search for: __sip_uri_options

Displaying 4 results from an estimated 4 matches for "__sip_uri_options".

2007 Jan 07
1
snom 360 auto answer
...======= ;exten => _99XXXX,1,SIPAddHeader(Call-Info: Answer-After=0) ;exten => _99XXXX,n,SIPAddHeader(Call-Info: <sip:192.168.1.113>\;answer-after=0) ;exten => _99XXXX,n,Dial(SIP/${EXTEN:2}) exten => _99XXXX,1,Set(__SIPADDHEADER=Call-Info: answer-after=0) exten => _99XXXX,n,Set(__SIP_URI_OPTIONS=intercom=true) exten => _99XXXX,n,Set(__ALERT_INFO=Ring Answer) exten => _99XXXX,n,Dial(SIP/${EXTEN:2}) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
2008 Apr 14
2
polycom auto answer
...rang it did not auto answer. Did I miss something? exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten => 22,n,SipAddHeader(Alert-Info: Ring Answer) exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten => 22,n,Set(__ALERT_INFO=Ring Answer) exten => 22,n,Set(__SIP_URI_OPTIONS=intercom=true) exten => 22,n,Dial(SIP/404) Jerry
2007 Feb 09
0
Conference & Page question
...gt; PAGE4441,n(AVAIL),ChanIsAvail(${DB(DEVICE/4441/dial)}|js) exten => PAGE4441,n(SKIPCHECK),Noop(Seems to be available (state = ${AVAILSTATUS}) exten => PAGE4441,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=6) exten => PAGE4441,n,Set(__ALERT_INFO=Ring Answer) exten => PAGE4441,n,Set(__SIP_URI_OPTIONS=intercom=true) exten => PAGE4441,n,Set(TIMEOUT(absolute)=60) exten => PAGE4441,n,Dial(${DB(DEVICE/4441/dial)},5, A(beep)) exten => PAGE4441,n(skipself),Noop(Not paging originator) exten => PAGE4441,n,Hangup exten => PAGE4441,AVAIL+101,Noop(Channel ${AVAILCHAN} is not available (stat...
2006 May 11
2
Paging and Auto Answer on Grandstream GXP2000
I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems