Sure.
; sip.conf ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
[gateway]
type=friend
host=111.111.111.111
; Ryan@Home
[1011]
type=friend
username=1011
secret=1011
host=dynamic
mailbox=101
context=intern
;canreinvite=no
;dtmfmode=rfc2833
;qualifier=200
nat=1
; Ryan@Office
[1012]
type=friend
username=1012
secret=1012
host=dynamic
mailbox=101
context=intern
nat=1
; Ryan@Office2
[1013]
type=friend
username=1013
secret=1013
host=dynamic
mailbox=101
context=intern
nat=1
; Eric@Office
[1021]
type=friend
username=1021
secret=1021
host=dynamic
mailbox=102
context=intern
nat=1
; Linda@Office
[1031]
type=friend
username=1031
secret=1031
host=dynamic
mailbox=103
context=intern
nat=1
; extensions.conf ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;
[intern]
exten => _.,1,NoOp
;exten => _.,1,Macro(record-on,${EXTEN},${CALLERIDNUM})
exten => _.,2,Goto(intern-post,${EXTEN},1)
[intern-post]
;exten => h,1,Macro(record-cleanup)
exten => h,1,Hangup
include => outbound-operator
include => outbound-e911
include => outbound-information
include => outbound-local-sip
include => outbound-local-pstn
include => outbound-local-pstn-toll
include => outbound-local-pstn-toll-free
include => outbound-iaxtel
exten => i,1,Playback(invalid)
exten => i,2,Hangup
[outbound-local-pstn]
exten => _8NXXXXXX,1,Dial(SIP/${EXTEN:1}@gateway,20,tr)
exten => _8NXXXXXX,2,Playback(invalid)
[outbound-local-sip]
exten => 101,1,Dial(SIP/1012,30,tr)
exten => 101,2,Hangup
exten => 1011,1,Dial(SIP/1011,30,tr)
exten => 1011,2,Hangup
exten => 1012,1,Dial(SIP/1012,30,tr)
exten => 1012,2,Hangup
exten => 1013,1,Dial(SIP/1013,30,tr)
exten => 1013,2,Hangup
exten => 102,1,Dial(SIP/1021,30,tr)
exten => 102,2,Hangup
exten => 103,1,Dial(SIP/1031,30,tr)
exten => 103,2,Hangup
exten => 110,1,Dial(SIP/1101,30,tr)
exten => 110,2,Hangup
exten => 111,1,Dial(SIP/1111,30,tr)
exten => 111,2,Hangup
exten => 112,1,Dial(SIP/1121,30,tr)
exten => 112,2,Hangup
exten => 113,1,Dial(SIP/1131,30,tr)
exten => 113,2,Hangup
exten => 114,1,Dial(SIP/1141,30,tr)
exten => 114,2,Hangup
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-Ryan
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of AstGrp
Sent: Sunday, April 04, 2004 12:44 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Asterisk - Cisco 7960 - NAT
Can you post some of your sip configs and your extension configs.
Thanks,
-gcc
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ryan Parlee
Posted At: Sunday, April 04, 2004 12:10 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Asterisk - Cisco 7960 - NAT
Subject: [Asterisk-Users] Asterisk - Cisco 7960 - NAT
We have 10 Cisco 7960 phones at our office and a single static IP. Our
asterisk server sits in the colo facility at our ISP. All phones are
setup with a unique voip_control_port and they are all able to dial out.
However, my phone is the only one that can receive a call.
Every phone in the office can dial my extension and it will ring. I
can
call our main number and my phone will ring. But no other phone will
ring! I get a fastbusy signal when trying to dial someone else's
extension from my phone or from another phone.
Can someone please help!
Thanks, Ryan
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