- hasexten=yes|no
If the context for a peer sets hasexten=yes, Asterisk creates a hint for the user in the default context as shown below for a SIP peer 6000.
CLI> dialplan show default
[ Context 'default' created by 'pbx_config' ]
'6000' => hint: SIP/6000 [pbx_config]
1. Dial(${HINT}) [pbx_config]
Therefore I can use Goto(default,6000,1) to ring it - [general] user in users.conf
It's set the default contexts for all other users. They can be overridden though. The following are in my [general] user (all my users are using SIP):
fullname = My Name
;
; Starting point of allocation of extensions
;
userbase = 6000
;
; Create SIP Peer
;
hassip = yes
;
; Create IAX friend
;
hasiax = no
registeriax = no
;
; Create manager entry
;
hasmanager = no
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
call-limit = 100
qualify = yes
disallow = all
allow = ulaw,alaw
type = friend
- Asterisk directed call pickup
I have two extensions: 6000 and 8888. Typically when there's a incoming call, only extension 6000 rings. I can pick up the call from the other extension though by pressing the # key from extension 8888. Therefore I have the following in features.conf
[general]
pickupexten = #
and the following in extensions.conf:
[globals]
voipbuster = SIP/voipbuster
[CallingRule_pickup]
exten = _#,1,Pickup(6000@default)
exten = _#,n,Hangup()
[DLPN_8888]
include = CallingRule_pickup
include = CallingRule_VBOut
include = default
[CallingRule_VBOut]
exten => _001.,1,Dial(Local/${EXTEN:2}@gv-outbound/n)
exten => _00[2-9]X.,1,Macro(trunkdial-failover-0.3,${voipbuster}/${EXTEN:0},,voipbuster,)
I had to add the following to the Dial Plan of the line that will pickup the call in my PAP2T to pass # key directly to Asterisk: #S0
- Blind transfer
I use the * key for Blind transfer. Therefore I have the following in features.conf
[featuremap]
blindxfer = *
and the following in extensions.conf:
[globals]
DIALOPTIONS = tT
[DLPN_6000]
include = CallingRule_VBOut
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension
exten = _*,1,Transfer(8888)
I had to add the following to the Dial Plan of the line that will initiate the transfer in my PAP2T to pass * key directly to Asterisk: *S0 - Connecting PAP2T to the telephone lines 1&2 in my house (T568A type socket): I cut one standard 2-wire RJ11 telephone cable assembly in half and connected them to Blue and Orange lines of the T568A. That will enable me to connect a phone onto the wall outlet at any room to my PAP2T.
Sunday, November 8, 2009
Asterisk (PBX) and PAP2T
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