Ben Schorr wrote:
Is there some reason why I keep getting this same message from “cool
dude” over and over and over? And under different subject lines?
I assumed that he was sending it to every post, trying to get a
response. Sorta spammy in my opinion.
Doug
--
Ben Franklin quote:
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
...@rolandschorr.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cool dude
Sent: Friday, February 12, 2010 21:24
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] extension not found
hi friend need ur help in dial plan
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else he'll be
routed to another call flow as you can see in the context below:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ?
Hi Michel,
Am Freitag, den 12.09.2008, 17:41 +0300 schrieb michel freiha:
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else
he'll be routed to another call flow as you can see in the context
Thanks :-D change the context to default and everithing works fine.
I assigned the sip context because that was the context on the example.
Thanks :-)
Nomar
Alex Balashov wrote:
Nomar Mora wrote:
Alex Balashov wrote:
Do you have dial plan routes for internal extension calls?
Dear Randulo,
Thanks for your suggention.
Now i am able to communicate between 2 computers.
Regards,
Baskar
--- randulo [EMAIL PROTECTED] wrote:
On Mon, May 19, 2008 at 8:44 AM, bas karan
[EMAIL PROTECTED] wrote:
[May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879
handle_request_invite: Call
Good day:
I recently install asterisk-now. Setup a pair of SipXpert 160 phones and
all fine. Later I try to setup some Octtels VoIP Gateways SP4220. I
config the proxy setings like this:
http://www.fundacitetachira.gob.ve/settings/Settings.png
the sip.conf entrys are like the asterisk manual
Do you have dial plan routes for internal extension calls?
Nomar Mora wrote:
Good day:
I recently install asterisk-now. Setup a pair of SipXpert 160 phones and
all fine. Later I try to setup some Octtels VoIP Gateways SP4220. I
config the proxy setings like this:
Alex Balashov wrote:
Do you have dial plan routes for internal extension calls?
Do you mean if I have configured the extension.conf? Yes, I config the
extensions on the extension.conf file
otherwise, no I have not.
Thanks in Advance
Nomar
--
2008 Año del satélite Simón Bolívar
Nomar Mora wrote:
Alex Balashov wrote:
Do you have dial plan routes for internal extension calls?
Do you mean if I have configured the extension.conf? Yes, I config the
extensions on the extension.conf file
otherwise, no I have not.
Thanks in Advance
Nomar
In the 'sip' context?
Dear Friends,
This is Baskar from Chennai, trying to configure
asterisk. Now I planned to start with communication
between 2 systems using soft phones.
When I tried to call the other computer I am getting
the following error message on asterisk terminal,
Connected to Asterisk 1.4.18 currently
On Mon, May 19, 2008 at 8:44 AM, bas karan [EMAIL PROTECTED] wrote:
[May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879
handle_request_invite: Call from 'Phone3' to extension
'5' rejected because extension not found.
-- Registered SIP 'Phone3' at 192.168.1.101 port
Extension.conf enteries
Dear Randulo,
Thanks for your replay.
I am new to this concept, Could you explain me little
bit extra please?
Thanks Regards,
Baskar
--- randulo [EMAIL PROTECTED] wrote:
On Mon, May 19, 2008 at 8:44 AM, bas karan
[EMAIL PROTECTED] wrote:
[May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879
On Mon, May 19, 2008 at 11:36 AM, bas karan [EMAIL PROTECTED] wrote:
I am new to this concept, Could you explain me little
bit extra please?
You will need to put extensions in contexts. The context is a
fundamental concept of the dialplan. All extensions are inside a
context. In your sip.conf
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