Noobie Needing Help with Station to Station calling

system · September 29, 2010, 4:37pm

I am New to the world of Asterisk. The geek in me stays up late at night trying to sip from the fire hose of information in the forums and other documentation. I have downloaded and installed version 1.6.2.11 with the Asterisk GUI on an old server that was collecting dust. The system does not have any telephony cards in it.

I have setup 2 SIP/IAX users, enabling VM on both.
I setup User1 as extension 6000 and User2 as 6001.
I setup *98 as the VM extension.
I downloaded X-Lite4 as my softphone and configured it as User2/6001.

When I call User1 by dialing 6000, I get the 3 tone error sound. When I call *98 to try to get into the VM system, I get the 3 tone error sound. However, User1 shows missed calls on the System Status screen.

Other than the initial install, SIP configuration and User setup, have created a VOIP SIP Trunk and an “internal” dialing plan using the GUI.

What else should I be doing to make station to station calls via the SIP softphones?

david55 · September 30, 2010, 10:29am

You need to define both devices and extensions. In practice, that means you need to show us the contents of both sip.conf and extensions.conf.

system · September 30, 2010, 6:02pm

Here is my sip.conf file

[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
subscribecontext=device-hints,device-hints,device-hints
subscribecontext=device-hints
subscribecontext=device-hints
allowexternaldomains=yes
allowguest=yes
allowsubscribe=yes
allowtransfer=yes
alwaysauthreject=no
autodomain=no
bindaddr=192.168.16.49
bindport=5060
callevents=no
canreinvite=yes
checkmwi=10
compactheaders=no
defaultexpiry=120
domain=
dtmfmode=
dumphistory=no
externrefresh=10
fromdomain=
g726nonstandard=no
jbenable=no
jbforce=no
jbimpl=
jblog=no
jbmaxsize=
jbresyncthreshold=
language=
maxcallbitrate=384
maxexpiry=3600
minexpiry=60
mohinterpret=default
mohsuggest=
mwi_from=
nat=yes
notifyringing=yes
pedantic=no
progressinband=never
promiscredir=no
realm=asterisk
recordhistory=no
registerattempts=0
registertimeout=20
relaxdtmf=no
rtpholdtimeout=
rtptimeout=
sendrpid=no
sipdebug=no
t1min=100
t38pt_udptl=no
tos_audio=none
tos_sip=none
tos_video=none
trustrpid=no
useragent=Asterisk PBX
usereqphone=no
videosupport=no
localnet=192.168.16.49
disallow=all
allow=ulaw,alaw,gsm,speex
subscribecontext=device-hints
subscribecontext=device-hints
subscribecontext=device-hints
[authentication]

Here is the extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=DAHDI/G2
TRUNKMSD=1
FEATURES=
DIALOPTIONS=
RINGTIME=20
FOLLOWMEOPTIONS=
PAGING_HEADER=Intercom
CID_6000=6000
CID_6001=6001
trunk_1=SIP/trunk_1
timeinterval_all_day_every_day=,sun-sat,,*
[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]

[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
include=parkedcalls
[outbound-freenum]
exten=_XX!,1,Goto(outbound-freenum2,${EXTEN},1)
exten=_XXX!,1,Goto(outbound-freenum2,${EXTEN},1)
exten=_XXXX!,1,Goto(outbound-freenum2,${EXTEN},1)
exten=_XXXXX!,1,Goto(outbound-freenum2,${EXTEN},1)
exten=_XXXXXX!,1,Goto(outbound-freenum2,${EXTEN},1)
exten=_XXXXXXX!,1,Goto(outbound-freenum2,${EXTEN},1)
[outbound-freenum2]
exten=_X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same=n,Set(SUFFIX=${CUT(EXTEN,*,2-)})
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)})
same=n,GotoIf($["${isnresult}" != “”]?from)
same=n,Set(DIALSTATUS=CONGESTION)
same=n,Goto(fn-CONGESTION,1)
same=n(from),Set(SIPFROMUSER=${CALLERID(num)})
same=n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = “”]?dial)
same=n,Set(SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
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(${dev},20)
exten=_X.,n,Goto(stdexten-${DIALSTATUS},1)
exten=stdexten-NOANSWER,1,Voicemail(${mbx},u)
exten=stdexten-NOANSWER,n,NoOp(Finish stdexten NOANSWER)
exten=stdexten-NOANSWER,n,Return()
exten=stdexten-BUSY,1,Voicemail(${mbx},b)
exten=stdexten-BUSY,n,NoOp(Finish stdexten BUSY)
exten=stdexten-BUSY,n,Return()
exten=_stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)
exten=a,1,VoicemailMain(${mbx})
exten=a,n,Return()
[stdPrivacyexten]
exten=_X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
exten=_X.,n,Set(LOCAL(ext)=${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(${dev},20,p)
exten=_X.,n,Goto(stdexten-${DIALSTATUS},1)
exten=stdexten-NOANSWER,1,Voicemail(${mbx},u)
exten=stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
exten=stdexten-NOANSWER,n,Return()
exten=stdexten-BUSY,1,Voicemail(${mbx},b)
exten=stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
exten=stdexten-BUSY,n,Return()
exten=stdexten-DONTCALL,1,Goto(${dontcntx},s,1)
exten=stdexten-TORTURE,1,Goto(${tortcntx},s,1)
exten=_stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)
exten=a,1,VoicemailMain(${mbx})
exten=a,n,Return
[macro-page]
exten=s,1,ChanIsAvail(${ARG1},s)
exten=s,n,GoToIf($[${AVAILORIGCHAN} = “”]?fail:autoanswer)
exten=s,n(autoanswer),Set(_ALERT_INFO=“RA”)
exten=s,n,SIPAddHeader(Call-Info: Answer-After=0)
exten=s,n,NoOp()
exten=s,n,Dial(${ARG1})
exten=s,n(fail),Hangup
[demo]
include=stdexten
exten=s,1,Wait(1)
exten=s,n,Answer
exten=s,n,Set(TIMEOUT(digit)=5)
exten=s,n,Set(TIMEOUT(response)=10)
exten=s,n(restart),BackGround(demo-congrats)
exten=s,n(instruct),BackGround(demo-instruct)
exten=s,n,WaitExten
exten=2,1,BackGround(demo-moreinfo)
exten=2,n,Goto(s,instruct)
exten=3,1,Set(LANGUAGE()=fr)
exten=3,n,Goto(s,restart)
exten=1000,1,Goto(default,s,1)
exten=1234,1,Playback(transfer,skip)
exten=1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
exten=1234,n,Goto(default,s,1)
exten=1235,1,Voicemail(1234,u)
exten=1236,1,Dial(Console/dsp)
exten=1236,n,Voicemail(1234,b)
exten=#,1,Playback(demo-thanks)
exten=#,n,Hangup
exten=t,1,Goto(#,1)
exten=i,1,Playback(invalid)
exten=500,1,Playback(demo-abouttotry)
exten=500,n,Dial(IAX2/guest@pbx.digium.com/s@default)
exten=500,n,Playback(demo-nogo)
exten=500,n,Goto(s,6)
exten=600,1,Playback(demo-echotest)
exten=600,n,Echo
exten=600,n,Playback(demo-echodone)
exten=600,n,Goto(s,6)
exten=76245,1,Macro(page,SIP/Grandstream1)
exten=7XXX,1,Macro(page,SIP/${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/${EXTEN})
[default]
exten=#6XXX,1,Set(MBOX=${EXTEN:1}@default)
exten=#6XXX,n,VoiceMail(${MBOX})
exten=a,1,VoicemailMain(${MBOX})
exten=*98,1,VoiceMailMain(${CALLERID(num)}@default)
[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(${CALLERID(ani)})
exten=_X.,n,Return()
[macro-stdexten]
exten=s,1,Set(_DYNAMIC_FEATURES=${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]
exten=*97,1,Directory(default,default,)
[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))
[DID_trunk_1]
include=DID_trunk_1_default
[DID_trunk_1_default]

[DLPN_internal]
include=default
include=parkedcalls
include=conferences
include=ringgroups
include=voicemenus
include=queues
include=voicemailgroups
include=directory
include=pagegroups
include=page_an_extension

david55 · September 30, 2010, 6:25pm

GUIs tend to put sip.conf information in users.conf. Basically, I’m looking to make sure that you have sip user information that matches the phone, which specifies a context that includes extension 6000.

The console log may also help, as it will probably say which context it was trying when it failed to find the extension

system · September 30, 2010, 7:34pm

I’m not sure how to go about getting a copy of the console log.

Here is the users.conf

[general]
fullname=New User
userbase=6000
hasvoicemail=yes
vmsecret=1234
hassip=yes
hasiax=yes
hasmanager=no
callwaiting=yes
threewaycalling=yes
callwaitingcallerid=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callgroup=1
pickupgroup=1
vmexten=*98

[6000]
username=6000
transfer=yes
mailbox=6000
call-limit=100
type=peer
fullname=User 1
registersip=no
host=dynamic
callgroup=1
type=peer
context=DLPN_internal
cid_number=6000
hasvoicemail=yes
vmsecret=5544
email=Bilbo@baggins.com
threewaycalling=no
hasdirectory=no
callwaiting=no
hasmanager=no
hasagent=no
hassip=yes
hasiax=yes
secret=
nat=yes
canreinvite=yes
dtmfmode=rfc2833
insecure=no
pickupgroup=1
disallow=all
allow=ulaw,alaw,gsm,speex
macaddress=00270E0B5B3A
autoprov=yes
label=6000
linenumber=1
LINEKEYS=1
callcounter=yes
[6001]
username=6001
transfer=yes
mailbox=6001
call-limit=100
type=peer
fullname=User 2
registersip=no
host=dynamic
callgroup=1
type=peer
context=DLPN_internal
cid_number=6001
hasvoicemail=yes
vmsecret=12345
email=Frodo@baggins.com
threewaycalling=no
hasdirectory=no
callwaiting=no
hasmanager=no
hasagent=no
hassip=yes
hasiax=yes
secret=
nat=yes
canreinvite=yes
dtmfmode=rfc2833
insecure=no
pickupgroup=1
macaddress=00215A70AF6B
autoprov=yes
label=6001
linenumber=1
LINEKEYS=1
callcounter=yes
disallow=all
allow=speex,alaw,gsm,ulaw
[trunk_1]
host=192.168.16.49
username=
secret=
trunkname=NorthStar_SIP
context=DID_trunk_1
group=null
hasexten=no
hasiax=no
hassip=yes
registeriax=no
registersip=yes
trunkstyle=voip
disallow=all
allow=all

david55 · October 1, 2010, 9:49am

On a quick scan, I can find devices 6000 and 6001, but no corresponding extensions.

However, for a newbie, I would strongly recommend buying or downloading Asterisk: The Future of Telephone, and learning to set up simple configurations directly. Once you understand how the configuration works, you can then add the GUI to ease day to day maintenance.

Finding the logging output and setting the appropriate level is something that anyone seeking support already needs to have learned.

You can probably do it by typing “asterisk -rvvv” at the shell prompt.

Many Asterisk installations also provide this information on console 9(?) (ctrl-alt-F9).