Dear All,
my client wants a asterisk pbx with 30 FXO 30 FXS analogue ports, please
suggest if sangoma A400 is a good option for that. Also please suggest
server hardware.
regards,
Umair
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Hello Sam,
use host=IP_ADDRESS when defining user in sip.conf
regards,
Umair Bari
On 1/26/06, Sam Tam [EMAIL PROTECTED] wrote:
Do anyone know how to setup asterisk to authenticate the user through IPrather than username and password?
I know most carriers will do that but smaller end user
Dear Michiel,
Would you be kind enough to put more light on RAND stuff. How you do the load balancing.
Regards,
Umair Bari
On 2/4/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On 04:47, Sat 04 Feb 06, Joseph Tanner wrote: This is probably a stupid question, but how do you specify multiple
Hello Gabriel,
IMHO, using IAX between * servers is a good choice, I dont see any problem in it. Actually I used it for sometime and never encounter any issue, but i had max 5 concurrent connections.
regards,
Umair bari
On 3/7/06, Gabriel Afana [EMAIL PROTECTED] wrote:
Hi everyone, I just spend
don't scare people for GOD sake :)
LOL
On 7/4/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
Think about voicesense which will sense what you are talking and pop in a
*relevant* voice ad to spice up conversation :P .
On 03/07/07, Dean Collins [EMAIL PROTECTED] wrote:
Ooops did Google just
in Advance,
Regards,
Umair bari
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Please confirm if PRI span is up
on CLI, type pri show span 1 it must be UP before you can dial through it.
regards,
Umair bari
On 8/25/05, root linux [EMAIL PROTECTED] wrote:
Yep, I am connecting to some other equipemnt...its a
Clarent gateway equipped with a National Microsystems
(Quad
Using RH 9 with *
Regards,
Umair Bari
David Choo wrote:
We used gentoo internally. I also have * running on CentOS, RHEL.
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
"Engineered for Changing Businesses"
Espore Corp
try putting
exten = _0.,4,Hangup
like
[ext-local-custom]
exten = _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten = _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten = _0.,3,Congestion
exten = _0.,4,Hangup
regards,
Umair bari
Tomasz Chmielewski wrote:
I'm trying to learn Asterisk.
So far I'm using kphone
exten = 1234,1,Dial(SIP/1234,Number_of_Sec_for_Ringing,tr)
Tim P wrote:
Maybe this marks me as a real newb but where do I set the number of
rings that a phone has before it sends it to voicemail?
Also for some odd reason when I ring an extension attached to my
sipura 2100 ATA it takes it
the process, then start over.
Please suggest what to do.
regards,
Umair bari
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Michael,
try relaxdtmf=yes in your iax.conf, or if you are using sip, then in
sip.conf
regards,
Umair bari
Michael Stearne wrote:
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
That's entirely possible. Had something similar with livevoip.com (with
the answered iax call issue
Hello,
Try http://www.freepbx.org, its written in PHP with mysql at the end, it also uses .conf files for configurations.
regards,
Umair bari
On 5/8/06, moona ather [EMAIL PROTECTED] wrote:
Hi,I have to make a web-based management interface of configuring asteriski wanted to know
freepbx has been improved since then, and I believe if you edit/add something in original asterisk .conf files, it stays there. I've tried it long ago when it was called AMP and it worked.
regards,
Umair Bari
On 5/9/06, Emmo ather [EMAIL PROTECTED] wrote:
Hello,In older version of freebpx if you
Hello,
IMHO, there are 2 ways to do this,
1) You can connect your VoIP modem to your asterisk box using x100p FXO card, you'll need to get one and install it properly.
2) Get SIP/IAX account from any VoIP provider and use it with asterisk.
Hope this helps.
Regards,
Umair Bari
On 5/12/06
No, i really dont think so,
here are few lines from extensions.conf.sample.
; Extension names may be numbers, letters, or combinations; thereof. If an extension name is prefixed by a '_'; character, it is interpreted as a pattern rather than a; literal. In patterns, some characters have special
Trywindows messenger 5
http://www.microsoft.com/downloads/details.aspx?FamilyID=16F3A735-FE18-4DF8-9A19-5C6C721CE715displaylang=en
Regards,
Umair Bari
On 11/18/05, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,I've found quite some docs on this, but many of them deprecated...I'm curious what
http://www.microsoft.com/downloads/details.aspx?FamilyID=a8d9eb73-5f8c-4b9a-940f-9157a3b3d774DisplayLang=en
sorry about that link, that was a doc. try the link above.
regards,
Umair
On 11/18/05, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,I've found quite some docs on this, but many of them
I think that delay in answering is due to caller ID detection.
I have no idea about rest of your question :)
regards,
Umair
On 11/18/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi. I'm a new user of Asterisk. My question is:I want to log outbound calls in a database ( postgres ). Everything
if you are dialing out from zap channel, you can set the caller id with before dialing out and then you can dial like,
exten = Dial(Zap/2/${EXTEN},15)
and if you are ringing an analog phone connected to fxs port, then you can set the caller id before dialing, then you can dial it like,
exten =
know if above method will work or not, I wrote what I think can solve the issue.Regards,
Umair bari
On 11/20/05, Abdul Lateef Khan [EMAIL PROTECTED] wrote:
Hi,I already install the agiphp from the following steps, i want to besure, is my agiphp installation is correct or not.
i copied all following
look for intel ambient MD3200 chipset in those modems, there were available here in pakistan a year back but now they are vanished from the market. I've been usingmodem with intel ambient md3200 chipsetwith asterisk and they work fine.
regards,
Umair bari
On 11/24/05, Kunhikrishnan, Salil
accurate. Also, it is ONLY configured for; standard U.S. tones;busycount=4
regards,
Umair bari
On 11/24/05, Faris Raouf [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote: Hello everybody:-)
This are my first line french zapata.conf settings. I have 3 like this, with only rx/tx gain a little bit
Let me know the cost.
regards,
Umair bari
On 8/31/05, Chris A. Icide [EMAIL PROTECTED] wrote:
In the next week to two weeks I'll be posting some informationconcerning a system I've been designing.It currently does three layer
hosted VoIP pbx services as well as hosted ITSP services (the model
try bindport=5062 and bind the IP address too
bindaddr=IP_ADDRESS
On 9/5/05, Aisling [EMAIL PROTECTED] wrote:
Hello,
Hope somebody can help me – Asterisk is behaving very oddly and I'm totally stumped! I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is)
for taking calls on broadvoice numbers, you need to put insecure=very and need a context in extensions.conf to handle incoming calls on the EXTEN you put in register statement in sip.conf
also in sip.conf, put user or friend instead of peer.
regards,
Umair bari
On 9/6/05, Bartosz Wegrzyn
to plug your telco or vonage line into ast box.
yes thenyour friend can use your vonage or telco line to dialout, and you can make dialplan which actually let your friend chose how to dial and from which line.
regards,
Umair bari
On 12/15/05, Ugo Bellavance [EMAIL PROTECTED] wrote:
Hi,I think
in your sip.cong [general] contexts
put
disallow=all
allow=ulaw
allow=alaw
and in your sip user, use disallow only ONCE, that is
disallow=all
allow=ulaw
allow=alawhope this helps.
regards,
Umair bari
On 12/15/05, Jason Chan (jasonOfficial) [EMAIL PROTECTED] wrote:
Hi there,I am writing
Dear Talat,
if you are trying to connect from within your LAN,
put nat=no and then try againregards,
Umair bari
On 12/14/05, Talat Ishtiaq [EMAIL PROTECTED] wrote:
-- Forwarded message --From: Talat Ishtiaq
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial
506045691-2
On 12/15/05, Pablo Allietti [EMAIL PROTECTED] wrote:
hi all. i have my asterisk with a 192.168.0.1 addresswhich ports i need to forward in my firewall to connect remote xten
clients and make calls?thsnk--.-___--Bandwidth and
its build into the codes,
IMHOfor replacing * with ## you need to hack asterisk source code, I cant think of anyother way.
regards,
Umair bari
On 12/15/05, Obelix [EMAIL PROTECTED] wrote:
I want to use '##' to terminate a call instead of the '*' used by the Dialcommand's H option
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Thanks Regards,
Umair Bari
,
Umair Bari
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