file: sip.conf
----------------------------------------------------------------------------------
[general]
static=yes
writeprotect=no
DYNAMIC_FEATURES=>automon
[default]
[sip_local]
include => parkedcalls
; incoming
exten => 989,1,Dial(SIP/1&SIP/2,60,tw)
exten => 989,2,Hangup
;local
exten => 1,1,Dial(SIP/1,60,tTrwg)
exten => 1,2,Goto(988,2)
exten => 2,1,Dial(SIP/2,60,tTrwg)
exten => 2,2,Goto(988,2)
;external
exten => _0.,1,Dial(SIP/999,60,tTrwgD(${LDPrefix}${EXTEN:1}))
exten => _0.,2,Goto(988,2)
file: sip.conf
----------------------------------------------------------------------------------
port=5060
bindaddr=0.0.0.0
context=default
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
allow=g723.1
musiconhold=default
musicclass=default
[1]
type=friend
username=xxx1
dtmfmode=inband
context=sip_local
host=dynamic
callgroup=1
pickupgroup=1
[2]
type=friend
username=xxx2
dtmfmode=inband
context=sip_local
host=dynamic
callgroup=1
pickupgroup=1
-----------------------------------------------------------------------------------------
file : features.conf
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
parkingtime => 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call
courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
xfersound = beep ; to indicate an attended transfer is complete
xferfailsound = beeperr ; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
findslot => next ; Continue to the 'next' parking space. Defaults to 'first' available
pickupexten = *8 ; Configure the pickup extension. Default is *8
featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation. Default is 500
[featuremap]
blindxfer => ## ; Blind transfer
disconnect => *0 ; Disconnect
automon => *1 ; One Touch Record
atxfer => *2 ; Attended transfer
----------------------------------------------------------------------------------
DEBUG and VERBOSE
*CLI>
*CLI>
*CLI> Nov 20 00:28:36 DEBUG[8088]: chan_sip.c:3101 sip_alloc: Allocating new SIP dialog for 787858-5573737d-13c4-3a50c1d4-2cbb5e-221f@85.115.115.120 - INVITE (With RTP)
Nov 20 00:28:36 DEBUG[8088]: chan_sip.c:7096 check_user_full: Setting NAT on RTP to 0
Nov 20 00:28:36 DEBUG[8088]: chan_sip.c:10320 handle_request_invite: Checking SIP call limits for device 2
Nov 20 00:28:36 DEBUG[8088]: chan_sip.c:6025 build_route: build_route: Contact hop: <sip:FXS3@85.115.115.125:5060>
Nov 20 00:28:36 DEBUG[8082]: channel.c:774 channel_find_locked: Avoiding initial deadlock for 'SIP/2-635c'
Nov 20 00:28:36 DEBUG[8090]: pbx.c:1667 pbx_extension_helper: Launching 'Answer'
Nov 20 00:28:36 DEBUG[8082]: channel.c:774 channel_find_locked: Avoiding initial deadlock for 'SIP/2-635c'
Nov 20 00:28:36 DEBUG[8090]: chan_sip.c:2505 sip_answer: sip_answer(SIP/2-635c)
Nov 20 00:28:36 DEBUG[8090]: pbx.c:1667 pbx_extension_helper: Launching 'BackGround'
Nov 20 00:28:36 DEBUG[8090]: channel.c:2335 set_format: Set channel SIP/2-635c to write format gsm
Nov 20 00:28:36 DEBUG[8090]: rtp.c:1341 ast_rtp_write: Ooh, format changed from unknown to ulaw
Nov 20 00:28:36 DEBUG[8090]: channel.c:1700 ast_settimeout: Scheduling timer at 160 sample intervals
Nov 20 00:28:36 DEBUG[8091]: app_queue.c:471 changethread: Device 'SIP/2' changed to state '2' (In use)
Nov 20 00:28:36 DEBUG[8092]: app_queue.c:471 changethread: Device 'SIP/2' changed to state '2' (In use)
Nov 20 00:28:36 DEBUG[8088]: chan_sip.c:1389 __sip_ack: Stopping retransmission on '787858-5573737d-13c4-3a50c1d4-2cbb5e-221f@85.115.115.120' of Response 1: Match Found
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI>
*CLI> Nov 20 00:28:39 DEBUG[8090]: channel.c:1700 ast_settimeout: Scheduling timer at 0 sample intervals
Nov 20 00:28:39 DEBUG[8090]: channel.c:1700 ast_settimeout: Scheduling timer at 0 sample intervals
Nov 20 00:28:39 DEBUG[8090]: channel.c:2335 set_format: Set channel SIP/2-635c to write format ulaw
Nov 20 00:28:39 DEBUG[8090]: chan_sip.c:2969 sip_rtp_read: * Detected inband DTMF '*'
Nov 20 00:28:39 DEBUG[8090]: chan_sip.c:2969 sip_rtp_read: * Detected inband DTMF '8'
Nov 20 00:28:39 WARNING[8090]: pbx.c:2394 __ast_pbx_run: Invalid extension '*8', but no rule 'i' in context 'sip_local'
Nov 20 00:28:39 DEBUG[8090]: channel.c:1312 ast_hangup: Hanging up channel 'SIP/2-635c'
Nov 20 00:28:39 DEBUG[8090]: chan_sip.c:2401 sip_hangup: Hangup call SIP/2-635c, SIP callid 787858-5573737d-13c4-3a50c1d4-2cbb5e-221f@85.115.115.120)
Nov 20 00:28:39 DEBUG[8090]: chan_sip.c:2409 sip_hangup: update_call_counter(2) - decrement call limit counter
Nov 20 00:28:39 DEBUG[8093]: app_queue.c:471 changethread: Device 'SIP/2' changed to state '1' (Not in use)
Nov 20 00:28:39 DEBUG[8088]: chan_sip.c:1389 __sip_ack: Stopping retransmission on '787858-5573737d-13c4-3a50c1d4-2cbb5e-221f@85.115.115.120' of Request 102: Match Found
*CLI>
*CLI>
*CLI>
*CLI> Nov 20 00:28:43 DEBUG[8088]: chan_sip.c:3101 sip_alloc: Allocating new SIP dialog for 0473c4aa78bfedde5cfde8415fedd2ee@217.199.98.33 - REGISTER (No RTP)
Nov 20 00:28:43 DEBUG[8094]: app_queue.c:471 changethread: Device 'SIP/xoffice' changed to state '1' (Not in use)