Hi,
I am just testing Asterisk with a softphone on a fedora box until my ip phones
arrive
and have a basic config so far. I am a bit confused over how to setup the
inbound.conf
file now. It appears as if outbound and demo works so far.
Any hints would be greatly appreciated!
jlc
My sip.conf is as follows:
[general]
context=default
allowguest=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
[100]
type=friend
context=default
disallow=all
allow=ulaw
host=dynamic
username=100
secret=100
register => username:password at sip.domain.com:5060
register => username:password at sip.domain.com:5060
[carrier-sw1]
context=default
type=friend
host=sip.domain.com
username=username
secret=password
canreinvite=no
insecure=port,invite
qualify=5000
dtmfmode=auto
nat=no
disallow=all
allow=ulaw
[carrier-sw2]
**similar to above**
My extensions.conf is as follows:
[general]
static = yes
writeprotect = no
clearglobalvars = no
[globals]
CONSOLE = Console/dsp
[local]
include => default
#include outbound.conf
#include inbound.conf
[default]
include => demo
include => inbound
include => outbound
[demo]
exten => _700,1,wait(1)
exten => _700,2,answer()
exten => _700,3,wait(1)
exten => _700,4,playback(tt-weasels)
exten => _700,5,hangup()
[macro-dialxxxx]
exten => s,1,Dial(SIP/${ARG1}@xxxx-sw1,120)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-ANSWER,1,Hangup
exten => s-CONGESTION,1,Dial(SIP/${ARG1}@xxxx-sw2,120)
exten => s-CONGESTION,2,Goto(ss-${DIALSTATUS},1)
exten => s-CANCEL,1,Hangup
exten => s-BUSY,1,Busy(30)
exten => s-CHANUNAVAIL,1,Dial(SIP/${ARG1}@xxxx-sw2,120)
exten => s-CHANUNAVAIL,2,Goto(ss-${DIALSTATUS},1)
exten => ss-ANSWER,1,Hangup
exten => ss-CONGESTION,1,Congestion(30)
exten => ss-CANCEL,1,Hangup
exten => ss-BUSY,1,Busy(30)
exten => ss-CHANUNAVAIL,1,Congestion(30)
My outbound.conf is as follows:
[outbound]
; NANPA
exten => _1NXXNXXXXXX,1,Macro(dialxxxx,${EXTEN})
; International Calling
exten => _011.,1,Macro(dialxxxx,${EXTEN})
On Sun, May 18, 2008 at 5:17 PM, Joseph L. Casale <JCasale at activenetwerx.com> wrote:> Hi, > I am just testing Asterisk with a softphone on a fedora box until my ip phones arrive > and have a basic config so far. I am a bit confused over how to setup the inbound.conf > file now. It appears as if outbound and demo works so far. > > Any hints would be greatly appreciated! > jlc > > My sip.conf is as follows: > [general] > context=default > allowguest=yes > disallow=all > allow=ulaw > dtmfmode=rfc2833 > canreinvite=no > > [100] > type=friend > context=default > disallow=all > allow=ulaw > host=dynamic > username=100 > secret=100 > > register => username:password at sip.domain.com:5060 > register => username:password at sip.domain.com:5060 > > [carrier-sw1] > context=default > type=friend > host=sip.domain.com > username=username > secret=password > canreinvite=no > insecure=port,invite > qualify=5000 > dtmfmode=auto > nat=no > disallow=all > allow=ulaw > > [carrier-sw2] > **similar to above** > > My extensions.conf is as follows: > [general] > static = yes > writeprotect = no > clearglobalvars = no > > [globals] > CONSOLE = Console/dsp > > [local] > include => default > #include outbound.conf > #include inbound.conf > > [default] > include => demo > include => inbound > include => outbound > > [demo] > exten => _700,1,wait(1) > exten => _700,2,answer() > exten => _700,3,wait(1) > exten => _700,4,playback(tt-weasels) > exten => _700,5,hangup() > > [macro-dialxxxx] > exten => s,1,Dial(SIP/${ARG1}@xxxx-sw1,120) > exten => s,2,Goto(s-${DIALSTATUS},1) > exten => s-ANSWER,1,Hangup > exten => s-CONGESTION,1,Dial(SIP/${ARG1}@xxxx-sw2,120) > exten => s-CONGESTION,2,Goto(ss-${DIALSTATUS},1) > exten => s-CANCEL,1,Hangup > exten => s-BUSY,1,Busy(30) > exten => s-CHANUNAVAIL,1,Dial(SIP/${ARG1}@xxxx-sw2,120) > exten => s-CHANUNAVAIL,2,Goto(ss-${DIALSTATUS},1) > exten => ss-ANSWER,1,Hangup > exten => ss-CONGESTION,1,Congestion(30) > exten => ss-CANCEL,1,Hangup > exten => ss-BUSY,1,Busy(30) > exten => ss-CHANUNAVAIL,1,Congestion(30) > > My outbound.conf is as follows: > [outbound] > ; NANPA > exten => _1NXXNXXXXXX,1,Macro(dialxxxx,${EXTEN}) > ; International Calling > exten => _011.,1,Macro(dialxxxx,${EXTEN}) >Attach to the Asterisk console and try making a call that usually fails with verbose set to 3 or so and post the output. It is probably something very simple. Your includes are probably the issue. Try bringing them all into extensions.conf and see if it works. Thanks, Steve Totaro
>Attach to the Asterisk console and try making a call that usually >fails with verbose set to 3 or so and post the output. It is probably >something very simple. > >Your includes are probably the issue. Try bringing them all into >extensions.conf and see if it works. > >Thanks, >Steve TotaroYup, I brought them in and its all working smooth now except there is a mis-configuration at my provider I am going to have to wait to get resolved. What about those includes was sketchy? Is that bad practice to separate them out? Thanks! jlc
Joseph L. Casale wrote:>>Attach to the Asterisk console and try making a call that usually >>fails with verbose set to 3 or so and post the output. It is probably >>something very simple. >> >>Your includes are probably the issue. Try bringing them all into >>extensions.conf and see if it works. >> >>Thanks, >>Steve Totaro >> >> > >Yup, I brought them in and its all working smooth now except there is a >mis-configuration at my provider I am going to have to wait to get resolved. > >What about those includes was sketchy? Is that bad practice to separate them out? > >Thanks! >jlc > >I use includes extensivly, and I provide phone service to several of my friends, the order you have them in can cause problems somtimes. That has been my experiance, for what its worth. -Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080519/c99183b1/attachment.htm