Nathan Bell wrote:
This is what I get from the asterisk CLI:
ast*CLI> sip show peers
Name/username HostDyn Nat ACL Port Status
202(Unspecified)D 0Unmonitored
201(Unspecified)D 0
That seems to be the problem. In my dhcp settings I wasn't giving it the
correct domain-name-servers option. I changed that and I changed the
phones to use [EMAIL PROTECTED] instead of [EMAIL PROTECTED] and that
seems to have taken care of it.
Thanks for the help.
Nathan Bell
IT Engineer Du J
Nathan Bell wrote:
> This is what I get from the asterisk CLI:
>
> ast*CLI> sip show peers
> Name/username HostDyn Nat ACL Port Status
> 202(Unspecified)D 0Unmonitored
> 201(Unspecified)D
This is what I get from the asterisk CLI:
ast*CLI> sip show peers
Name/username HostDyn Nat ACL Port Status
202(Unspecified)D 0Unmonitored
201(Unspecified)D 0Unmonitored
2 si
and
sip show users
Noah Miller wrote:
Hi Nathan -
No loop now, but instead I get this:
Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of
type 'SIP' (cause 3 - No route to destination)
Mar 26 15:42:18 VERBOSE[1854] logger.c: == Everyone is busy/
Hi Nathan -
No loop now, but instead I get this:
Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of
type 'SIP' (cause 3 - No route to destination)
Mar 26 15:42:18 VERBOSE[1854] logger.c: == Everyone is busy/congested
at this time (1:0/0/1)
Mar 26 15:42:18 DEBUG[1854] app_dia
No loop now, but instead I get this:
Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of
type 'SIP' (cause 3 - No route to destination)
Mar 26 15:42:18 VERBOSE[1854] logger.c: == Everyone is busy/congested
at this time (1:0/0/1)
Mar 26 15:42:18 DEBUG[1854] app_dial.c: Exitin
nathan,
try dial() directly to the extension
[to-sip]
exten => _X.,1,Dial(SIP/[EMAIL PROTECTED],120)
try
exten => _X.,1,Dial(SIP/${EXTEN},20)
where ${EXTEN} = 201
and
[201] in /etc/sip.conf is
[201]
type=friend; Friends place calls and receive calls
context=from-sip
Sorry, forgot to attach the sip.conf and extensions.conf files. Attached
now.
[general]
context=from-sip; Default context for incoming calls
; if asterisk was compiled with OSP support.
realm=actarg.com; Realm for digest authenticati
