[asterisk-users] Re: Polycom 601 loop

[Nathan Bell]

Sorry, forgot to attach the sip.conf and extensions.conf files. Attached 
now.
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[general]
context=from-sip		; Default context for incoming calls
				; if asterisk was compiled with OSP support.
realm=actarg.com		; Realm for digest authentication
				; defaults to "asterisk"
				; Realms MUST be globally unique according to RFC 3261
				; Set this to your host name or domain name
bindport=5060			; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes			; Enable DNS SRV lookups on outbound calls
				; Note: Asterisk only uses the first host 
				; in SRV records
				; Disabling DNS SRV lookups disables the 
				; ability to place SIP calls based on domain 
				; names to some other SIP users on the Internet
autodomain=yes			; Turn this on to have Asterisk add local host
				; name and local IP to domain list.
				; and multiline formatted headers for strict
qualify=yes
disallow=all
allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
progressinband=no              ; Polycom phones don't work properly with "never"
dtmfmode=rfc2833               ; Choices are inband, rfc2833, or info
nat=no                         ; there is not NAT between phone and Asterisk
canreinvite=no                 ; disallow RTP voice traffic to bypass Asterisk

[201]
type=friend                    ; Friends place calls and receive calls
context=from-sip               ; Context for incoming calls from this user
secret=asteriskpassword
host=dynamic                   ; This peer register with us
callerid=John Doe <201>

[202]
type=friend                    ; Friends place calls and receive calls
context=from-sip               ; Context for incoming calls from this user
secret=asteriskpassword
host=dynamic                   ; This peer register with us
callerid=Jane Doe <202>


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; from outside T1
[from-ptsn]
exten => s,1,Answer()
include => cac-ext
include => sip-ext
include => intertel-ext
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()

; from sip lines
[from-sip]
include => internal

; generic interal route
[internal]
exten => s,1,Answer()
include => cac-ext
include => sip-ext
include => intertel-ext
include => to-ptsn

; check if extension is to sip
[sip-ext]
exten => _20X,1,Goto(to-sip,${EXTEN},1)

; send call to sip
[to-sip]
exten => _X.,1,Dial(SIP/${EXTEN}@192.168.2.13,120)
exten => _X.,2,Playback(vm-nobodyavail)
exten => _X.,3,Hangup()
exten => _X.,102,Playback(tt-allbusy)
exten => _X.,103,Hangup()