[asterisk-users] Re: Polycom 601 loop

[dave cantera]

nathan,
try dial() directly to the extension

[to-sip]
exten => _X.,1,Dial(SIP/${EXTEN}@192.168.2.13,120)

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               ; Context for incoming calls from this user


in extensions.conf
[from-sip]
exten => 201,1,Wait(1)
exten => 201,n,Answer()
exten => 201,n,Dial(SIP/201,15)
exten => 201,n,VoiceMailMain
exten => 201,n,Hangup()


Nathan Bell wrote:
> 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 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>
>
>
>   
> ------------------------------------------------------------------------
>
> ; 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()
>
>   
> ------------------------------------------------------------------------
>
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