app_dial.c: dial_exec_full: Unable to create channel of

Hi

I’ve set up a test box with Asterisk 1.6 and Asterisk GUI, and when trying to call Softphone to Softphone within our LAN, I’m getting an error message on the CLI Console…

SIP Debugging Disabled == Using SIP RTP CoS mark 5 -- Executing [1000@phones:1] Verbose("SIP/test-00000027", "1,Extension 1000") in new stack Extension 1000 -- Executing [1000@phones:2] Dial("SIP/test-00000027", "SIP/ian,30") in new stack == Using SIP RTP CoS mark 5 [Oct 14 10:05:26] WARNING[16224]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [1000@phones:3] Hangup("SIP/test-00000027", "") in new stack == Spawn extension (phones, 1000, 3) exited non-zero on 'SIP/test-00000027'

I’ve only changed sip.conf and extensions.conf, everything else is standard… I currently have no access to external sip trunks to make calls, etc and is purely softphone (sip) to softphone (sip) on our LAN, i’ve tried X-Lite, Ekiga, etc and they all produce the same results, i’ve also tried clients on XP and Linux and still no luck…

Am I missing something that’s need to create SIP channels ?

I’ve had a look through the forum and although this error is shown in other posts, they all seem to be related to different issues, and don’t have resolutions that would apply here…

Here are the contents of the two changed files - I’ve remove all the comments due to post size…

/etc/asterisk/extensions.conf

[code]
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
[globals]
CONSOLE = Console/dsp ; Console interface for demo
IAXINFO = guest ; IAXtel username/password
TRUNK = DAHDI/G2 ; Trunk interface
TRUNKMSD = 1 ; MSD digits to strip (usually 1 or 0)
FEATURES =
DIALOPTIONS =
RINGTIME = 20
FOLLOWMEOPTIONS =
PAGING_HEADER = Intercom
CID_6000 = 08453714910
trunk_1 = SIP/trunk_1

[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)}/${FILTER(0-9,${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

[outbound-freenum]
exten => _XX!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XX
X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXX!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXX
X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXXX!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXXX
X!,1,Goto(outbound-freenum2,${EXTEN},1)

[outbound-freenum2]
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ; make sure the suffix is all digits as well
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != “${SUFFIX}”]?fn-CONGESTION,1)
same => n,Set(TIMEOUT(absolute)=10800)
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,1,freenum.org)}) ; perform our lookup with freenum.org
same => n,GotoIf($["${isnresult}" != “”]?from)
same => n,Set(DIALSTATUS=CONGESTION)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = “”]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
same => n,Set(SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we’ll use it for our outbound dialing domain
same => n(dial),Dial(SIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)

exten => fn-BUSY,1,Busy()

exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()

[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

[stdexten]
exten => _X.,50000(stdexten),NoOp(Start stdexten)
exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
exten => _X.,n,Set(LOCAL(dev)=${ARG1})
exten => _X.,n,Set(LOCAL(cntx)=${ARG2})

exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? “@${cntx}” :: “”])
exten => _X.,n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,NoOp(Finish stdexten NOANSWER)
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start

exten => stdexten-BUSY,1,Voicemail(${mbx},b)
exten => stdexten-BUSY,n,NoOp(Finish stdexten BUSY)
exten => stdexten-BUSY,n,Return() ; If they press #, return to start

exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return()

[stdPrivacyexten]
exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
exten => _X.,n,Set(LOCAL(ext)=${ARG1})
exten => _X.,n,Set(LOCAL(dev)=${ARG2})
exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
exten => _X.,n,Set(LOCAL(cntx)=${ARG5})

exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? “@${cntx}” :: “”])
exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start

exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
exten => stdexten-BUSY,n,Return() ; If they press #, return to start

exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1) ; Callee chose to send this call to a polite “Don’t call again” script.

exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1) ; Callee chose to send this call to a telemarketer torture script.

exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return

[macro-page];
exten => s,1,ChanIsAvail(${ARG1},s) ; s is for ANY call
exten => s,n,GoToIf($[${AVAILORIGCHAN} = “”]?fail:autoanswer)
exten => s,n(autoanswer),Set(_ALERT_INFO=“RA”) ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!
exten => s,n,Dial(${ARG1})
exten => s,n(fail),Hangup

[demo]
include => stdexten
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…"
exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
exten => 1234,n,Goto(default,s,1) ; exited Voicemail

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@pbx.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)
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)

exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})

[default]
exten => s,1,Verbose(1,Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[time]
exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
exten => _X.,n,Wait(0.25)
exten => _X.,n,Answer()
exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
exten => _X.,n,SayPhonetic(z)
exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
exten => _X.,n,Playback(spy-local)
exten => _X.,n,WaitUntil(${FUTURETIME})
exten => _X.,n,Playback(beep)
exten => _X.,n,Return()

[ani]
exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
exten => _X.,n,Wait(0.25)
exten => _X.,n,Answer()
exten => _X.,n,Playback(vm-from)
exten => _X.,n,SayDigits(${CALLERID(ani)})
exten => _X.,n,Wait(1.25)
exten => _X.,n,SayDigits(${CALLERID(ani)}) ; playback again in case of missed digit
exten => _X.,n,Return()
[macro-stdexten]
exten = s,1,Set(_DYNAMIC_FEATURES=${FEATURES})
exten = s,2,Set(ORIG_ARG1=${ARG1})
exten = s,3,GotoIf($["${FOLLOWME
${ARG1}}" = “1”]?6:4)
exten = s,4,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,5,Goto(s-${DIALSTATUS},1)
exten = s,6,Macro(stdexten-followme,${ARG1},${ARG2})
exten = s-NOANSWER,1,Voicemail(${ORIG_ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ORIG_ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ORIG_ARG1})
[macro-stdexten-followme]
exten = s,1,Answer
exten = s,2,Set(ORIG_ARG1=${ARG1})
exten = s,3,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,4,Set(_FMCIDNUM=${CALLERID(num)})
exten = s,5,Set(FMCIDNAME=${CALLERID(name)})
exten = s,6,Followme(${ORIG_ARG1},${FOLLOWMEOPTIONS})
exten = s,7,Voicemail(${ORIG_ARG1},u)
exten = s-NOANSWER,1,Voicemail(${ORIG_ARG1},u)
exten = s-BUSY,1,Voicemail(${ORIG_ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ORIG_ARG1})
[macro-pagingintercom]
exten = s,1,SIPAddHeader(Alert-Info: ${PAGING_HEADER})
exten = s,2,Page(${ARG1},${ARG2})
exten = s,3,Hangup
[conferences]
[ringgroups]
[queues]
[voicemenus]
[voicemailgroups]
[directory]
[page_an_extension]
[pagegroups]
[asterisk_guitools]
exten = executecommand,1,System(${command})
exten = executecommand,n,Hangup()
exten = record_vmenu,1,Answer
exten = record_vmenu,n,Playback(vm-intro)
exten = record_vmenu,n,Record(${var1},0,500,k)
exten = record_vmenu,n,Playback(vm-saved)
exten = record_vmenu,n,Playback(vm-goodbye)
exten = record_vmenu,n,Hangup
exten = play_file,1,Answer
exten = play_file,n,Playback(${var1})
exten = play_file,n,Hangup
[macro-trunkdial-failover-0.3]
exten = s,1,GotoIf($[${LEN(${FMCIDNUM})} > 6]?1-fmsetcid,1)
exten = s,2,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} > 1]?1-setgbobname,1)
exten = s,3,Set(CALLERID(num)=${IF($[${LEN(${CID
${CALLERID(num)}})} > 2]?${CID
${CALLERID(num)}}:)})
exten = s,n,GotoIf($[${LEN(${CALLERID(num)})} > 6]?1-dial,1)
exten = s,n,Set(CALLERID(all)=${IF($[${LEN(${CID
${ARG3}})} > 6]?${CID
${ARG3}}:${GLOBAL_OUTBOUNDCID})})
exten = s,n,Goto(1-dial,1)
exten = 1-setgbobname,1,Set(CALLERID(name)=${GLOBAL_OUTBOUNDCIDNAME})
exten = 1-setgbobname,n,Goto(s,3)
exten = 1-fmsetcid,1,Set(CALLERID(num)=${FMCIDNUM})
exten = 1-fmsetcid,n,Set(CALLERID(name)=${FMCIDNAME})
exten = 1-fmsetcid,n,Goto(1-dial,1)
exten = 1-dial,1,Dial(${ARG1})
exten = 1-dial,n,Gotoif(${LEN(${ARG2})} > 0 ?1-${DIALSTATUS},1:1-out,1)
exten = 1-CHANUNAVAIL,1,Dial(${ARG2})
exten = 1-CHANUNAVAIL,n,Hangup()
exten = 1-CONGESTION,1,Dial(${ARG2})
exten = 1-CONGESTION,n,Hangup()
exten = 1-out,1,Hangup()
[queue-member-manager]
exten = handle_member,1,Verbose(2, Looping through queues to log in or out queue members)
exten = handle_member,n,Set(thisActiveMember=${CHANNEL(channeltype)}/${CHANNEL(peername)})
exten = handle_member,n,Set(queue_field=1)
exten = handle_member,n,Set(thisQueueXtn=${CUT(QUEUES,${queue_field})})
exten = handle_member,n,While($[${EXISTS(${thisQueueXtn})}])
exten = handle_member,n,Macro(member-loginlogout)
exten = handle_member,n,Set(queue_field=$[${queue_field} + 1])
exten = handle_member,n,Set(thisQueueXtn=${CUT(QUEUES,${queue_field})})
exten = handle_member,n,EndWhile()
[macro-member-loginlogout]
exten = s,1,Verbose(2, Logging queue member in or out of the request queue)
exten = s,n,Set(thisQueue=${thisQueueXtn})
exten = s,n,Set(queueMembers=${QUEUE_MEMBER_LIST(${thisQueue})})
exten = s,n,MacroIf("${queueMembers}" = “”]?q_login)
exten = s,n,Set(field=1)
exten = s,n,Set(logged_in=0)
exten = s,n,Set(thisQueueMember=${CUT(queueMembers,${field})})
exten = s,n,While($[${EXISTS(${thisQueueMember})}])
exten = s,n,GotoIf($["${thisQueueMember}" != “${thisActiveMember}”]?check_next)
exten = s,n,Set(logged_in=1)
exten = s,n,ExitWhile()
exten = s,n(check_next),Set(field=$[${field} + 1])
exten = s,n,Set(thisQueueMember=${CUT(queueMembers,${field})})
exten = s,n,EndWhile()
exten = s,n,MacroIf($[${logged_in} = 0]?q_login:q_logout)
[macro-q_login]
exten = s,1,Verbose(2, Logging ${thisActiveMember} into the ${thisQueue} queue)
exten = s,n,AddQueueMember(${thisQueue},${thisActiveMember})
exten = s,n,Playback(silence/1)
exten = s,n,ExecIf($["${AQMSTATUS}" = “ADDED”]?Playback(agent-loginok):Playback(an-error-has-occurred))
[macro-q_logout]
exten = s,1,Verbose(2, Logged ${thisActiveMember} out of ${thisQueue} queue)
exten = s,n,RemoveQueueMember(${thisQueue},${thisActiveMember})
exten = s,n,Playback(silence/1)
exten = s,n,ExecIf($["${AQMSTATUS}" = “REMOVED”]?Playback(agent-loggedoff):Playback(an-error-has-occurred))
[DLPN_DialPlan1]
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension

[incoming_calls]

[internal]
exten => s,1,Verbose(1,Echo test application)
exten => s,n,Echo()
exten => s,n,Hangup()

exten => 1000,1,Verbose(1,Extension 1000)
exten => 1000,n,Dial(SIP/ian,30)
exten => 1000,n,Hangup()

exten => 1002,1,Verbose(1,Extension 1002)
exten => 1002,n,Dial(SIP/test,30)
exten => 1002,n,Hangup()

[phones]
include => internal
include => default
[DID_trunk_1]
include = DID_trunk_1_default
[DID_trunk_1_default][/code]

/etc/asterisk/sip.conf

[code]
[general]
context = default ; Default context for incoming calls
allowoverlap = no ; Disable overlap dialing support. (Default is yes)
udpbindaddr = 0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
bindport = 5060
bindaddr = 0.0.0.0
tcpenable = no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr = 0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)

srvlookup = yes ; Enable DNS SRV lookups on outbound calls
subscribecontext = device-hints
subscribecontext = device-hints
subscribecontext = device-hints
subscribecontext = device-hints

[authentication]
basic-options; a template
dtmfmode = rfc2833
context = from-office
type = friend

natted-phone; another template inheriting basic-options
nat = yes
directmedia = no
host = dynamic

public-phone; another template inheriting basic-options
nat = no
directmedia = yes

my-codecs; a template for my preferred codecs
disallow = all
allow = ilbc
allow = g729
allow = gsm
allow = g723
allow = ulaw

ulaw-phone; and another one for ulaw-only
disallow = all
allow = ulaw

[ian]
type = friend
context = phones
host = dynamic

[test]
type = friend
context = phones
host = dynamic[/code]

Please provide sip show peers and sip show users output.

ian has probably never registered.

localhost*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 6000 password123 DLPN_DialPlan1 No Always ian phones No RFC3581 test phones No RFC3581

localhost*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 6000/6000 (Unspecified) D N 5060 Unmonitored ian/ian 192.168.7.25 D 5060 Unmonitored test/test 192.168.7.63 D 5060 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]