in your sip.conf
bind it to the ip you want to bind it , change the value from 0.0.0.0 to the ip
u want to
bind it to.
Rehan
Hi, all
Can someone tell me where to tell asterisk no to use 127.0.0.1 IP
(localhost)?
When I am registering with VoIP providers, they get my info as
[EMAIL
Hello,
I need some assistance in
1. When call comes in on the sipura on the pstn, it should come to an x
extention.
2. When I dial a particular/ my local NPA NXX it goes out via the Sipura 3000
Can any one assist.
Rehan
Super Technologies Inc., Pensacola, Florida
http://www.SuperTec.com -
, etc?
regards
On 1/13/06, Rehan AllahWala [EMAIL PROTECTED] wrote:
Hello All,
Anyone here has experience of accepting a ilbc call and sending it
on g711 or g729
I am having problem in VOICE , call goes though but there is no
voice.
Senario:
Call is coming in from Machine
We have a system for transcoding from ilbc to g711
Totally there are 3 machines
A B and C
A is sending the call with user name
B is transcoding it
C is reciving the transcoded call.
When a call comes to user B its getting coverted but changing the user name
from
[EMAIL PROTECTED] to [EMAIL
exten = _X.,1,AGI,catchall.agi,${EXTEN}
should do it for u
Hi all,
What do you think about having a single extension in the dialplan that
matches everything and then delegates the next action to an external
application through AGI? I mean something like this:
exten =
Hello All,
Anyone here has experience of accepting a ilbc call and sending it on g711 or
g729
I am having problem in VOICE , call goes though but there is no voice.
Senario:
Call is coming in from Machine A to Machine B, sending to Machine C
Machine B is an asterisk box, transcoding it from
The guy who wrote it is
Babar Shafiq babarnazmi at yahoo.com
Price i belive is 500$ for it
Babar do comment !
Rehan
I have seen but not used
http://www.silicontechnix.com/webtelefone/start.html
--
--
Steven
May you have the peace and freedom that come from abandoning all hope
Dear All,
For those who celebrate Eid.
I would like to wish you a very Happy Eid Mubarak.
For those who do not know what it is, Its a Prayer in memory of Ishmael son of
Abraham, when he attempted to sacrifice his beloved son on request by god.
Muslim's celebrate it with a sacrifice of a goat
Dear All,
How to register a peer register command in [general] area of sip.conf via
database.
IE i have in my sip.conf
register = 22100140:[EMAIL PROTECTED]:5060/121222100140
Where do i take it in the Database ?
Rehan
Super Technologies Inc., Pensacola, Florida
http://www.SuperTec.com
I think babar wrote a sip version also
Rehan AllahWala wrote:
The guy who wrote it is
Babar Shafiq babarnazmi at yahoo.com
Price i belive is 500$ for it
Babar do comment !
Rehan
It looks very good, however im looking for a sip solution instead of
iax, another option
www.antek.com.tw
Had 4 port fxo, for around 200 to 250$
They are OEM, and can change things if u need.
I tested it breifly in there office last year in Computex 2005
You can contact [EMAIL PROTECTED] for wholesale.
Rehan
On Fri, 2005-12-30 at 17:53 -0800, [EMAIL PROTECTED] wrote:
Does Quintum has a 4 port fxo box ?
Hi,
Not very reliable for commercial setups, they do have issues hanging
up ports etc. Quintum over Antek any day.
Regards,
Sahil Gupta
VoiceValley
On Mon, 2 Jan 2006, Rehan AllahWala wrote:
www.antek.com.tw
Had 4 port fxo, for around
We are using perl for agi
I will try this command
Thank You
Rehan
What are you using for AGI
The correct command to send
Would be:
EXEC Set(${CALLERID(num)}=0005551212)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rehan
Where did you change the port
General Info or the user info
You have to change it in the GENERAL info for it to change the port of asterisk.
Rehan
Hi
I am running asterisk SIP on port 5060, in my sipura i changed the
5060 port to 6060. but it's still tring to register it to asterisk.
Dear Leandro,
I do not think you can avoid the translation here, You must have the licences
to be able
to talk to your provider OR you may want to allow your ip phone to use g729
I do not think there is any other way.
You can try the intel's trial version to see if that works for you, at no
Dear Harry,
What would you like to be debugged ?
Is nxs.yi.org your server ?
Rehan
Hello,
nobody use an ip phone on these mailing lists !
your call will put on queue .
I just need some people to dial sip:[EMAIL PROTECTED] to
check and debug my config .
Regards
Harry
OK Carlos,
Let me know if you need any assistance.
Rehan
We re trying to use 1.2.1. Im finishing compiling a new version, as soon as
I got it Ill let you
know.
Thanks a lot for the intention.
Regards,
Carlos
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]
Do u know how to instert it in the agi ?
$AGI-exec(SetCIDNum(8504338555));
but it didn't work
www.voip-info.org/wiki-asterisk
or you could try the CLI show application Set, and show function
CALLERID
On 12/28/05, Rehan Ahmed [EMAIL PROTECTED] wrote:
Hi
Can you send any
18 matches
Mail list logo
