[solved] Internal extension problem

Thanks to the help hear, I found my outgoing call origination problem with Broadvoice. But somehow when I got that corrected, I lost my abilityu to dial extensions. It says in the asterisk CLI that “extension XXX not found”.

Sip peers shows all extensions registered, and they can all make outgoing calls thru the provider. Do I need something else in my dialplan? Note that before the outoing was working, the extensions worked fine. I am confused why they worked, why they do not now, and what to do!

Here is my sip and extensions:

sip.conf

[authentication]

[general]
disallow=all
allow = ulaw
allow=alaw
allow=gsm
allow=g729
allowoverlap = no
authname = 2122021963
bindaddr = 0.0.0.0
bindport = 5060
context = default
register = 2122021963@sip.broadvoice.com:Password:2122021963@sip.broadvoice.com/707
registerattempts = 20
registertimeout = 30
callerid = “ESFVCI Support”

[sip.broadvoice.com]
disallow=all
allow = ulaw
allow=alaw
allow=gsm
allow=g729
authname=2122021963
context=from-broadvoice
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=2122021963
host=sip.broadvoice.com
;insecure=very
secret=password
type=peer
user=phone
username=2122021963

[706]
callerid = "ESFVCI Support"
canreinvite = no
dtmfmode = rfc2833
host = dynamic
nat = no
secret =
username = 706
type = friend

[708]
type = friend
host = dynamic
dtmfmode = rfc2833
nat = no
username = 708
secret = canreinvite = no
callerid = “ESFVCI Support”

[707]
type = friend
host = dynamic
dtmfmode = rfc2833
nat = no
username = 707
secret =
canreinvite = no
callerid = “VCI”

end sip.conf

============================
begin extensions.conf

[general]
static = yes
writeprotect = no
userscontext = default
clearglobalvars = yes

[default]
; from broadvoice support docs:
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()
; end broadvoice docs

[DID_trunk_1]
include = DID_trunk_1_default

[DID_trunk_1_default]
include = DLPN_DialPlan1

[CallingRule_Rule1]
; not real sure how or why I put this in, since using broadvoice for
; all outbound, there is no backup trunk
exten = 1NXXNXXXXXX,1,Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},trunk_1,)

[DLPN_DialPlan1]
include = CallingRule_Rule1
include = pagegroups
include = conferences
include = default

[from-broadvoice]
; i think the incoming ivr will go here

[pagegroups]
exten = 771,1,Macro(pagingintercom,SIP/706&SIP/708,d)

[conferences]
exten = 751,1,MeetMe(${EXTEN},MsIx)
exten = 752,1,MeetMe(751,MsIxaA)

[globals]
CONSOLE = Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
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
GLOBAL_OUTBOUNDCID = 2122021963
GLOBAL_OUTBOUNDCIDNAME = VCI-ESF
trunk_1 = SIP/trunk_1
CID_trunk_1 = VCI-ESF
CID_707 = 707

[macro-stdexten]
; -------------------------------------------------------
; no idea why need this, not referenced that I can find,
; but if i delete it the whole system fails to work
; ----------------------------------------------------------
exten = s,1,Set(_DYNAMIC_FEATURES=${FEATURES})
exten = s,2,GotoIf($["${FOLLOWME
${ARG1}}" = “1”]?5:3)
exten = s,3,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,4,Goto(s-${DIALSTATUS},1)
exten = s,5,Macro(stdexten-followme,${ARG1},${ARG2})
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})

[macro-stdexten-followme]
; see note above about deleting
exten = s,1,Answer
exten = s,2,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,3,Set(__FMCIDNUM=${CALLERID(num)})
exten = s,4,Set(__FMCIDNAME=${CALLERID(name)})
exten = s,5,Followme(${ARG1},${FOLLOWMEOPTIONS})
exten = s,6,Voicemail(${ARG1},u)
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
[macro-pagingintercom]
exten = s,1,SIPAddHeader(Alert-Info: ${PAGING_HEADER})
exten = s,2,Page(${ARG1},${ARG2})
exten = s,3,Hangup

[ringgroups]
[queues]
[voicemenus]
[voicemailgroups]
[directory]
[page_an_extension]

[macro-trunkdial-failover-0.3]
; see note above about deleting
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()

end extensions.conf

Thanks!

You need an edition to the dialplan to allow calling to to your 7XX extensions.

What would such an addition look like? (It had always worked before as is.)

I tried this:
exten => 706, 1, answer()
exten => 706, 2, dial(sip${exten},15)
exten => 706, 3, voicemail({$exten})

and can ring the extension, but the voicemail part does not work. It says in asterisk no voicemail application for 706.

This:

exten => 706, 1, answer()
exten => 706, 2, dial(sip${exten},15)
exten => 706, 3, voicemail({$exten}) 

should be:

exten => 706, 1, Dial(sip/${EXTEN},15)
exten => 706, 2, Voicemail({$EXTEN}) 
  1. In general it is a bad habit to Answer the call unless it was really answered. If you have the Answer in there you do not know if the remote user really picked up. The Answer command sends Answer supervision.
  2. You needed a / between SIP and ${EXTEN}.
  3. Did you set up mailbox 706 in voicemail.conf ?

There was a /, I was just cut-and-paste challenged.

I removed the 1st line with the answer. Then I double checked voicemail.conf:
[general]
format = wav49|gsm|wav
serveremail = support@blah.com
attach = yes
maxmsg = 100
maxsecs = 600
minsecs = 3
maxgreet = 180
skipms = 3000
maxsilence = 20
silencethreshold = 128
maxlogins = 6
moveheard = yes
forward_urgent_auto = no
userscontext = default
fromstring = The VCI Phone System
;odbcstorage=asterisk
;odbctable=voicemessages
emailsubject = [PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
emailbody = Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t–Asterisk\n
emaildateformat = %A, %B %d, %Y at %r
mailcmd = /usr/sbin/sendmail -t
sendvoicemail = yes ; Allow the user to compose and send a voicemail while inside
maxgreet = 30
operator = no
maxmsg = 100
maxmessage = 300
minmessage = 1
saycid = no
sayduration = yes
saydurationm = 0
envelope = yes
review = no
; listen-control-forward-key=# ; Customize the key that fast-forwards message playback
; listen-control-reverse-key=* ; Customize the key that rewinds message playback
; listen-control-pause-key=0 ; Customize the key that pauses/unpauses message playback
; listen-control-restart-key=2 ; Customize the key that restarts message playback
; listen-control-stop-key=13456789 ; Customize the keys that interrupt message playback, probably all keys not set above

[zonemessages]
eastern = America/New_York|‘vm-received’ Q ‘digits/at’ IMp
central = America/Chicago|‘vm-received’ Q ‘digits/at’ IMp
central24 = America/Chicago|‘vm-received’ q ‘digits/at’ H N 'hours’
military = Zulu|‘vm-received’ q ‘digits/at’ H N ‘hours’ 'phonetic/z_p’
european = Europe/Copenhagen|‘vm-received’ a d b ‘digits/at’ HM

[default]
706 => 1111,Shana,shana@blah.com,attach=yes|saycid=no|envelope=yes|delete=no
707 => 1111,Tuvia Vinitsky,tuvia@blah.com,attach=yes|saycid=no|envelope=yes|delete=no
708 => 1111,Naomi,naomi@blah.com,attach=yes|saycid=no|envelope=yes|delete=no
730 => 0000,SupportVMAN,support@vinitskyinc.com,attach=yes|saycid=no|envelope=yes|delete=no
731 => 0000,SupportESF,support@vinitskyinc.com,attach=yes|saycid=no|envelope=yes|delete=no
732 => 0000,Sales,support@vinitskyinc.com,attach=yes|saycid=no|envelope=yes|delete=no
733 => 0000,Operator,support@vinitskyinc.com,attach=yes|saycid=no|envelope=yes|delete=no
734 => 0000,Finance,support@vinitskyinc.com,attach=yes|saycid=no|envelope=yes|delete=no

[other]

Same response as my last. Get me in to the box and I will have a peak.

Anyone got any ideas here?

Found a type in the voicemail command. All is good.