from:"Steve Hickel"

[asterisk-users] RDNIS Question

2008-05-20 Thread Steve Hickel

We are running Asterisk 1.4. Does anyone have thought about how to pass
RDNIS from a callmanager to a just an Asterisk 1.4 voicemail server?

I have tried to use CALLERID(rdnis) with a NoOp statement prior, but I
get confused by the syntax of whether or not to use the SET in front of
the CALLERID(rdnis). 

Thoughts?

Steve




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Re: [asterisk-users] Out-Going Calleriid

2008-05-20 Thread Steve Hickel

Perhaps your carrier has changed the way they handle outbound caller id.
Could they be blocking it somehow all of a sudden?

Might be worth a phone call?

steve

On Thu, 2008-05-08 at 08:29 +0100, Tim Guy wrote:
 Thanks for the heads up again guys.
 
 Still no go.
 
 It's a ISDN30 PRI on NTL(Virgin) in the UK
 
 I currently have a Mitel 3300 connected happily sending CallerID's so I
 know it the teleco supports it. 
 
 The Mitel is set to send 01926xx so that's what I'm trying to get
 Asterisk to send.
 
 Running an Openvox D210E that runs with wct4xxp drivers.
 
 It definitely work before so it must be something I've done, OR, a
 certain driver / zaptel version
 
 Caller id in-coming is fine, just won't send out.
 
 Huff
 
 Tim
 p.s Sorry for the disclaimer. Should be gone on this one.
 
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Re: [asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...

2008-05-06 Thread Steve Hickel

These are the instructions that I followed. I did managed to get the
fast busy to go away, but the RDNIS simply does not seem to work. These
are the instructions that I followed on this project. I have run out of
time trying to get Call Manager 4.x to talk to Asterisk 1.4. 

http://www.voip-info.org/wiki/index.php?page_id=2596#editcomments

These instructions although a good start, simply lack the pictures or
images to set up CCM properly, and because of the coding change from
earlier versions, this just doesn't seem to allow voice mail to work.

I have learned a lot about asterisk, but am frustrated by this
experience.

Thanks Sean for the info about the change of the rdnis command format.

Kind regards,

Steve

On Mon, 2008-05-05 at 23:33 -0400, Steve Hickel wrote:
 Sean,
 
 Here is what I changed. Now I have a fast busy... 
 
 Steve
 
  [demo]
   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,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
   exten=1235,1,Voicemail(1234,u)
   exten=1236,1,Dial(Console/dsp)
   exten=1236,n,Voicemail(1234,b)
   exten=#,1,Playback(demo-thanks)
   exten=#,n,Hangup
   exten=t,1,Goto(#,1)
   exten=i,1,Playback(invalid)
   exten=500,1,Playback(demo-abouttotry)
   exten=500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
   exten=500,n,Playback(demo-nogo)
   exten=500,n,Goto(s,6)
   exten=600,1,Playback(demo-echotest)
   exten=600,n,Echo
   exten=600,n,Playback(demo-echodone)
   exten=600,n,Goto(s,6)
   exten=76245,1,Macro(page,SIP/Grandstream1)
   exten=_7XXX,1,Macro(page,SIP/${EXTEN})
   exten=7999,1,Set(TIMEOUT(absolute)=60)
   exten=7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL 
 PROTECTED]Local/[EMAIL PROTECTED]/n|d)
   exten=,1,VoicemailMain
   exten=,n,Goto(s,6)
 
   [general]
   static=yes
   writeprotect=no
   clearglobalvars=no
   autofallthrough=yes
   priorityjumping=no
   
 
   [default]
   exten=_230,1,SetCallerID(${EXTEN:3})
   exten=_230,2,Dial(SIP/[EMAIL PROTECTED])
   exten=_230,3,Answer
   exten=_230,4,Wait,1
   exten=_230,5,Hangup
   exten=_231,1,SetCallerID(${EXTEN:3})
   exten=_231,2,Dial(SIP/[EMAIL PROTECTED])
   exten=_231,3,Answer
   exten=_231,4,Wait,1
   exten=_231,5,Hangup
   exten=,1,VoiceMailMain
 
   [incoming]
   exten=,1,GotoIf($[${CALLERID(rdnis)}]?2:400)
   exten=,2,MailboxExists(${CALLERID(rdnis)[EMAIL PROTECTED])
   exten=,3,Congestion
   exten=,103,Voicemail(su${CALLERID(rdnis)}
   exten=,104,Playback(vm-goodbye)
   exten=,105,Hangup
   exten=,400,VoicemailMain
 
 __
 From: Sean Dennis [mailto:[EMAIL PROTECTED]
 To: [EMAIL PROTECTED], Asterisk Users Mailing List -
 Non-Commercial Discussion
 [mailto:[EMAIL PROTECTED]
 Sent: Mon, 05 May 2008 17:58:32 -0400
 Subject: Re: [asterisk-users] Call manager using Asterisk as
 voicemail server (SIP) not working ...
 
 Steve Hickel wrote:
  I have sip set up on Callmanager 4.x. When others call my
 ext of 2016 on
  ccm after a busy or no answer, asterisk voice mail answers
 by saying,
  Mailbox  password. I want it to put them into my
 mailbox so they
  can leave a message. Somehow I must be missing something...
 Please
  help! 
 
  I have spent 19 hours easy on trying to figure this one
 out. 
 
  SIP DN is  on CCM 
  VOICEMAIL on Asterisk is . 
 
  Here is my sip.conf: 
 
  [general] 
  context=default 
  allowoverlap=no 
  bindport=5060 
  bindaddr=0.0.0.0 
  srvlookup=yes 
  allowexternaldomains=yes 
  allowexternalinvites=no 
  allowguest=yes 
  allowsubscribe=no 
  allowtransfer=yes 
  alwaysauthreject=no 
  autodomain=no 
  callevents=no 
  compactheaders=no 
  dumphistory=no 
  g726nonstandard=no 
  ignoreregexpire=no 
  jbenable=no 
  jbforce=no 
  jblog=no 
  maxcallbitrate=384 
  maxexpiry=3600 
  minexpiry=60 
  nat=no 
  notifyringing=no 
  pedantic=no 
  promiscredir=no 
  recordhistory=no 
  relaxdtmf=no 
  rtcachefriends=no 
  rtsavesysname=no 
  rtupdate=no 
  sendrpid=yes

[asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...

2008-05-05 Thread Steve Hickel

I have sip set up on Callmanager 4.x. When others call my ext of 2016 on
ccm after a busy or no answer, asterisk voice mail answers by saying,
Mailbox  password. I want it to put them into my mailbox so they
can leave a message. Somehow I must be missing something... Please
help! 

I have spent 19 hours easy on trying to figure this one out. 

SIP DN is  on CCM 
VOICEMAIL on Asterisk is . 

Here is my sip.conf: 

[general] 
context=default 
allowoverlap=no 
bindport=5060 
bindaddr=0.0.0.0 
srvlookup=yes 
allowexternaldomains=yes 
allowexternalinvites=no 
allowguest=yes 
allowsubscribe=no 
allowtransfer=yes 
alwaysauthreject=no 
autodomain=no 
callevents=no 
compactheaders=no 
dumphistory=no 
g726nonstandard=no 
ignoreregexpire=no 
jbenable=no 
jbforce=no 
jblog=no 
maxcallbitrate=384 
maxexpiry=3600 
minexpiry=60 
nat=no 
notifyringing=no 
pedantic=no 
promiscredir=no 
recordhistory=no 
relaxdtmf=no 
rtcachefriends=no 
rtsavesysname=no 
rtupdate=no 
sendrpid=yes 
sipdebug=no 
t1min=100 
t38pt_udptl=no 
[authentication] 

[sip] 
type=friend 
context=incoming 
host=172.20.1.57 
ipaddr=172.20.1.57 
allow=ulaw 
allow=alaw 
nat=no 
canreinvite=yes 
qualify=yes 

Here is my voicemail.conf 

[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 
[other] 

[general] 
format=wav49|gsm|wav 
serveremail=asterisk 
attach=yes 
skipms=3000 
maxsilence=10 
silencethreshold=128 
maxlogins=3 
emaildateformat=%A, %B %d, %Y at %r 
sendvoicemail=yes 
attachfmt=wav 
deletevoicemail=no 
envelope=no 
maxgreet=60 
maxmessage=120 
maxmsg=100 
minmessage=1 
operator=yes 
review=yes 
saycid=no 
sayduration=yes 
mailcmd=/usr/sbin/sendmail -t 
externotify=/var/libasterisk/scripts/vm.sh 
[default] 
2016=1234,Steve,[EMAIL PROTECTED] 

Here is the relevant parts of my extensions.conf: 

[macro-dialout-callmanager] 
exten=s,1,ChanIsAvail(SIP/sip) 
exten=s,2,Cut(AVAILCHAN=AVAILCHAN,,1) 
exten=s,3,Dial(${AVAILCHAN}/${ARG1}) 
exten=s,4,Hangup 
exten=s,102,Congestion 
[incoming] 
exten=,1,GotoIf($[${RDNIS}]?2:400) 
exten=,2,MailboxExists([EMAIL PROTECTED] 
exten=,3,Congestion 
exten=,103,Voicemail(su${RDNIS}) 
exten=,104,Playback(vm-goodbye) 
exten=,105,Hangup 
exten=,400,VoicemailMain 
[general] 
static=yes 
writeprotect=no 
clearglobalvars=no 
autofallthrough=yes 
priorityjumping=no 
[default] 
exten=_230,1,SetCallerID(${EXTEN:3}) 
exten=_230,2,Dial(SIP/[EMAIL PROTECTED]) 
exten=_230,3,Answer 
exten=_230,4,Wait,1 
exten=_230,5,Hangup 
exten=_231,1,SetCallerID(${EXTEN:3}) 
exten=_231,2,Dial(SIP/[EMAIL PROTECTED]) 
exten=_231,3,Answer 
exten=_231,4,Wait,1 
exten=_231,5,Hangup 

I am using users.conf, but don't know how that ties in or whether I even
need it...??? 

thanks, 

Steve



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Re: [asterisk-users] Call manager using Asterisk as vo icemail server (SIP) not working ...

2008-05-05 Thread Steve Hickel

Sean,

Here is what I changed. Now I have a fast busy... 

Steve

 [demo]
  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,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
  exten=1235,1,Voicemail(1234,u)
  exten=1236,1,Dial(Console/dsp)
  exten=1236,n,Voicemail(1234,b)
  exten=#,1,Playback(demo-thanks)
  exten=#,n,Hangup
  exten=t,1,Goto(#,1)
  exten=i,1,Playback(invalid)
  exten=500,1,Playback(demo-abouttotry)
  exten=500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
  exten=500,n,Playback(demo-nogo)
  exten=500,n,Goto(s,6)
  exten=600,1,Playback(demo-echotest)
  exten=600,n,Echo
  exten=600,n,Playback(demo-echodone)
  exten=600,n,Goto(s,6)
  exten=76245,1,Macro(page,SIP/Grandstream1)
  exten=_7XXX,1,Macro(page,SIP/${EXTEN})
  exten=7999,1,Set(TIMEOUT(absolute)=60)
  exten=7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL 
PROTECTED]Local/[EMAIL PROTECTED]/n|d)
  exten=,1,VoicemailMain
  exten=,n,Goto(s,6)

  [general]
  static=yes
  writeprotect=no
  clearglobalvars=no
  autofallthrough=yes
  priorityjumping=no

   
   
  
 
  
  [default]
  exten=_230,1,SetCallerID(${EXTEN:3})
  exten=_230,2,Dial(SIP/[EMAIL PROTECTED])
  exten=_230,3,Answer
  exten=_230,4,Wait,1
  exten=_230,5,Hangup
  exten=_231,1,SetCallerID(${EXTEN:3})
  exten=_231,2,Dial(SIP/[EMAIL PROTECTED])
  exten=_231,3,Answer
  exten=_231,4,Wait,1
  exten=_231,5,Hangup
  exten=,1,VoiceMailMain

  [incoming]  exten=,1,GotoIf($[${CALLERID(rdnis)}]?2:400)
  exten=,2,MailboxExists(${CALLERID(rdnis)[EMAIL PROTECTED])
  exten=,3,Congestion
  exten=,103,Voicemail(su${CALLERID(rdnis)}
  exten=,104,Playback(vm-goodbye)
  exten=,105,Hangup
  exten=,400,VoicemailMain
  _  

From: Sean Dennis [mailto:[EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion 
[mailto:[EMAIL PROTECTED]
Sent: Mon, 05 May 2008 17:58:32 -0400
Subject: Re: [asterisk-users] Call manager using Asterisk as voicemail server 
(SIP) not working ...

Steve Hickel wrote:
   I have sip set up on Callmanager 4.x. When others call my ext of 2016 on
   ccm after a busy or no answer, asterisk voice mail answers by saying,
   Mailbox  password. I want it to put them into my mailbox so they
   can leave a message. Somehow I must be missing something... Please
   help! 
  
   I have spent 19 hours easy on trying to figure this one out. 
  
   SIP DN is  on CCM 
   VOICEMAIL on Asterisk is . 
  
   Here is my sip.conf: 
  
   [general] 
   context=default 
   allowoverlap=no 
   bindport=5060 
   bindaddr=0.0.0.0 
   srvlookup=yes 
   allowexternaldomains=yes 
   allowexternalinvites=no 
   allowguest=yes 
   allowsubscribe=no 
   allowtransfer=yes 
   alwaysauthreject=no 
   autodomain=no 
   callevents=no 
   compactheaders=no 
   dumphistory=no 
   g726nonstandard=no 
   ignoreregexpire=no 
   jbenable=no 
   jbforce=no 
   jblog=no 
   maxcallbitrate=384 
   maxexpiry=3600 
   minexpiry=60 
   nat=no 
   notifyringing=no 
   pedantic=no 
   promiscredir=no 
   recordhistory=no 
   relaxdtmf=no 
   rtcachefriends=no 
   rtsavesysname=no 
   rtupdate=no 
   sendrpid=yes 
   sipdebug=no 
   t1min=100 
   t38pt_udptl=no 
   [authentication] 
  
   [sip] 
   type=friend 
   context=incoming 
   host=172.20.1.57 
   ipaddr=172.20.1.57 
   allow=ulaw 
   allow=alaw 
   nat=no 
   canreinvite=yes 
   qualify=yes 
  
   Here is my voicemail.conf 
  
   [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 
   [other] 
  
   [general] 
   format=wav49|gsm|wav 
   serveremail=asterisk 
   attach=yes 
   skipms=3000 
   maxsilence=10 
   silencethreshold=128 
   maxlogins=3 
   emaildateformat=%A, %B %d, %Y at %r 
   sendvoicemail=yes 
   attachfmt=wav 
   deletevoicemail=no 
   envelope=no 
   maxgreet=60 
   maxmessage=120 
   maxmsg=100 
   minmessage=1 
   operator=yes 
   review=yes 
   saycid=no 
   sayduration=yes 
   mailcmd=/usr/sbin/sendmail -t 
   externotify=/var/libasterisk/scripts/vm.sh 
   [default] 
   2016=1234,Steve,[EMAIL PROTECTED] 
  
   Here is the relevant parts of my extensions.conf: 
  
   [macro-dialout-callmanager] 
   exten=s,1,ChanIsAvail(SIP/sip) 
   exten=s,2,Cut(AVAILCHAN=AVAILCHAN,,1) 
   exten=s,3,Dial(${AVAILCHAN}/${ARG1}) 
   exten=s,4

[asterisk-users] RDNIS Question

Re: [asterisk-users] Out-Going Calleriid

Re: [asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...

[asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...

Re: [asterisk-users] Call manager using Asterisk as vo icemail server (SIP) not working ...

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