
If they're polycom 501s or higher, you could have each phone use a
different homepage. Those pages cou
Awesome, thanks for this tip!
Moj
Dave Fullerton wrote:
Actually they do, but only if you're running SIP firmware 2.1 or higher.
Mojo with Horan & Company, LLC wrote:
I think you mean 60x not 50x. The polycom 501s don't have the
microbrowser.
Rob Schall wrote:
If they'
Actually, sorry to not research this first:
"14759: Added microbrowser support to the SoundPoint IP 501 platform"
from
http://www.voip-info.org/wiki/index.php?page=Polycom+Microbrowser
But I'm not sure which SIP firmware this is talking about being present in.
Mojo with Horan
I think you mean 60x not 50x. The polycom 501s don't have the microbrowser.
Rob Schall wrote:
If they're polycom 501s or higher, you could have each phone use a
different homepage. Those pages could be loaded dynamically (say in php)
and then you could just store the "message" to display in a d
Don't forget that you might be able to write that:
exten => s,3,...
exten => s/8585979857,4,Goto(15)
exten => s/8585970327,4,Goto(15)
exten => s,4,Goto(5)
exten => s,5,...
The closest-matching priority 4 will be chosen, even if it's simply
"well go on to priority 5 then"
Moj
C. Chad Wallace
theoretically, with canreinvite=yes, it's phone <-> phone. with
canreinvite=no, it's phone <-> asterisk <-> phone. BUT there are a few
reasons which canreinvite=yes will not be this way. If for example you
have a T or a t in the Dial string, asterisk will _remain_ in the media
path so it ca
I guess he might mean don't include the -g on the command line?
I'm wondering if asterisk is running in the background of the console
you're logged in at, so it's dumping messages to the console, AND you've
connected with -r?
Moj
Barton Fisher wrote:
Eric "ManxPower" Wieling wrote:
This i
I guess the problem with these mirrors is you can't see the mirror links
unless you can GET to voip-info.org so here's mine:
http://voip-info.sitkavoip.com/
And the mirrored list of mirrors, to choose one closer:
http://voip-info.sitkavoip.com/wiki/view/Voip-Info+Mirrors.html
Moj
Ed Nuñe
I don't think OP meant "the phone generates a flash even though the
phone wasn't picked up" I think they meant "the flash generated when the
phone's flash feature is activated is not picked up by asterisk"
am I right?
I believe the setting is in zapata.conf, in the section pertinent to the
FX
Have you tried something along the lines of:
System("swift blah blah blah -o blah.wav")
Playback("blah.wav")
It does have an inherent delay for the generation step but maybe swift
binary segfaults less? I've only used cepstral via swift binary, and it
has never segfaulted for me. My swift and
I must be not understanding your question very well, because it seems
like an easy answer :)
In the following Dial event, we have Source and Destination. Like Eric
said, Destination can contain multiple devices, so can't be trusted.
But Source should only contain one device Does it have
Oops, I meant to include in my prior note that you could of course
generate the wave file with flite or swift for text-to-speech
Matthew Pease wrote:
Hi all -
Not really sure where to post this question as I am just starting to
research this issue.
We want to allow users to dial into our did
I think you're on the right track. You need to decide where to store
the CID->data mappings (files on disk, astdb, mysql, generated
on-the-fly) and come up with what the wave files are (text to speech?
selected from pre-made recordings?) I would do the brunt of the work
with a script instead o
the 'make' command would typically recompile and re-link only the files
that have changed. Not sure how well this works with asterisk, but I
think that's the idea.
Mojo
Arpit Mehta wrote:
hi,
This might be the most obvious thing to you. I need to change some parts
of the source code of Ast
what version of asterisk has that application? Not seeing it in 1.2 --
but I seem to recall that maybe the "core blah blah blah..." command was
from 1.4?
Mojo
Shane Young wrote:
Quoting Curt Shaffer <[EMAIL PROTECTED]>:
I wanted to see if there was anything reasonable in price out there
ye
Although they're not free, cepstral voices are an option. They sound
really nice -- http://cepstral.com/ . They range between $7 and $30.
Moj
Nitesh Divecha wrote:
Thanks Shanon and everyones input...
Finally, got the application working as planned with PHPAGI...
Now the only draw back is
No, this is just reboot -- no factory reset.
Rob Townley wrote:
On 5/30/07, *Mojo with Horan & Company, LLC* <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> wrote:
An answer to your original question: if you can get someone _to_ the
phones, on the polycom 30x'
Maybe you've got asterisk out of the media path, and the DTMF is heading
straight for the remote phone? Keep asterisk in there, possibly with
canreinvite=no or either with a t or T option in the dial string. There
may be other ways to accomplish this too.
Moj
Carlos Chavez wrote:
I
An answer to your original question: if you can get someone _to_ the
phones, on the polycom 30x's you would hold the VolDn, VolUp, DND, and
Hold buttons for a while to reboot.
For anyone with the 50x or 60x, you would hold the VolDn, VolUp,
Messages, and Hold buttons.
Moj
Forum wrote:
I hav
SIP//peer/-/id/ is so confusion until I realized it was SIP/peer-id with
peer and id made italics...
Mojo
Marc Hurstel wrote:
Hello,
In Asterisk 1.4.4, the SIP channel names (SIP//peer/-/id/) do not seem
to be unique. That means that the "id" associated to a "peer" is not
random.
Is that
Does the non-Asterisk server _answer_ the line? :)
Gavin Henry wrote:
Dear All,
I have a tiny dial plan like:
[testing]
exten => 454,s,Ringing()
exten => 454,n,Wait(4)
exten => 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10)
exten => 454,n,Hangup
This connects fine when I dial 454 from any extensi
The latest and greatest Astsee is now available at
http://www.astsee.com/ I'm up to v0.5 today -- the "light at the end of
the tunnel" edition.
In progress is a way to audit this sort of traffic _without_ manager
credentials ;) Just by sniffing it off the wire or out of the air...
You can te
r
surpass my expectations I've learned by working with the Windows version
:) No offense meant to the maintainers of the gastman project!
Moj
Mojo with Horan & Company, LLC wrote:
Hiya everyone. I have been working on a fun little app to watch what's
going on in your aste
Hiya everyone. I have been working on a fun little app to watch what's
going on in your asterisk box via its manager interface. There's a
screenshot up and some info at http://sitkavoip.com/astsee/ -- Sorry it
requires allegro, but I was more keen about getting the ideas down than
worrying ab
Have you tried:?
modprobe ztdummy
Mojo
Manolet Gmail wrote:
2007/4/18, Ronaldo <[EMAIL PROTECTED]>:
Hi Manolet,
You have to install zaptel in order to make MeetMe application to work.
MeetMe needs a kind of timer device that is provided by zaptel package.
Eventhough you don't have a zaptel
To follow up on the don't need to dial 9 to get out topic, in some
places, there are so few phone prefixes, you can simply match them
exactly. Here's for where I live:
exten => _747,1,Dial
exten => _966,1,Dial
exten => _738,1,Dial
exten => _752,1,Dial
exten => _1NXXNXX,1,Di
When you are dealing with the PSTN network, you generally DON'T need an
'r' in your Dial command options. This will force ring indication for
situations where the PSTN carrier does NOT provide them already. I'd
bet yours *does* provide ring... Hook up a regular phone to the wall
jack (skippi
IIRC, you need an extension named 'callpark' in your extensions.conf
that calls the ParkAndAnnounce application.
This should get you started:
exten =>
callpark,1,ParkAndAnnounce(PARKED|600|Local/4${BRIDGEPEER:5:[EMAIL PROTECTED]|incoming,s,1)
in the CLI:
Show Application ParkAndAnnounce
for
You don't need the power cable. It is only there to provide the
necessary ring voltage to anything you may have plugged into installed
_FXS_ modules.
Henry Cobb wrote:
I've just moved into 3.3v PCI servers and found that my clone X100P
cards were lying about the 3.3v supported notch.
Can I u
I've used the Dock 'n Talk, and I can say that it worked as well for us
as it claimed to be able to. Only need an analog Zaptel card of some
sort. I know there are a few other brands available, as well as some
"GSM Bridges" available that you insert the SIM card directly into,
bypassing the c
In 1.2, try adding "noload => res_musiconhold.so" to your modules.conf.
In 1.4 though, it would be worth a try, but I don't know for sure if
that's how it's done.
Moj
David Thomas wrote:
On 3/5/07, C F <[EMAIL PROTECTED]> wrote:
Could be its trying but does it actualy play the music?
It's
Possibly the called party is not sending their DTMF properly? maybe
experiment with inband/rfc2833/etc in the CALLED party's peer definition
Denis V. Gudtsov wrote:
Hello!
I'm using asterisk 1.2.13, in extension.ael is set Dial(SIP/${EXTEN},12,tT)
but only calling party can do forward. How t
Another option is to have the user hit the forward button on their phone
and manually type in their cellphone number when they're going to be out
of the office.
Jason Walker wrote:
exten => 111,1,Wait(1)
exten => 111,2,Playback(Randy)
exten => 111,3,Dial(Sip/Randy,20)
exten => 111,4,Goto(111-$
I DID receive it. Please don't re-send it.
C F wrote:
Hi, I'm the admin on the list your test didin't work you should resend it.
Well I am not the admin, just wanted you to realized
All the calls you listed in your example were simultaneous calls *from
the same user*. Is this what you were intending?
I'm assuming this 'cause the Peer column contains the same IP address
each time, and the User column contains the same User number. Only the
Call ID changes.
Moreover, in
But just to handle 10 simultaneous calls, you probably don't even need 1
GHz!
Matt Richards wrote:
I don't see any reason why a single server wont handle 700 phones as
long as its powerful enough.
I would think that anything over 1GHz should be fine maybe less :)
Matty.
Jerry Geis wrote:
I h
As far as I can tell, the only way to do this using Polycom soundpoint
phones and NOT asterisk's built-in blindxfer function, is to hit their
Transfer button first, and then the Blind softkey that appears on the
screen. Then continue as normal; dial the number and hit Send I
believe. If you c
Yuan LIU wrote:
Doesn't seem to happen in TDM400P and X100P cards, though. Could it be
some feature configured in your particular card?
Notice just after my name:
P.S. Polycom Soundpoint 501, *TDM* w/ 4xFXO, Asterisk 1.2.13
As El ManxPower mentioned, have you tried using ZapBarge to detect t
When we do SIP <-> SIP with asterisk 1.2, we do NOT experience this.
Polycom 501s, the instant you hit Send on the phone or the digit map
times out, the target phone rings AND you hear ringback. it's instant,
so I would guess this would be configuration on your end. back to the
digit map timeo
One thing I've noticed with SIP -> ZAP calls for quite some time is that
when asterisk is dialing n digits out the zap line, it dials n-1
digits, pauses for *TWO* seconds, and then sends the nth digit. Doesn't
matter how many numbers I want to send out the ZAP channel, this always
seems to ha
The DISA application ("show application disa" at the CLI) will let you do
this.
Mojo
On Saturday, January 06, 2007 10:19 pm, Yuan LIU wrote:
> After the user navigated some voice menus, how do I give him another (fake)
> dial tone?
>
> Yuan Liu
>
>
> __
So if your ATA or phone has to connect to your computer, your computer
then has to bridge its two ethernet connections together to give access
to to the 'external' network. Keep that in mind, it's usually not too
difficult.
Those phones you provided info on are SIP phones, not IAX phones.
I
Sorry, s/b "Zap/Channel/Phone_number" like "Zap/g1/18005551212" -- I
forgot the zap channel part :)
Moj
Mojo with Horan & Company, LLC wrote:
Well, as a hopefully helpful pointer:
Shawn Kelley wrote:
all I see there is creating calls and sending them to system ph
Well, as a hopefully helpful pointer:
Shawn Kelley wrote:
all I see there is creating calls and sending them to system phones, etc. I
I assume by system phone you must mean an internal SIP phone for
example, like "SIP/110" or something. Couldn't you replace that channel
name with "Zap/phonenu
working close to the
edge. I, too, have experienced similar problems when power was limited
and have had to, temporarily, resort to a "bigger" power supply to get a
system installed. Then fell back to a "smaller" one in operation.
Good luck.
Bob...
On Wed, 2006-10-18 at 08:
Tzafrir Cohen wrote:
Is there a simple and safe way to query the astdb database outside of
Asterisk?
after writing to it with:
asterisk -rx 'database put phones locked 1'
something like
asterisk -rx 'database get phones locked'
returns 1...
Is this what you mean by outside of asterisk? Sorry
I set up a similar system on an VIA Epia 5000, and I had issues when I
included the CDROM in the mix. I had to use another ATX power supply to
complete the install, but then once I removed the CDROM drive I had no
power issues.
I presume you could install the OS with the CDROM drive installed
You don't run a function in the GotoIfTime application, you point to
another context/extension/priority to jump to that DOES have the
applications you need, as Conrad exampled.
Moj
[EMAIL PROTECTED] wrote:
Is there a way to run a macro in a GotoIfTime statement ??
from the wiki documentation
As mentioned recently on the list in other posts, don't forget to allow
emergency calls through no matter what unless you have tremendous
lawyers
Conrad Wood wrote:
On Tue, 2006-10-17 at 10:25 -0500, Carlos Chavez wrote:
I have a customer that wants to lock his phone when he goes
You're a little backwards. When you connect to a remote server via HTTP
protocol, for example, you ARE connected to their remote port 80. They
do not send data to YOUR port 80 though.
Moj
Time Bandit wrote:
On 10/16/06, Melcon Moraes <[EMAIL PROTECTED]> wrote:
OMG, please read more about n
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