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<pre>
Hi
I am using asterisk 0.7.1
I am testing with 3 SIP phone.
Phone A call to Phon B , and Phone B is Ringing.
I want to pickup that call , So I press '*8' for pickup the call on Phone
C.
But I can not pickup the call.
I can see "NOTICE[6151]:chan_sip.c:5198 handle_requst: Nothing to pick up"
in console.
;============== sip.conf ====================
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
;bindaddr = 61.36.179.152 ; Address to bind to
bindaddr = 0.0.0.0
;externip = 200.201.202.203 ; Address that we're going to put in SIP
messages if we're behind a NAT
;localnet = 61.36.179.0 ; Internal NETWORK address
;localmask = 255.255.255.128 ; Internal netmask
;context = default ; Default for incoming calls
context = from-sip
;srvlookup = yes ; Enable SRV lookups on outbound calls
;pedantic = yes ; Enable slow, pedantic checking for Pingtel
;tos=lowdelay
;tos=184
maxexpirey=3600 ; Max length of incoming registration we allow
defaultexpirey=160 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
;disallow=all ; Disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc
;
register => 9002000@61.36.179.167 ; Register with a SIP provider
;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as
1234 here.
;
callgroup=1
pickupgroup=1
[hst239]
type=friend
secret=young
dtmfmode=inband
host=61.36.179.239
threewaycall = yes
callgroup=1
pickupgroup=1
context=from-sip
[hst220]
type=friend
host=61.36.179.220
callgroup=1
pickupgroup=1
threewaycall = yes
context=from-sip
[hst238]
type=friend
host=61.36.179.238
dtmfmode=inband
callgroup=1
pickupgroup=1
threewaycall = yes
context=from-sip
[hst155]
type=friend
host=210.98.251.155
callgroup=1
pickupgroup=1
threewaycall = yes
context=sip-from
[61.36.179.167]
type=friend
username=9002000
host=61.36.179.167
callgroup=1
pickupgroup=1
context=from-sip
================= extensions.conf ===========================
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without the
';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk configuration
files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[iaxtel700]
exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
;
; International long distance through trunk
;
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXXXXXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
exten => 6601,1,WaitMusicOnHold(30) ; hur
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
;
; switch => IAX2/user:password@bigserver/local
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten => s,2,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/
unavail announce
exten => s,3,Goto(default,s,1) ; If they press #, return to start
exten => s,102,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy
announce
exten => s,103,Goto(default,s,1) ; If they press #, return to start
[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct) ; Play some instructions
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,2,Goto(s,6)
exten => 3,1,SetLanguage(fr) ; Set language to french
exten => 3,2,Goto(s,5) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,2,Voicemail(u1234) ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,2,Hangup ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,2,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the
Asterisk demo
exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,4,Goto(s,6) ; Return to the start over message.
;
; Create an extension, 600, for evaulating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,2,Echo ; Do the echo test
exten => 600,3,Playback(demo-echodone) ; Let them know it's over
exten => 600,4,Goto(s,6) ; Start over
;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,2,Goto(s,5)
;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,2,Background(thanks) ; "Thanks for calling press 1 for sales, 2
for support, ..."
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
;exten => s,2,Wait,2
;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales
department. Press 1 for steve, 2 for..."
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
; Real extensions would go here. Generally you want real extensions to be 4
or 5
; digits long (although there is no such requirement) and start with a
single
; digit that is fairly large (like 6 or 7) so that you have plenty of room
to
; overlap extensions and menu options without conflict. You can alias them
with
; names, too and use global variables
;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is
something like Zap/2
;exten => mark,1,Goto(6275|1) ; alias mark to 6275
;exten => 6236,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
;exten => wil,1,Goto(6236|1)
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,2,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme,1234
;
; Or playing an announce to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type "show applications" at
your
; friendly Asterisk CLI prompt.
;
[from-sip]
include => demo
include => parkedcalls
include => foo1
;include => foo2
;include => foo3
exten => h,1,Hangup
[foo1]
exten => 9002000,1,Ringing
exten => 9002000,2,Wait,2
exten => 9002000,3,Goto(1002|1)
exten =>
9002000,4,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/${EXTEN:1}|default,${
EXTEN:1},1)
exten => 9002000,5,BackGround(demo-congrats)
;exten => _9XXXXXX,1,Dial(SIP/9001000@61.36.179.167)
exten => **,1,ParkedCall(701)
;exten => 1002,1,Ringing
;exten => 1002,2,Wait,2
exten => 1002,1,Dial(SIP/hst239|30|Tt)
;exten =>
1002,3,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/9005000@61.36.179.239|de
fault,${EXTEN:1},1)
exten => 1003,1,Dial(SIP/hst155|30|Tt)
exten => 1004,1,Dial(SIP/hst220|30|Tt)
exten => 1005,1,Dial(SIP/hst238|30|Tt)
exten => 1006,1,Dial(SIP/1006)
exten => 5555,1,Dial(SIP/5555@210.98.251.195)
exten => 2000,1,MusicOnHold(default)
</pre>
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