I am unable to dial out over a Wildcard TDM400P. This was working previously,
so must have
messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX
2.5.2.2.
When I dial, I see:
-- Executing [...@macro-dialout-trunk:19] Dial(DAHDI/1-1,
DAHDI/g0/9239220,300,) in
new stack
Andy Howell wrote:
I am unable to dial out over a Wildcard TDM400P. This was working previously,
so must have
messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX
2.5.2.2.
When I dial, I see:
-- Executing [...@macro-dialout-trunk:19] Dial(DAHDI/1-1,
DAHDI/g0
I also see this from time to time. Running:
Asterisk CVS-Nv1-0-9-09/02/05-14:07:10 built by [EMAIL PROTECTED] on a i686
running Linux
Setup a little cron job that, in the wee hours, does a:
service asterisk restart
and then * will start answering the lines again. Not ideal,
Kevin P. Fleming wrote:
Andy Howell wrote:
I have a weird problem in which my digium card stops answering. After
running for a couple days, incoming calls are not seen. Running asterisk
-r shows no incoming calls. Restarting Asterisk does not help. After a
reboot it is fine.
This problem
Leonardo Gomes Figueira wrote:
Hi,
Andy Howell wrote:
I have a weird problem in which my digium card stops answering. After
running for a couple days, incoming calls are not seen. Running asterisk
-r shows no incoming calls. Restarting Asterisk does not help. After a
reboot it is fine.
Any
I have a weird problem in which my digium card stops answering. After
running for a couple days, incoming calls are not seen. Running asterisk
-r shows no incoming calls. Restarting Asterisk does not help. After a
reboot it is fine.
Any ideas?
Thanks,
Andy
I'm trying to set a variable in one context and use it in another:
[c1]
MY_GROUP = SIP/200SIP/201
[c2]
include c1
exten s,1,NoOp(${MY_GROUP})
The noop prints out a blank string. In [c1], I've tried
exten s,1,SetVar(MY_GROUP=SIP/200SIP/201)
and
exten s,1,SetVar(_MY_GROUP=SIP/200SIP/201)
and
Rich Adamson wrote:
It is just sending a sip invite to [EMAIL PROTECTED] Does the
X-Lite need to connect to via a proxy?
No. You should work on configuring xlite to register with asterisk.
Thanks. I can get it to work that way. What I was trying to simulate was
an external user calling in.
Hello,
I'm trying to place a call to asterisk using X-Lite. Asterisk is setup
with some Grandstream phones. I can call from one grandstream extension
to another. When I try to an extension with X-Lite, it comes back with
Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk