asterisk to version 1.2.4
Are you using OH323 or H323 ?
I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323
and everything worked fine.
Cheers,
Oliver
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Mejía Evertsz
Sent
exten = _sipuserX.,1,blah
to match sipuser01, sipuser99... ?
or
exten = sipuser01,1,blah
to match sipuser01 only ?
Not to usefull when you want to match domain also :S
Eg [EMAIL PROTECTED]
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jon-o Addleman
Nop.
. matches one or more from the previous carácter
More info...
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Eric
ManxPower Wieling
Enviado el: Friday, April 21, 2006 2:36 PM
I'm running:
OS: FreeBSD 6.0
Asterisk: 1.2.4
Installing OH323:
0.7.3
I have this error
when compiling
chan_oh323.c: In
function `reload_config':
chan_oh323.c:4677:
warning: implicit declaration of function `sscanf'
chan_oh323.c: At
top level:
chan_oh323.c:3244:
warning:
One billing solution that Works with your CDRs is AsterBill
(www.cybexdev.com) for postpaid. (not opensource, you have to buy)
It runs a cronjob to get the latest CDRs and bill each accountcode.
For prepaid of course it does it with AGI.
-Mensaje original-
De: [EMAIL PROTECTED]
Hello.
I have a codec problem to send calls from a SIP device to a H323 gateway.
First I'll explain the scenario:
- Asterisk 1.2.1
- The SIP phone can use any codec I want.
- The H323 gateway can only use g729 (cause it's not under my
administration)
- SIP phone has g729 configured, so my
1.2.4
Are you using OH323 or H323 ?
I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323
and everything worked fine.
Cheers,
Oliver
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Mejía Evertsz
Sent: Thursday, April 20
To which context of the dial-plan does asterisk tries to
match incoming calls when acting as a sip client?
To be more specific:
In extensions.conf Under which context should I place
exten = 648064,1,Dial(TECH/peer)
for an entry like this register = 648064:[EMAIL PROTECTED]/648064 ?
Please help me out with this
To which context of the dial-plan does asterisk tries to
match incoming calls when acting as a sip client?
To be more specific:
In extensions.conf Under which context should I place
exten = 648064,1,Dial(TECH/peer)
for an entry like this register =
Hi list.
Im having problems handling the incoming calls when
using asterisk as a sip client.
For example, on sip.conf I have lines like this:
register = user1:[EMAIL PROTECTED]/777
register = user2:[EMAIL PROTECTED]/888
[host1]
context=hostnumber1
type=friend
insecure=very
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