This can happen if the (mac-addr).cfg file is bad. The first time you
load the phone it will corrupt the flash and you will have to send the
phone in for repair. Make sure when you edit the file you don't add any
carriage returns...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EM
Hello,
I cannot make any call with my new quadBRI card
from Junghanns.net in my asterisk box.
My asterisk box is built on Fedora Core 3 linux
system. After compiling drivers from junghanns.net the card driver was loaded
correctly with
- modprobe zaptel
- insmod qozap.ko ( as FC3 is running
Hi
In voicemail.conf I have
attach=yes (tried with =1 and = thrue)
but I cant get asterisk to attach the voicemail.
Any clue ??? (using ast_data)
/HHA
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Joseph wrote:
Joseph,
1 - 0.9 still uses IAX2 (I think - pretty sure).
2 - Why are you using 0.9?
Maybe they should be the other way around...
I'm just using default installation whatever Gentoo is providing; this
is their stable version.
Joseph,
While I also use Gentoo(as do many oth
On Sun, 16 Jan 2005, Mike wrote:
We would like to know if there is a way to broadcast (in realtime) a
conferance.
http://www.voip-info.org/wiki-Asterisk+cmd+Ices
I haven't tried it though.
--
Regards,
Tobias Jönsson, Lund SE___
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On Sun, 16 Jan 2005, Dorn Hetzel wrote:
> On Sun, Jan 16, 2005 at 01:33:54PM +0100, Roy Sigurd Karlsbakk wrote:
> >
> > You can modify and/or link to GPLed code with commercial code and get
> > away with it as long as you don't distribute the stuff. That's the
> > story with G.729, with nVidia
Just on the off chance that Canadian Asterisk users might be
interested in a place to discuss topics specific to the "great
white north" (sources, services, telcos, etc.), I created
the asterisk-canada mailing list:
http://lists.syonex.com/mailman/listinfo/asterisk-canada
or
[EMAIL PROTECTE
How do I solve the problem with between patterns:
_1800
_1NXX
I would like all numbers 1800, 1877 etc to go through iaxtel
but all other numbers 1xxx via voipjet
Example in my extension.conf I have:
[iaxtel]
exten => _1700NXX,1,Dial(IAX2/:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten => _18
Hi,
I have an external Motorola BitSufR Pro ISDN modem and an
ISDN BRI line. Is that possible to get this to work with Asterisk for dial in/out?
Somebody ever did this? Where should I start?
Thanks in advance,
Stephane
_
polycom is better for the same quality and lower price.
On Sun, 16 Jan 2005 17:27:20 -0800 (PST), Robert Augustyn
<[EMAIL PROTECTED]> wrote:
> Any preferences?
> And why?
> Thanks in advance.
> robert
> ___
> Asterisk-Users mailing list
> Asterisk-Users
I have in iax.conf
register => name:[EMAIL PROTECTED]
but I can not make a call, it hangs up on me.
How can I check if I'm registered with iaxtel?
What do I have to have in iax.conf in order to register?
--
#Joseph
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Ast
> Joseph,
>
>
> 1 - 0.9 still uses IAX2 (I think - pretty sure).
> 2 - Why are you using 0.9?
>
> Maybe they should be the other way around...
I'm just using default installation whatever Gentoo is providing; this
is their stable version.
--
#Joseph
___
> Making asterisk work through NAT is a pain and some of the Wiki
stuff> is wrong/out dated. This works for me:Please
feel free to fix or point out what is wrong/outdated so someone else can
fix.
Thanks.
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Aster
Hello All, I have Vonage and Lingo and like the service, but I would like
to drop there ATA equipment. I tried BroadVoice had them for less then
24hrs.
Anyways I would like to connect Asterisk directly to a VoIP provider
without the use of there ATA equipment.
Thanks, David
--
This message has
You probably want to use IAX to talk to FWD. It tunnels through NAT
without any special changes. See
http://www.fwd.pulver.com/advanced/iax
Making asterisk work through NAT is a pain and some of the Wiki stuff
is wrong/out dated. This works for me:
In sip.conf:
localnet: 192.168.1.0/255.255.25
Joseph wrote:
I'm still using asterisk v. 0.9 I think it is using IAX1 as the ixa.conf
is set port=5036
Can I register with a provider who is using IAX2 ?
When I set it up and run:
iax2 show registry - it is not displaying any registered provider.
Joseph,
1 - 0.9 still uses IAX2 (I think - pret
Joseph wrote:
When loading iax.conf I get warning:
WARNING[884753]: chan_iax2.c:5631 set_config: Ignoring port for now
In iax.conf I have:
[general]
port=5036
Joseph,
You are probably going to want to change that to 4569 anyways...
--
Kristian Kielhofner
__
Sounds like you've got a problem with your microphone. I got my
SPA-841 a while ago and the microphone works just fine. I don't have
to scream into it.
I like the phone a lot. I agree with you that the buttons have a
somewhat odd feel. They're sort of rubbery and don't slide like
plastic ones, bu
On Wed, 2005-01-05 at 16:01 -0800, Michael Swan wrote:
> Hi all,
>
> I've struggled for several days trying to get a Digium TDM04B 4-port
> wxfco card working on a Dell 1U PowerEdge 750 machine running
> Fedora Core 1. I finally got a call back from Digium who indicated that
> there is a fundament
Well I just need to say I got my phone last week.
Here is my quick review of the phone and hope that someone has a possible fix
for it or I will be sending it back.
First the phone is nice looking in my view and it's
heavy so it feels like a real desk phone. But it has these stick, gummy o
I found this configuration file on Wiki for FWD behind firewall
; SIP Configuration for Asterisk
;
[general]
disallow=all
allow=ulaw
port=5060 ; Port to bind to
bindaddr=0.0.0.0; Address to bind SIP channel to
externip=xxx.xxx.xxx.xxx
localnet=172.16.1.0
loca
We would like to know if there is a way to broadcast (in realtime) a
conferance. We hold large phone conferances
and would like to know if we could have some of our users listen over a
streaming services. Formats we have looked at include: Shoutcast,Real
Networks,QuickTime, and dare I say Windo
What would be my best option to receive calls via VOIP.
I would like to use it as an alternative number when my main number is
busy.
The solution is not that easy as in order for customer to be a free call
DID=Direct Inward Dialing provider would need to be a local company, I
think. Correct my a
Any preferences?
And why?
Thanks in advance.
robert
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I've heard problems with the Grandstream G729 and the new digium G729
by MAC ID. Could be a compatibility issue with the implementations.
Did you ever use the Grandstream against asterisk with the old
Voiceage G729? I've heard that works just fine.
-- William
__
Joshua Colp wrote:
This is person normally and it is NOT AN ERROR.
:)
Dats grate england you have they're...
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk New
I am afraid i do not have a solution for you, but we also had this problem
occur, exactly the same. It happened overnight, with no changes to the server.
With help from our IAX provider, we did many tests, no solution, we then moved
to a SIP connection to our provider, problem solved.
Our * s
Hi, can't get
X100P (fully zapata compatible clone) to work (I'm in Australia).*
recognises the card and the channel (1) but has definetely some problems
talking to the pots line.I set up this simple dialplan for ZAP
("incoming" context, as setup in zapata.conf, for channel
1)[incoming]exte
This is person normally and it is NOT AN ERROR. It just states that it's
ignoring the port. Simple as that? Okay? Okay? Everyone repeat after me:
WARNINGS ARE NOT ERRORS. Thank you.
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sen
When loading iax.conf I get warning:
WARNING[884753]: chan_iax2.c:5631 set_config: Ignoring port for now
In iax.conf I have:
[general]
port=5036
--
#Joseph
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On Sun, Jan 16, 2005 at 01:33:54PM +0100, Roy Sigurd Karlsbakk wrote:
>
> You can modify and/or link to GPLed code with commercial code and get
> away with it as long as you don't distribute the stuff. That's the
> story with G.729, with nVidia drivers etc etc etc
>
I suppose it's even possible
http://www.sjlabs.com/sjp.html
SJphone® is
a VOIP softphone that allows you to speak with any
PC, PDA, stand-alone IP-phone and with any legacy
wired or mobile phone (using your VOIP gateway or purchasing service from
Internet Telephony Service Provider). It supports both SIP and H.323
I'm still using asterisk v. 0.9 I think it is using IAX1 as the ixa.conf
is set port=5036
Can I register with a provider who is using IAX2 ?
When I set it up and run:
iax2 show registry - it is not displaying any registered provider.
--
#Joseph
___
As
Andres,
Thanks for your answer, but as you can see in the output from show
translation in my original post my Asterisk DOES have G729 support.
Also the fact that softphones work but the Grandstream does not work
stumbles me.
Rene Kluwen
Chimit
- Original Message -
From: "Andres" <[EMAIL
Any suggestions about what I can change to make this work?
Yes, you should get a G729 license for your Asterisk.
Cheers!
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To U
Hi Michael
On 16 Jan 2005, at 20:22, Michael Johnston wrote:
Currenly the inbound lines do not have callerid on them so callerid=no
in my zapata.conf file. What happens on inbound calls is that the SIP
extensions are dialed but their callerid shows '[EMAIL PROTECTED]:X.com'.
Does anyone kno
The the 'common' factor here appers to be the Intel E7520 Chipset.
I have a NEC 120Rg-2 here with this chipset with the same problem.
This chipset exists in the HP DL380 G4 Server, and the machine
mentioned below.
Someone else mentioned the same issue on a new Dual Xeon EM64T
capable Tatu
On Sun, 2005-01-16 at 16:52 -0500, Steve Kann wrote:
> If the delay goes down after a couple of minutes after the transfer,
> this could be the problem.
Just fyi, this is what I observed with those delays between iaxcomm
and firefly, i.e. they occurred on a transfer attempt and normalized
after
On Jan 16, 2005, at 2:53 PM, Dan wrote:
Hi Steve,
- Original Message - From: "Steve Kann" <[EMAIL PROTECTED]>
On Jan 14, 2005, at 2:03 PM, Dan wrote:
Hi,
\> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> I dont have problems when calling PSTN extensions, and calling
> VoceMail, EchoTest, etc. Th
Tomorrow (monday) I will post my kernel oops messages together with my dmesg to
Junghanns.
I have noticed I cannot use the init.d script more then 1 or 2 times before the
server dies completely. Prolly cause of the half unloaded module.
Remco: I have simply connected the 2 NT1 boxes with a cat5 u
On Wed, 12 Jan 2005, Wilson Pickett wrote:
Any chance to post a small how to with the correct server settings et al?
I'm replying to the list, that what it's for.
First of all, I think the big problem wasn't configuring asterisk but
getting the username and password.
To do this you'll need to set u
I'm looking for a company that offers Guatemala DID's. I saw that Lingo does,
but Lingo isn't easily compatible w/ Asterisk, so they're a last resort.
Thanks in advanced, Phil Astin.
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I am using chan_sccp on bristuffed asterisk (0.2RC3 on asterisk 1.0.3).
Things seem fine but I am seeing some weird stuff. I have a Kirk IP600
connecting to * with 2 handsets.
The weird thing is that for incoming calls the handset that is put second
as my dialstring, never rings.
This is my dia
Are you almost done sorting the files?
- Original Message -
From: "Rob Fugina" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
; "Asterisk Developers Mailing List"
Sent: Thursday, January 13, 2005 12:19 PM
Subject: [Asterisk-Users] Status of latest round
I had exactly the same issue with the newest card I got. I tried it with
Zaptel drivers from CVS HEAD and the problem disappeared. It could be
that older drivers don't work with the latest cards.
Mark wrote:
Do you have your zaptel drivers set to start when the system is rebooted?
If not, try re
On Sat, 15 Jan 2005, Begumisa Gerald M wrote:
> > Yup, I found their support very unhelpful and unwilling to go the
> > extra (or even the first) mile..
>
> Might ACPI (not APIC) have anything to do with this condition? I once had
> a hard time with a bunch of cards which were not
On Fri, 14 Jan 2005, Steve Hanselman wrote:
> Has anyone also logged a support call with Digium, it has to be either the
> card, Linux or the Zaptel drivers.
>
Yes of course - we have a call open.
Steve
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On Sun, 16 Jan 2005, Marc Storck wrote:
> how can I read the PRI type of number:
>
> [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan
> E.164/E.163) (1)
> < Presentation: Presentation allowed of network provided number (3)
> '061706161' ]
>
> (in this case TON = 2)
>
> D
Hello,
how can I read the PRI type of number:
[ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan
E.164/E.163) (1)
< Presentation: Presentation allowed of network provided number (3)
'061706161' ]
(in this case TON = 2)
Does a variable like ${TON} exist??? Or how can i read th
I had the same issue. did you ever find a solution. The Fritz card
worked fine with FC2, but no go with FC3, I think it has to do with
udev.
On Thu, 06 Jan 2005 19:36:17 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote:
>
> Though you probably won't use them, I'd still like to mention fyi th
I have a number of inbound analog lines connecting through Digium cards to an
Asterisk box.
Asterisk then bridges the calls over to the internal extensions which are all
SIP phones.
Currenly the inbound lines do not have callerid on them so callerid=no in my
zapata.conf file. What happens o
Hi Steve,
- Original Message -
From: "Steve Kann" <[EMAIL PROTECTED]>
On Jan 14, 2005, at 2:03 PM, Dan wrote:
Hi,
\> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> I dont have problems when calling PSTN extensions, and calling
> VoceMail, EchoTest, etc. The problem is related with the
conversa
I solved this issue. DIP switches marked Option A & Option B need to be
off (down).
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320 ext 2010
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Richard Cook
> Sent: Sunday, January 16, 2005
On Jan 14, 2005, at 2:03 PM, Dan wrote:
Hi,
\> Em Sex 14 Jan 2005 16:43, Dan escreveu:
> I dont have problems when calling PSTN extensions, and calling
> VoceMail, EchoTest, etc. The problem is related with the
conversation
> between two DIAX Softphones.
Between 2 DIAX phone and the delay is in o
Hi all,
I am sure there is a way to get a Multitech MVP110 working with * in
H.323 mode. I have just not been able to figure out how from the MVP110
side. Could someone please share their config setup with me for the
MVP110 and the * side??
TIA,
Robert Webb
__
Do you have your zaptel drivers set to start when the system is rebooted?
If not, try rebooting and issue the "modprobe zaptel" and "modprobe wctdm"
commands to manually start them. You could also issue the "lsmod" command
after a reboot to see if zaptel and wctdm are running. I had problems with
Hi
list,
I was just
wondering, is there any H.323 soft-phone that can be installed on a pocket PC
(iPAQ).
Walid
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To UNSUB
> Subject: Re: [Asterisk-Users] CAC Channel Bank I - FXS
>
> On January 15, 2005 03:39 pm, [EMAIL PROTECTED] wrote:
> > I have a CAC Channel Bank I with FXS cards. I've the system up and
> > running, with just 1 issue.
>
> > When I make an inbound call, Asterisk says "Zap/26 is ringing",
> >
Googling the archives there is some debate about what
are good analog phones to use with *. Aastra seems
popular, but they are somewhat pricey and the
proprietary seems like it can be a headache. Can
someone weigh in on what would be good analog phones
for a small office (8 lines and 20 phones) t
Just in case anyone else has this problem, I'll list my solution:
The latest CVS stable version (either zaptel or asterisk CVS) seemed to
be the problem. When I installed 1.0.3, everything worked.
matt
Matthew Henkler wrote:
I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected
t
sorry I copied and pasted from the already posted stuff it should read:
You can try this:
exten => 101/,1,Dial(device,options...)
exten => 101,1,Dial(device,M(acallerid))
exten => 101,2,Voicemail(u${EXTEN})
exten => 101,102,Voicemail(b${EXTEN})
[macro-acallerid]
;assuming that:
; incoming.gsm exis
You can try this:
exten => 101/,1,Dial(device,options...)
exten => 101,1,Dial(device,M(acallerid))
exten => 101,2,Voicemail(u${EXTEN})
exten => 101,102,Voicemail(b${EXTEN})
[macro-acallerid]
;assuming that:
; incoming.gsm exists and says:
; You have an incming call from..
; and options.gsm exists
I have an Asterisk related problem with mutualphone.
I can connect to any number with any softphone that I am using (iaxcomm,
SJPhone, and a few others).
Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to
mutualphone destinations. Other destinations go fine.
A working phone call
Hello All,
I am trying very hard to learn what it takes to set up an VoIP service
that will allow users to download some comunications software possibly
like "SIP Communicator" or something better and using the Asterisk PBX
software.
The problem is that I am a little confused as to what I all I n
Jake Franklin wrote:
Hello,
I've signed up for a NuFone account, and added the following
instructions to my config files per NufFones directinos:
iax.conf
[NuFone]
type=peer
host=switch-1.nufone.net
secret=password
extensions.conf
(under the [default] context)
exten => _1NXXNXX,1,Dial,IAX2/[E
Sorry about the HTML post, I was sending from my laptop and forgot to turn
off html in outlook. Have a nice day.
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 16, 2005 12:50 AM
To: [EMAIL PROTECTED]
Subject: Re: asterisk-users list and html posts
Pl
Aryeh wrote:
Hi all,
I have a Polycom Soundpoint IP 600 that looks like it is fried. It either
has a bad or corrupt bootrom (I'm guessing). It never gets to the prompt
that asks if I want to enter the setup configuration or continue booting.
When plugged into power, it turns on, shows the Polycom
On Sun, 2005-01-16 at 14:49 +0100, Robert Rozman wrote:
> could you please give some more info how to do this ?
Use Custom ring 1 tone with with a blank Caller ID
--
Dave Cotton <[EMAIL PROTECTED]>
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Hi all,
I have a Polycom Soundpoint IP 600 that looks like it is fried. It either
has a bad or corrupt bootrom (I'm guessing). It never gets to the prompt
that asks if I want to enter the setup configuration or continue booting.
When plugged into power, it turns on, shows the Polycom logo for 3
I'm experiencing a similar issue, but it's with IAX2 / IAX2 calls. I've
started to think that it's a router or something upstream. For me, if I
keep the call bridged through asterisk (notransfer=yes), after about a
minute of conversation, the called party can't hear the caller. I watched
the tr
Chris Polk wrote:
Any one have any solution for this?
We need to have the caller id information announced when the phone is answered.
for example
I am sitting at my desk, my phone rings.
I pick it up and hear call from 55 to except press 1 to decline press to
any help would be grately app
Thanks! Thanks! Thanks!
I've got it work!!! :-)
Message: 13
Date: Sun, 16 Jan 2005 12:17:21 -
From: "Bill Seddon" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] failed to compile zaptel
on redhat
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
Message-ID:
<[EMAIL P
i just checked out the digium site, but its a bit expensive and i'll end up
w/ two fxo modules that i'll never need. if anybody would be interested in
swapping two fxs modules for two fxo's it would be a great help, please
contact me offlist.
thanks,
jon
- Original Message -
From: <[EM
> doh! i assumed the x100p and TDM400p worked the same, because i thought was
> able to do both on that card...well thanks for the help :(
Side note : you just have to get 2 FXS modules for your TDM400, the
card can use FXO or FXS modules, and you can mix them as you wish
__
doh! i assumed the x100p and TDM400p worked the same, because i thought was
able to do both on that card...well thanks for the help :(
-jon
- Original Message -
From: Henry Devito
To: Asterisk-Users@lists.digium.com ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Sunday,
try iax2 debug
> Hello,
>
> I've signed up for a NuFone account, and added the following
> instructions to my config files per NufFones directinos:
>
> iax.conf
> [NuFone]
> type=peer
> host=switch-1.nufone.net
> secret=password
>
> extensions.conf
> (under the [default] context)
> exten => _
On incoming calls it seems that * is finding the callerid correctly but my
BudgeTone is not showing it in the display.
What am I doing wrong?
The * console shows:
-- Accepting call from '6' to '6' on channel 0/1, span 1
(numbers changed)
but I guess that'c correct?
__
Hi
My card is working, but when I reboot the machine, most of the times
it is not working,
I get "ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6)"
To make it work again I have to shut down, remove the card, reboot so
kudzu will remove the config. shut down again, put the
Hi,
- Original Message -
From: "Wilson Pickett" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, January 16, 2005 10:20 AM
Subject: Re: [Asterisk-Users] announcing caller id?
> > Any one have any solution for this?
> > We need to have the
I can see that Asterisk supports TDD
text channels, as there is a tdd.c file in the source, this appears to
be exposed via TDD MODE in the AGI interface?
I cant find any documentation anywhere
on how to use this. Has anyone done this? tdd.c seems to only support 45.5
baud calls.
Ideally I'd like
Erm, at the risk of getting flamed, where does IAX come into this
picture? If I re-implement IAX(2) in a different language (not using
iaxcomm except as a refererence or test ) and want to sell a product
based on it can I do that, or do I need a license ?
You are probably ok without a comercial lic
> >>I have done my homework on this, I hope.
> >>
> >>I have a customer with an ATA186 who uses Nufone as his IAX provider.
> >>His network operations center in the Bahamas was destroyed by the
> >>hurricanes, and I'm helping him rebuild.
> >
> >
> > I can help, but I think it might require bei
Soren, thanks for the information and advice.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje
Sent: January 14, 2005 1:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zaptel in HEAD
<< then it looks like the does not have the full kernel sources installed>>
...or isn't running the 2.6 kernel.
I had the same problem with the CVS on Friday (but not from the week
before). It turns out that moduleparam.h is included as part of a bug fix
on 2.6 but instead of being #ifdef'd fo
My problem used to come and goe without knowing the
cause and the remedy .But now it is consistent.
Each time I boot my Redhat9, I get the error message
["gnome-sound-recorder" (process #) has crashed due to
fatal error (Aborted)]. After I close the error window
and open the application "Sound Re
Hi,
Does the context defined in sip.conf have to be the same context to which
the extension belongs to in sip.conf?
I have all my local SIP phones in context=local, and are in a context call
local in extensions.conf. I then signed up with a few voip providers and I
only wanted to allow one of the
Hi - I found the trunkfreq directive in iax.conf so I've put the directive
line into the iax peers section (along with "trunk=yes") - I'm sure you
meant iax.conf rather than zapata.conf ?
Thanks for the help,
Derek
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
On Fri, 14 Jan 2005, Rich Adamson wrote:
Are there security concerns with the * application software?
I know there are with the Linux installation.
:-)
You should always be concerned with security. Not to say that Asterisk
has any security problems (it is audited regularly).
If you are administeri
On Sunday 16 January 2005 04:29, Steven Critchfield wrote:
> linux/moduleparam.h is actually part of the kernel source. It is created
> when you config and compile the kernel. It holds the version symbols
> needed to properly link the new drivers into the kernel.
No, it is part of the virgin kerne
> Any one have any solution for this?
> We need to have the caller id information announced when the phone is
> answered.
> for example
> I am sitting at my desk, my phone rings.
> I pick it up and hear call from 55 to except press 1 to decline
The Grandstream BT100 series phones will
> -- Call accepted by 66.225.202.72 (format gsm)
> -- Format for call is gsm
> -- Hungup 'IAX2/NuFone/1'
>== No one is available to answer at this time
Is the callerid a number like 7073131 ?
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Any one have any solution for this?
We need to have the caller id information announced
when the phone is answered.
for example
I am sitting at my desk, my phone rings.
I pick it up and hear call from 55 to
except press 1 to decline press to
any help would be grately appreciated!
On Sun, 16 Jan 2005, David Norton wrote:
> If you know the brand of wireless access point, you are able to try find a
> repeater or bridge for that particular brand. Eg. If they using dlink
> equipmenk, the DWL2100 can act as a repeater for it. Quite a few brands of
> wireless equipment will only
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