Hello…
Here is the CLI from asterisk whenever I try to make an inbound call:
................ ]
Asterisk Ready.
*CLI> [Oct 20 11:06:08] NOTICE[28467]: chan_sip.c:12336 handle_response_peerpoke: Peer 'varphonex' is now Reachable. (73ms / 2000ms)
[Oct 20 11:06:08] NOTICE[28467]: chan_sip.c:12336 handle_response_peerpoke: Peer 'luke' is now Reachable. (221ms / 2000ms)
[Oct 20 11:06:08] NOTICE[28467]: chan_sip.c:12336 handle_response_peerpoke: Peer 'sipura1' is now Reachable. (341ms / 2000ms)
[Oct 20 11:06:20] NOTICE[28525]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/varphonex-081e1df0' not posted
All I end up getting is a “fast busy”
Here is my sip.conf
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will listen on
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
canreinvite=no
[sipura1]
type=friend
secret=******
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=phone1
[luke]
type=friend
secret=******
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=phone1
trustrpid = yes
sendrpid = yes
register => 4213002:dci123@sip.varphonex.com:5060
[varphonex]
type=friend
username:1234567
realm=sip.varphonex.com
canreinvite=yes
secret=*****
insecure=very
host=sip.varphonex.com
disallow=all
allow=ilbc
allow=ulaw
allow=alaw
qualify=yes
nat=no
outboundproxy=sip.varphonex.com
#regexten=yes
auth=1234567:******@sip.varphonex.com:5060[/code]
and here is my extensions.conf:
[code][general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[dundi-e164-canonical]
[dundi-e164-customers]
[dundi-e164-via-pstn]
[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
switch => DUNDi/e164
[dundi-e164-lookup]
include => dundi-e164-local
include => dundi-e164-switch
[macro-dundi-e164]
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext
[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunkld]
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunklocal]
exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunktollfree]
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[international]
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
ignorepat => 9
include => local
include => trunkld
[local]
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
include => parkedcalls
[macro-trunkdial]
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp
[macro-stdexten];
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain
[macro-stdPrivacyexten];
exten => s,1,Dial(${ARG2},20|p) ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call screening)
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten => s-DONTCALL,1,Goto(${ARG3},s,1) ; Callee chose to send this call to a polite "Don't call again" script.
exten => s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to send this call to a telemarketer torture script.
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain
[macro-page];
exten => s,1,ChanIsAvail(${ARG1}|js) ; j is for Jump and s is for ANY call
exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1}||)
exten => s,n(fail),Hangup
[demo]
exten => s,1,Wait(1) ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
exten => 1235,1,Voicemail(1234,u) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(1234,b) ; Unless busy
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})
[default]
include => demo
[phone1]
exten => 100,1,Dial(SIP/luke)
exten => 101,1,Dial(SIP/sipura1)
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
exten => _X.,1,Answer
exten => _X.,2,Wait(2)
exten => _X.,3,Dial(SIP/varphonex)
#exten => _X.,4,Hangup
#exten => _9.,1,Dial(SIP/${EXTEN:1}@my_provider,30,r)[/code]
And here is the Debug from my outbound sip channel:
[code]SIP Debugging Enabled for IP: 208.239.76.163:5060
*CLI> Audio is at 72.167.14.79 port 15110
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 208.239.76.163:5060:
INVITE sip:sip.varphonex.com SIP/2.0
Via: SIP/2.0/UDP 72.167.14.79:5060;branch=z9hG4bK7acfc37c;rport
From: "luke" <sip:luke@72.167.14.79>;tag=as47f1eb1f
To: <sip:sip.varphonex.com>
Contact: <sip:luke@72.167.14.79>
Call-ID: 5a5a31493f49c69a08d898b41dcbde4d@72.167.14.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 20 Oct 2007 18:30:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 309
v=0
o=root 28432 28432 IN IP4 72.167.14.79
s=session
c=IN IP4 72.167.14.79
t=0 0
m=audio 15110 RTP/AVP 0 97 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from 208.239.76.163:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 72.167.14.79:5060;received=72.167.14.79;branch=z9hG4bK7acfc37c; rport=5060
From: "luke" <sip:luke@72.167.14.79>;tag=as47f1eb1f
To: <sip:sip.varphonex.com>
Call-ID: 5a5a31493f49c69a08d898b41dcbde4d@72.167.14.79
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from 208.239.76.163:5060 --->
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 72.167.14.79:5060;received=72.167.14.79;branch=z9hG4bK7acfc37c; rport=5060
From: "luke" <sip:luke@72.167.14.79>;tag=as47f1eb1f
To: <sip:sip.varphonex.com>;tag=SDec0m199-14ec6e58
Call-ID: 5a5a31493f49c69a08d898b41dcbde4d@72.167.14.79
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Transmitting (no NAT) to 208.239.76.163:5060:
ACK sip:sip.varphonex.com SIP/2.0
Via: SIP/2.0/UDP 72.167.14.79:5060;branch=z9hG4bK7acfc37c;rport
From: "luke" <sip:luke@72.167.14.79>;tag=as47f1eb1f
To: <sip:sip.varphonex.com>;tag=SDec0m199-14ec6e58
Contact: <sip:luke@72.167.14.79>
Call-ID: 5a5a31493f49c69a08d898b41dcbde4d@72.167.14.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
[Oct 20 11:30:27] NOTICE[16335]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/var phonex-081e1df0' not posted
Really destroying SIP dialog '5a5a31493f49c69a08d898b41dcbde4d@72.167.14.79' Met hod: INVITE
Reliably Transmitting (no NAT) to 208.239.76.163:5060:
OPTIONS sip:sip.varphonex.com SIP/2.0
Via: SIP/2.0/UDP 72.167.14.79:5060;branch=z9hG4bK24cbba20;rport
From: "asterisk" <sip:asterisk@72.167.14.79>;tag=as103d6411
To: <sip:sip.varphonex.com>
Contact: <sip:asterisk@72.167.14.79>
Call-ID: 72a4e86f0a449d04317d54963d63739f@72.167.14.79
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 20 Oct 2007 18:31:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 208.239.76.163:5060 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 72.167.14.79:5060;received=72.167.14.79;branch=z9hG4bK24cbba20;rport=5060
From: "asterisk" <sip:asterisk@72.167.14.79>;tag=as103d6411
To: <sip:sip.varphonex.com>;tag=aprqngfrt-fsuhvb20000c6
Call-ID: 72a4e86f0a449d04317d54963d63739f@72.167.14.79
CSeq: 102 OPTIONS
Allow: INVITE,ACK,BYE,REGISTER,CANCEL,PRACK,INFO,NOTIFY,REFER
I know its alot to read through. If anyone can see what I am doing, let me wrong… let me know. thanks.