If the users have a bluetooth device like a cellphone-with-bluetooth or
their laptop, this might work: http://mundy.org/blog/index.php?p=78 -
you'll have to modify the script in the tutorial a bit.
essentially - you have a presence server at the two offices - when they
enter the building, the
On 2/6/06, Conrad Wood <[EMAIL PROTECTED]> wrote:
> Please note the spelling of uniqueid. I find the spelling in
> res_features.c - but only once I patched it with bristuff patches.
> Does anyone know whether that is a known problem with bristuff? If so is
> it fixed in a later version?
What versi
Hi All, Please help me solving this problem. Thanks Somesh S. ShanbhagSomesh S Shanbhag <[EMAIL PROTECTED]> wrote: Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and
Hi the list,
I can confirm you that I have not noticed any echo issue in this
configuration (analog phones on quadFXS modules AND analog lines on quadFXO
modules) at the same place and Asterisk when some echo issues occured with
IP-Phones.
TDM2400E is an excellent choice :-)
Best Regards,
Franco
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Mark Phillips
> Sent: 07 February 2006 19:23
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk native sounds now available!
>
> Kirs
Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Appl
Hi All
Has anyone come across a handset that can somehow replace FOP? Some
users don't like FOP unless it is on a dedicated PC.
Thanks
Garth
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On 2/7/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
> Tower Server with Digium TDM04B (4FXO Card) - Roughly $1000
>
> 8 Port FXS gateway - $600-$1000
(snip)
For an application like this, what would be the advantage of spending
$600-$1000 on an 8 port FXS gateway rather than spending $280 on four
2-
The 5220's I have are the Dual Mode versions, so I do have them working good
with Asterisk in SIP mode. I'm just wondering if anyone had any luck with more
advanced features.
good luck and if you find out let the list know ps I have 30 5220's for sale
with MiNet. Same as you - never got them w
Here are a few possibilities:
Tower server with Intel P4 proc, 1GB RAM, ATA or SATA Hard Drive, NIC Card,
CD-Rom, etc. with available PCI slot - $500 - $700
Digium TDM2421B (4FXO/8FXS) - Roughly $1K
~ OR ~
Sangoma Remora A20204 Analog PCI Card Assembly (4FXO/8
Word. I'm doing a dupe of my production server this week as a CYA. Guess
what: FC2. Once I yum update to the current kernel, no more yum. There's no
reason to. You may have your own reasons (publicly avaliable server, for
example) but why add uncertainty to an, at best, quite uncertain process
(tha
Hi Guys,
I want to setup an asterisk PBX for a small office. I'm looking at
connecting 2-4 PSTN lines and having about 4-8 analog phones for
extension. I'm looking for some hardware suggestion from you folks so
that I can do this pretty economically. Or if there is a guide for cheap
SOHO setups Id
good luck and if you find out let the list know ps I have 30 5220's for sale
with MiNet. Same as you - never got them working good with *.
Mitel won't let you know either unless you are one of their anointed VAR's
-Original Message-
From: Bromont [mailto:[EMAIL PROTECTED]
Sent: Tuesday, F
Zach A wrote:
What is recommended for a production quality system, FC3 or FC4. Once
installed, is it necessary to run yum update, does that make things any
better or just take up more memory?
I wouldn't recommend Fedora Core for a production system - at least not
a server. For one thing, FC3
try sangoma carrier grade 104d hardware EC card. we're using it ourself.
Best Regards
Matt
- Original Message -
From: "Anthony Rodgers" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, February 07, 2006 12:57 PM
Subject: Re: [Asterisk-User
Will not work, and also not all 537ep's work either, this is from my own
personal tests
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of stevanus
Sent: Monday, February 06, 2006 3:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [A
Hi all,
I've got a problem with my FXO cards. I've configured
them to give a service to people on PSTN network, to
call the lines connected to my Asterisk by a digium
fxo card, and dial my VoIP network numbers.
PSTN -> Asterisk -> SIP Client
The problem is when a call is made by a user from PSTN
Has anyone here had any experience with Mitel 5220 IP phones with Asterisk?
Basic features are working good, but I'm looking for more advanced
features like
sending text to the display or having the lights on when an extension is
in use via the
hint subscription. Thanks.
__
This is sound advice worth taking. If you get a system stable in
production, LEAVE IT ALONE!!
I say this to spare you lost nights and weekends wondering how things
could have gone s wrong...
Test and tweak on a duplicate system if it needs to be done.
Technical Support wrote:
We run
That was exactly it! Thanks you VERY much!
Mike
For the sip setting in sip.conf that setsup your voip provider add:
canreinvite=no
On 2/6/06, Michakl Gaudette <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I've had a bit of a problem with one way audio, and it happens exactly
when
> I believe it
You cant go by pings. ICMP traffic is given lowest priority on internet
routers, where voip rtp or iax might be given much higher priority. Plus I
have 2 providers, the provider with the 90ms ICMP ping time is way better
than the provider with the 15ms ping time. It depends on so many factors,
We run FC4 on our production installs. It runs great. I should caution you
that just because an update is available, it doesn't mean you SHOULD update.
Treat your FC4 install as frozen - if it works don't update it!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi everyone,
What is recommended for a production quality system, FC3 or FC4. Once
installed, is it necessary to run yum update, does that make things any
better or just take up more memory?
Zach A.
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My sip show channels shows some channels active that I can not make
sense out of, and they have been that way for days, so I am pretty sure
they are orphans.
Is there a way to show active CALLS (instead of channels) to try and
determine the source?
Does the output below provide any clues as to wh
I have a 2811 working as a SIP gateway. My IOS version is 12.3(11)T5. Looking through my config I notice:sip-ua sip-server ipv4:Everything else in the config file is for our h323 call manager gear. I can't remember if I needed to add the above line to make a sip server run on the router. In order
On Wednesday 08 February 2006 14:46, Chris Bagnall wrote:
> This is incorrect. Whilst the SPA3000 *can* work this way if you wish, it
> doesn't have to.
Apologies, you are correct, there is more than one mode of operation.
hads
--
timesharing, n:
An access method whereby one computer ab
Been covered Ad Nauseum on the list.
Asterisk does NOT detect dialtone
w, or a series of w's befor dialing begins will help, EXCEPT when doing
pulse dialing.
w does NOT work with pulse dialing
No one seems to think this is a problem, so it doesn't get addressed.
John Novack
Joseph Tanner wr
> The ATA will answer the POTS line, therefore the caller will
> be charged as soon as the ATA has tried to grab caller id and
> picked up the line (usually around two rings).
This is incorrect. Whilst the SPA3000 *can* work this way if you wish, it
doesn't have to.
Regards,
Chris
--
C.M. Bag
hi
I recently spoke to mr McNamara on IRC, and he mentioned there was a "far better way to do realtime-stuff than the usual realtime in asterisk, and that this was GPL". He failed, however, to ever mention how this could be done, so I just wonder if someone else might know... ?
At no point
> Would a call coming in on the pstn line be answered by the
> ATA or just get passed through to the * server (depending on
> dialplan) to handle?
Either. It's your choice. I have an SPA3000 here at home working in the way
you describe. When a call comes in on the SPA3000 it's forwarded (without
On Wednesday 08 June 2005 12:25, Richard Smith wrote:
> Would a call coming in on the pstn line be answered by the ATA or just get
> passed through to the * server (depending on dialplan) to handle?
>
> So basically, the caller does not get charged until the appropriate
> extension hanging of the *
Hi
all,
I was
wondering whether anybody here would help me clarify this minor issue please.
If I have
the following setup;
Asterisk -- Sipura SPA 3000 (fxo) - Pstn
Line
Would a call coming in on the pstn line be answered by the ATA or just
Log live the Python crew!!
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Colin Anderson wrote:
unfortunately the federal government in Canada mandates this and in Quebec
if you don't do it, you can be charged with a criminal offense.
French Canada farts in your general direction.
-
On 2/7/06, Imran Ahmed <[EMAIL PROTECTED]> wrote:
> > here is a little explanation:
> >
> > End user (You) -> Your Telco --> Carrier 1 --->
> > Carrier 2 Carrier 3 ---> Carrier 4(PTT)
> > --- > Far End User
> >
> > So basically, the Echo canc
> here is a little explanation:
>
> End user (You) -> Your Telco --> Carrier 1 --->
> Carrier 2 Carrier 3 ---> Carrier 4(PTT)
> --- > Far End User
>
> So basically, the Echo cancelling work backwards usually cancellation
> for you would be do
For the sip setting in sip.conf that setsup your voip provider add:
canreinvite=no
On 2/6/06, Michaël Gaudette <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I've had a bit of a problem with one way audio, and it happens exactly when
> I believe it shouldn't (and works perfectly when I would guess I could
I have tried it, and as far as I can tell, they both work, I did not
however test them both live with a T1 connected to both, just 1 T1
with both cards in, the lights settle on the other one without the T1,
while the first one with the T1 works, therefore I'm assuming it
works.
Are you sure that it
I had bad echo as well using the Te406 card. Swapped the card, swapped
the box, nothing helped, until I got a Tellabs 2572 echo canceler, and
echo is now gone.
On 2/6/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
> Doug Lytle wrote:
> > [EMAIL PROTECTED] wrote:
> >
> > I put a Tellabs 64ms echo cance
hi
I recently spoke to mr McNamara on IRC, and he mentioned there was a
"far better way to do realtime-stuff than the usual realtime in
asterisk, and that this was GPL". He failed, however, to ever mention
how this could be done, so I just wonder if someone else might know... ?
roy
--
Ro
IIRC with the 253c it can only be changed using the dip swithces on the shelf.
On 2/7/06, Dan Elder <[EMAIL PROTECTED]> wrote:
> 30 says it's view only in the docs & I can't seem to change it, any other
> options?
>
> > Option 30 allows to set Module Shelf Address/ID.
>
>
I "think" you can add "w" (without the quotes) to your dialplan to
wait. Perhaps putting a few in front of the number, or even one in
between each number? Not sure, haven't had to use this feature,
sorry.
Perhaps your provider doesn't like the duration of the dtmf tones
themselves. For that I t
Anyone have any SIP experience with the Coppercom softswitch?
Will asterisk interface reasonably well?
Does the Coppercom switch interface well with OTC sip phones (eg, Cisco,
Polycom, etc)?
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Config:AAH 2.2Digium TDM card connecting to 3 x Telus POTS linesPolycom 501 phonespretty basic setup, working mostly just fine...When I dial a number such as:96045551212Telus automation will sometimes come online and tell me that the number I have dialled cannot be completed as dialled.
If I hang u
Does anyone know of a good solution for secure (read: not plaintext)
passwords for voicemail? We'd rather not have to move configuration
in to a database just to be able to encrypt the passwords. We're
running the latest stable release (1.2.3).
Any hints are greatly appreciated!
Scott
Campon, mini-queues, see asterisk tips and tricks on voipinfo...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zach A
Sent: Monday, February 06, 2006 1:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Help
Hey guys,I'm trying to get music on hold working. I have a wav file. It plays fine on my windows laptop in all sorts of audio applications. If I put it on our asterisk 1.2.4 box and do something like:sox -V nov_2005.wav /var/lib/asterisk/mohmp3/nov_2005.raw
sox: Detected file format type: wavsox: C
You cant go by pings. ICMP traffic is given lowest priority on internet
routers, where voip rtp or iax might be given much higher priority. Plus I
have 2 providers, the provider with the 90ms ICMP ping time is way better than
the provider with the 15ms ping time. It depends on so many factors
Hello
How to use letters with xlite?
Thank you very much
--
Bayrouni
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unfortunately the federal government in Canada mandates this and in Quebec
if you don't do it, you can be charged with a criminal offense.
French Canada farts in your general direction.
-Original Message-
From: Mark Phillips [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 07, 2006 1:1
Hi,
I am checking out the quality at a few vendors, and althought I know it
doesn`t totally reflect call quality I am using ping as a cheap subsitute to
having a real VoIP testing system
The question I have is this one: given that one service gives me a 80ms ping
(pretty consistantly) and another
For what it's worth, we have been going through very similar issues
with a TE411P - with Digium support, we have basically gone as far as
we can with the HW EC, and are now using MG2 with much better results.
We have a Ditech EC box on order.
Regards,
--
Anthony Rodgers
Business Systems An
Far as I know, you cannot use a usb cable to connect a cellphone
directly to asterisk. You need something called a cellsocket or a
dock-n-talk. You use these to connect directly to a regular
telephone, so to connect to asterisk you'll need an FXO port.
I'd love to find something that would direc
Mark,
It definitely sounds like the carrier is
looking for something more than just ‘911’ on the D channel. Please
let us know what the carrier says about 911 dialing so that we can make sure
our *’s are all setup properly.
Thanks,
MC
From: [EMAIL PROTECTED]
[mailto:[E
Interesting - ours don't do that. Here's what we have in our
.cfg:
msg.mwi.1.callBack="*98"/>
Do you have that?
Anthony
On Feb 3, 2006, at 4:36 PM, Ken D'Ambrosio wrote:
Anthony Rodgers wrote:
> Hi Ken,
>
> When you say -any-, what do you mean? Messages in the Old folder, or
> wha
Dan Elder wrote:
30 says it's view only in the docs & I can't seem to change it, any other
options?
Not really, I just remember seeing the option when I was configuring
mine. Maybe do it without the shelf?
Doug
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Dov Bigio wrote:
Hi,
I made a simple menu using the
Background application and some wav files. I converted the wav files
using
for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm"; done
(from http://www.voip-info.org/wiki/index.php?page=Convert%20WAV%2
office wrote:
Hi,
i use in my extensions.conf a testline for an internal test :
exten => 10,1,MP3Player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3)
When i call 10, Asterisk answer and i see in the CLI, that MP3player
works without problems - but i can't hear the sound at the phone ?
Mark Phillips wrote:
One problem I can see is that you're not using the keys that come with
asterisk.
Mine (which works!) looks like this
iax.conf
register => user:[EMAIL PROTECTED]
[iaxfwd]
type=peer
context=from-fwd
permit=65.39.205.0/24
auth=rsa
host=iax2.fwdnet.net
inkeys=freeworlddialup
I forgot to add that you must have an IAX acount with FWD. A regular SIP
account won't let you then use IAX. You have to register for it.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Mark Phillips wrote:
One problem I can see is that you're not using the keys that come with
asterisk.
I have a call in with the carrier, below is the PRI debug, looks like it is getting hungup because of "Invalid Number format", I did try to use Setcallerid to change the callerID to a DID number in a previous attempt, but it still didn't go through. Not sure if that "invalid number format" is the
You can use the three-way calling feature on the cellphone, so one
user could talk to two different people at once. If you have more
than one cellphone, this might be tricky (you want only one actual
call going out per cellphone, but go ahead and let a second call be
placed through one sometimes f
As evident in the SuperDial script and others based
upon groups, you can place a call into a group, which can have a limit on the
number of concurrent calls. Can a call belong to multiple groups?
IE: I have only a limited number of channels to upstream X.
Downstream Y is only paying me f
30 says it's view only in the docs & I can't seem to change it, any other
options?
> Option 30 allows to set Module Shelf Address/ID.
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Try adding "insecure=very" to the guest user account in iax.conf. This
should not do a user/pass challenge on the incoming call.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
kevin ling wrote:
Not sure answer your question? Try to write some html code and let user
register the username
Aha!! why didn't I think of that.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Gonzalo Servat wrote:
On 2/6/06, Mark Phillips <[EMAIL PROTECTED]> wrote:
A customer of mine wants an IVR where the first 3 choices are
1 English
2 Spanish
3 French
I can build the IVR but how do I get
I've come across this in my dealings with my customers in Toronto. As an
Englishman I find it most infuriating. French is after all, the most
hated language in the world from an Englishmans perspective ;-}
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Derek Whitten wrote:
Colin Ander
Hi,
I made a simple menu using the Background
application and some wav files. I converted the wav files using
for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm"; done
(from http://www.voip-info.org/wiki/index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Aster
The same "7" sound file is used to indicate both time and quantity. The
sound file could be easily recorded to say "sept heure" but then every
time the VM system tells a user that they have 7 messages they'll hear
something like "vous avez sept heure notification" (excuse my schoolboy
French).
Signate runs asterisk on a SGI box.
Nothing special, do yourself a favor and just buy the SGI box yourself. In
fact I have 3 SGI boxes for sale. I’ll rip off the Signate labels and
sell them to you.
I worked out an asterisk load
balance solution, so I don’t need one all powerful P
Erm ... sorry. That should read "Kris et al"
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Mark Phillips wrote:
Kirs et al,
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One problem I can see is that you're not using the keys that come with
asterisk.
Mine (which works!) looks like this
iax.conf
register => user:[EMAIL PROTECTED]
[iaxfwd]
type=peer
context=from-fwd
permit=65.39.205.0/24
auth=rsa
host=iax2.fwdnet.net
inkeys=freeworlddialup
disallow=all
allow=ul
I dunno about your provider but I know that 2 of my 3 MCI PRI circuits
have no 911 abilities. MCI tells me this is becasue I have no local
dialing plan on them.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Michael Collins wrote:
911 **should** work on a PRI. If you are getting a hang
Hi,I am forcing caller ID to be sent to our VoIP provider using the SetCallerID app:exten => _91.,1,SetCallerPres(allowed)exten => _91.,2,SetCallerID("Company Name" <5>)
exten => _91.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])Ever since I started doing this however, the CDR gets overwritten with
> And (as GSM Restriction) one can do only one call per phone (conferences
> and "onHold" are managed by the GSM-"AP").
This was what i was actualy interested in. My idea was, when conferecnces
work, it should be possible to make 2 calls over 1 GSM phone at a time. But
apparently this wont work.
Kirs et al,
I did this already. It's on my website. Your most welcome to use them
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Kristian Kielhofner wrote:
Alex Barnes wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kris
I have a
fresh install of AMP. In the AMPortal, Setup, Devices or Users, I get:
Cannot connect to Asterisk Manager with user/password (set respectively)
This module requires access to the Asterisk Manager. Please ensure Asterisk is
running and access to the manager is available.
I chec
Benoît Mérouze wrote:
Kristian Kielhofner wrote:
Hello everyone,
As I promised at eTel last week, I have finished up work on my
"Asterisk Native Sounds" project. Here's a little diddy from
astlinux.org:
---
Asterisk Native Sounds are a collection of a
I have used 911 with PRI with nothing else
configured. Telco had to make changes to their router for DID numbers to call
through.
Adam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail
Sent: Tuesday, February 07, 2006
12:10 PM
To: Asterisk Users Ma
Alex Barnes wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner
Sent: 06 February 2006 17:48
To: Discussion of AstLinux - Asterisk on Compact Flash; Asterisk-
[EMAIL PROTECTED]; [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED
Brian J. Murrell wrote:
On Mon, 2006-02-06 at 11:48 -0600, Kristian Kielhofner wrote:
Hello everyone,
As I promised at eTel last week, I have finished up work on my
"Asterisk Native Sounds" project. Here's a little diddy from astlinux.org:
Which format would be best/cpu-easiest on an ana
Colin Anderson wrote:
Also if you have a nice linux script to take out some of the effort that
would be fantastic but if not I am sure the sox man page will help me
out.
Prep your WAV's as 8Khz mono. In a pinch, Windows sound recorder will do.
Then:
GSM:
#/bin/sh
for I in *.wav
do sox $I `
Douglas Garstang wrote:
You know, I'm still a little confused. Kristian, the original poster, said...
"I had Allison Smith (the voice of Asterisk) re-record all of the sound prompts
present in Asterisk 1.2. "
Was there really an extra 1400 sound files added from Asterisk 1.2 to Asterisk
1.2.4
I upgraded to 1.2.4 today and am having issues and can't figure this
out. Here's the bottom part of a "gdb" and a backtrace. Any
thoughts? May roll back to 1.2.3?
Mark
Reading symbols from /usr/lib/asterisk/modules/app_saycountpl.so...done.
Loaded symbols for /usr/lib/asterisk/modules/app
911 *should*
work on a PRI. If you are getting a hangup and you don’t see a valid
hangupcause, it might be best to get your carrier on the line and have them
monitor the circuit while you dial 911. They might be able to tell you what
the problem is.
-MC
From: [EMAIL PROTEC
Kevin,
Sorry for the interruption but I was replying here because the message
thread was on this list. Thanks for being gentle ;-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, February 07, 2006 9:24 AM
To: Asterisk Use
Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk?
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What do you do with the other 15 channels?
your zapata.conf says:
channel => 1-15 ;,17-31 => only 15 first channels on PRI
but your zaptel.conf says:
span=1,1,0,ccs,hdb3
bchan = 1-15, 17-31
You use all 30 channels in Zaptel.conf but only 15 in zapta.conf
I never configured Zap on asterisk and f
I'm not receving anything from the list, is this a Gmail problem? or
just my account?
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This can easily be accomplished with AMP using the Users and Devices mode.
http://voipspeak.net/index.php?/content/view/49/28/
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Alex Ongena
> Sent: Tuesday, February 07, 2006 8:55 AM
> To: asteris
AnyOne? any help?
As I'm looking at your zapata.conf I recall a problem in receiving dial-outs from a non-asterisk IVR to an * server1 and server1 routs the call to server2 with IAX2 in order to make a final dial command to a ZAP channel, but in server2 cli console I get the error (UNABLE TO CREA
No, what was rerecorded was the sounds that come with the asterisk
package. Digium has another package called asterisk-sounds that has
many additional sounds - that package was not rerecorded.
Douglas Garstang wrote:
You know, I'm still a little confused. Kristian, the original poster, sa
Dan Elder wrote:
I have the cards set to auto address assignment, but changed it to shelf255d
setting (option 31) & still get the same behaviour... is there someplace else
that this can be set?
Option 30 allows to set Module Shelf Address/ID.
Doug
___
On 2/6/06, Chuck Bunn <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I have tried both the stable version ARI-00.04.006 and the development
> version ARI-00.05.018 with the same results. I can see call detail
> records just fine but I cannot see any voicemail. I am using the
> voicemail extension and passwor
certainly on his first call, but it should be possible for him to explicitly
'register' and 'unregister'
On Tuesday 07 February 2006 17:06, Joe Tahan wrote:
> when exactly would you like to stream this "register me" thingy? whenever
> an employee picks up the phone to dial? or when? Please specify
You know, I'm still a little confused. Kristian, the original poster, said...
"I had Allison Smith (the voice of Asterisk) re-record all of the sound prompts
present in Asterisk 1.2. "
Was there really an extra 1400 sound files added from Asterisk 1.2 to Asterisk
1.2.4? Sorry, but I'm just not
I have the cards set to auto address assignment, but changed it to shelf255d
setting (option 31) & still get the same behaviour... is there someplace else
that this can be set?
Thx!
Dan Elder wrote:
> Anyone gotten two of the 2572 echo canceller cards to work in a 253c mounting
> assembly? I c
Not really sure, but once I had a problem when I changed the txgain and rxgain, so set them again to 0.0 and see how it will work.
Truely/
Ammar
From: "Jerome SOUCANY" <[EMAIL PROTECTED]>Reply-To: Asterisk Users Mailing List - Non-Commercial DiscussionTo: Subject: [Asterisk-Users] No sound on 10%
I badly need to get the callerid of the person who hanged up along with the extension dialed, and I need to do it with DeadAGI where channel variables are destroyed,
any ideas?
Or at least someone tells me why my * does not take PGSQL or MYSQL when I try to insert or retrieve data from a DB, as it
It's more helpful to learn more about pre-defined variables in asterisk, then you'll be able to develope more complicated agi scrips or dialplan checks, follow the below link:
http://www.voip-info.org/wiki-Asterisk+variables
Truely/
Joe
From: Joseph Tanner <[EMAIL PROTECTED]>Reply-To: Asterisk Us
I've used Voicetronix FXO/FXS ports and noted pretty heavy echo on both
short and long runs to other switches. We went through some steps to try to
tune the echo out using some settings on the card, and it helped with some
of the higher frequencies, but the problem still remains for many users. W
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