login | register
Wed 01 of Aug, 2007 [09:27 UTC]

voip-info.org

Navigation
Today's Deals
Recently Visited Pages
Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Last Changes
Shoutbox
  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.50s
  • Memory usage: 2.24MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 2.51

Asterisk tips voicemail live

Voicemail Live

Answering machine mimic: Listen while caller is leaving voicemail for you; with pick-up option
Posted by Philipp von Klitzing in Dec. 2005

Idea

You'd like to have the PBX voicemail act just like your answering machine at home? It can be done!
First of all you'd probably want to listen in on the new call while the caller dictates her message to the system. Then, if you wish, press # to stop the recording and connect to the caller. All this is most useful if your phone comes with auto-answer/intercom support and a decent speaker.

Concept

  • ideally you have a multi-line speaker phone that allows to configure one of the lines for auto-answer/intercom (SNOM or the like)
  • place caller, voicemail and speaker phone into a first dynamic MeetMe conference with the help of AGI-generated .call files
  • dissolve that first MeetMe upon a # key press by the callee on the speaker phone
  • now engage in a second dynamic MeetMe to begin a conversation; we allow for a maximum of two simultaneous instances of this second MeetMe

Potential enhancements

Note: For testing purposes the freely available (for private use) SNOM softphone proved to be a useful tool

  1. Use app_bridge instead of 2nd MeetMe for duplex talking - bug/patch 5841
  2. Consider ChanSpy instead of the 1st MeetMe (for listening in)
  3. Insert an astdb (DBGet/Put) switch for "Voicemail Live" on/off
  4. Use 'dE' instead of just 'd' to create a free dynamic conference and then do away with MeetMeCount
  5. Find out why we sometimes get the VM prompts rather late?! Possible cause is CPU load, should be low (check with 'top')
  6. Dedicate a SNOM button of type "DTMF" to issue the # and label it "voicemail pick-up"
  7. Use the SNOM "URL Action" for Off-Hook to talk to the vm caller by simply lifting the handset (no need to press #)
  8. Rewrite the dialplan part without priority jumping
  9. Play the VM intro _before_ entering the MeetMe room so that the listener (=VM box owner) does not always have to listen to it
  10. Fix me: If the caller hangs up before having entered MeetMe, then the listener and voicemail will remain in MeetMe until time out . Cause: the x option of MeetMe first of all needs to marked user to enter before it gets a chance to close the conference. Solutions:
    1. use ChanSpy instead of MeetMe (but then owner can't exit by pressing #), or
    2. let a 4th fake & marked user (option A) quickly enter and leave the MeetMe room after caller hangup (this caller's presence must then overlap with the entry of the caller into MeetMe; we would need to stay below vm config for minimum message duration), or
    3. set a channel variable as flag, and then use the h extension to call MeetMeAdmin on hangup and kick those that remain in MeetMe (downsides: a channel variable is probably already destroyed when we reach the h extension, but maybe we can address this with a global variable? Also employing the h extension might mess up our CDRs a little, and this h extension will be called by any closing call, not just our VM calls)
  11. consider to introduce the 'w' flag for the listener and set up special music-on-hold that plays pick-up instructions for the owner/listener
  12. check if $agi["callerid"] gives us trouble when a Caller ID Name is present and we thus have a space in the filename

extensions.conf

Note: You'll most probably need priorityjumping=yes in the [general] section if you are using Asterik v1.2.x. If you don't like that then you'll need to rework the parts where jumpts to n+101 are performed, e.g. look for Dial() and ChanIsAvail()

 [vm-meet-listener]
 ; We need this (and the Local channel) in order to set the Caller ID correctly
 ; Used by macro-vm-meetme (initiated by macro-vm-listen.agi)
 exten => listen,1,NoOp  ; Maybe set a timeout here > vm max. message length
 ; if you have a SNOM then EITHER set the header below, OR configure the phone's line to auto-answer
 ;exten => listen,2,SIPAddHeader(Call-Info: <sip:domain>\;answer-after=0)       ; try SNOM auto-answer, add your domain
 exten => listen,2,NoOp
 exten => listen,3,Set(CALLERID(Name)=${ORIGCIDNAME})
 exten => listen,4,Set(CALLERID(Num)=${ORIGCIDNUM})
 exten => listen,5,Dial(${TARGET},5)    ; give intercom/auto-answer 5 sec to answer
 
 exten => t,1,NoOp(=== The auto-answer setup of ${TARGET} might not be correctly configured ===)
 exten => t,2,HangUp

 [vm-meet-join]
 ; — Dialplan logic for the callee (aka 'silent listener' aka 'vm-owner') --
 ; Press # to exit the voicemail meetme room and kick the calling user
 ; OPTIONAL: Dedicate a "DTMF" type SNOM button to send #
 ; REQUIRES: Language directory "mm" with silenced "conf-kicked.gsm" soundfile (short!)
 ;        e.g. [root@sounds]# cp silence/1.gsm mm/conf-kicked.gsm
 ; REQUIRES: vm-instructions-short.gsm (2nd half of vm-instructions.gsm)
 ; NOTE: The two bugs/patches mentioned below are not necessary for Asterisk v1.2.1 or later
 ; NOTE: Use bug/patch 5773 to fix MeetMe X option in Asterisk v1.2.0 (ref. bug 5631)
 ; NOTE: Use bug/patch 5810 to pass variables from .call files to Local channels in Asterisk v1.2.0
 ; NOTE: We can't transfer from within 2nd MeetMe
 exten => join,1,Set(TIMEOUT(absolute)=300)
 ; Possible ToDo here: Check with MeetMeCount if 99${ROOMNO} is empty = caller already hung up
 exten => join,2,Playback(vm-instructions-short) ; play "#-pickup" instructions
 exten => join,3,Wait(.5)
 exten => join,4,NoOp
 exten => join,5,MeetMe(99${ROOMNO}|dmqpx)       ; due to playback we enter after caller
 exten => join,6,MeetMeAdmin(99${ROOMNO},K)      ; Kick all users after exiting MeetMe by pressing #
 exten => join,7,MeetMeCount(98${ROOMNO}|mcount)
 exten => join,8,GotoIf($[${mcount} > 1]?13)
 exten => join,9,Set(ROOMSELECT=98${ROOMNO})     ; remember the chosen room
 exten => join,10,MeetMe(98${ROOMNO}|dqpx)       ; TODO for future: Use app_bridge instead!
 exten => join,11,MeetMeAdmin(98${ROOMNO},K)     ; Kick with # to make sure its closed
 exten => join,12,HangUp
 exten => join,13,MeetMeCount(97${ROOMNO}|mcount)
 exten => join,14,GotoIf($[${mcount} > 1]?18)
 exten => join,15,Set(ROOMSELECT=97${ROOMNO})
 exten => join,16,MeetMe(97${ROOMNO}|dqpx)       ; TODO for future: Use app_bridge instead!
 exten => join,17,MeetMeAdmin(97${ROOMNO},K)     ; Kick with # to make sure its closed
 exten => join,18,HangUp                         ; we dont allow more than 2 vm callers

 exten => h,1,GotoIf($["${ROOMSELECT}" = ""]?3)
 exten => h,2,MeetMeAdmin(${ROOMSELECT},K)       ; in case we went on-hook before caller
 exten => h,3,NoOp(ROOMSELECT=${ROOMSELECT})

 [vm-meet-exit]
 ; — Exit context for the VM caller (for pressing #) --
 ; This way we can distinguish between own # and being kicked by MeetMeAdmin!
 exten => #,1,NoOp(== VM caller pressed: # ==)
 exten => #,2,HangUp

 [macro-vm-meetme]
 ; — This is the main macro for 'voicemail live' handling the caller --
 ;   ${ARG1} - Device(s) to ring (TARGET)
 ;   ${ARG2} - Extension
 ;   ${ARG3} - Mailbox
 ;
 ; Required contexts: vm-meet-listener and vm-meet-join and vm-meet-exit
 ;
 ; — we are unavailable; don't go here if we are busy! --
 exten => s,1,Set(TIMEOUT(absolute)=300)         ; need to synchronize this with max vm rec
 exten => s,2,Set(TARGET=${ARG1})                ; we read this in AGI
 exten => s,3,ChanIsAvail(${ARG1},j)             ; bug: only works for SIP peers?!
 exten => s,4,Set(ORIGLANG=LANGUAGE)             ; preserve the original language setting
 exten => s,5,MeetMeCount((99${ARG2}|mcount)
 exten => s,6,GotoIf($[${mcount} > 0]?101)         ; do we already have someone on VM recording?
 exten => s,7,Agi(macro-vm-listen.agi)           ; put the VM owner into MeetMe for Intercom
 exten => s,8,Wait(.1)
 exten => s,9,Agi(macro-vm-record.agi)           ; put the VM record app into MeetMe
 exten => s,10,Playback(beep)                    ; play short sound so caller knows we answered
 ; at this point we sometimes get a delay with silence due to slow call-setup work of Asterisk
 exten => s,11,Set(LANGUAGE()=mm)                ; we have to avoid "You have been kicked..."!
                                                 ; replace conf-kicked.gsm with plain silence
 exten => s,12,Set(MEETME_EXIT_CONTEXT=vm-meet-exit)
 ; now put the caller into MeetMe together with voicemail and the silent listener
 exten => s,13,MeetMe(99${ARG2},dAXq)            ; Option X works after applying patch 5773
 exten => s,14,MeetMeCount((98${ARG2}|mcount)    ; we arrive here after 1) kick or 2) # press
 exten => s,15,GotoIf($[${mcount} > 1]?18)
 exten => s,16,MeetMe(98${ARG2}|dAM)
 exten => s,17,HangUp
 exten => s,18,MeetMeCount((97${ARG2}|mcount)
 exten => s,19,GotoIf($[${mcount} > 1]?101)        ; we tried twice, so no meetme, now pure vm
 exten => s,20,MeetMe(97${ARG2}|dAM)
 exten => s,21,HangUp
 exten => s,101,Voicemail(u${ARG3})              ; we already have another VM-MeetMe active
 exten => s,102,HangUp
 exten => s,104,Goto(101)                        ; ChanIsAvail gave a negative result

 [default]
 ;  Join voicemail to meetme (for macro-vm-meetme) 
 ; NOTE: Adjust macro-vm-record.agi accordingly if you change the '8600' prefix here
 exten => _8600X.,1,Set(CALLERID(name)=${ORIGCIDNAME}) ; needed for the e-mail notification
 exten => _8600X.,2,Set(CALLERID(number)=${ORIGCIDNUM})
 exten => _8600X.,3,VoiceMail(u${EXTEN:4})
 exten => _8600X.,4,HangUp

 ; here's the extension for our own phone (SIP/myPeer); send caller to voicemail if we are not available
 exten => 1234,1,Dial(SIP/myPeer,20,t)
 exten => 1234,2,Macro(vm-meetme,SIP/myPeer,1234,881234)  ; we are unavailable
 exten => 1234,3,HangUp
 exten => 1234,102,Voicemail(b881234)
 exten => 1234,103,HangUp


AGI scripts (PHP)

On Debian you'll need 'php-cli' installed for this. Of course any other language can do the job as well, you are not bound to PHP.

macro-vm-listen.agi


 #!/usr/bin/php -q
 <?php

 ob_implicit_flush(true);
 set_time_limit(5);
 $in = fopen("php://stdin","r");

 // toggle debugging output (more verbose)
 $debug = false;
 //$debug = true;

function __read__() {
  global $in, $debug;
  $input = str_replace("\n", "", fgets($in, 4096));
  if ($debug) echo "VERBOSE \"read: $input\"\n";
  return $input;
}

function __write__($line) {
   global $debug;
   if ($debug) echo "VERBOSE \"write: $line\"\n";
   print $line."\n";
}

//read the standard agi variables
while (!feof($in)) {
       $temp = str_replace("\n","",fgets($in,4096));
       $s = split(":",$temp);
       $agi[str_replace("agi_","",$s[0])] = trim($s[1]);
       if (($temp == "") || ($temp == "\n")) {
               break;
       }
}

 //get the variables and strip off all the extra stuff around
 __write__("GET VARIABLE TARGET");
 $res = substr(strrchr(__read__(),"("),1,-1);
 __write__("GET VARIABLE ARG2");
 $arg2 = substr(strrchr(__read__(),"("),1,-1);

 $cf = fopen("/tmp/cb".$agi["callerid"],"w+");
 fputs($cf,"Channel: Local/listen@vm-meet-listener/n\n");
 fputs($cf,"Set: _ORIGCIDNAME=".$agi["calleridname"]."\n");
 fputs($cf,"Set: _ORIGCIDNUM=".$agi["callerid"]."\n");
 fputs($cf,"Set: CALLERID(name)=".$agi["calleridname"]."\n");
 fputs($cf,"Set: CALLERID(number)=".$agi["callerid"]."\n");
 fputs($cf,"Set: _TARGET=".$res."\n");
 fputs($cf,"Set: _ROOMNO=".$arg2."\n");
 fputs($cf,"MaxRetries: 0\n");
 fputs($cf,"RetryTime: 10\n");
 // --- We are the first so we create the dynamic conference ---
 fputs($cf,"Context: vm-meet-join\n");
 fputs($cf,"Extension: join\n");
 fputs($cf,"Priority: 1\n");

 //Now move (!) the file to the outgoing dir AFTER we closed it
 fclose($cf);
 exec("mv /tmp/cb".$agi["callerid"]." /var/spool/asterisk/outgoing");

 fclose($in);
 ?>

macro-vm-record.agi


 #!/usr/bin/php -q
 <?php

 ob_implicit_flush(true);
 set_time_limit(5);
 $in = fopen("php://stdin","r");

 // toggle debugging output (more verbose)
 $debug = false;
 //$debug = true;

function __read__() {
  global $in, $debug;
  $input = str_replace("\n", "", fgets($in, 4096));
  if ($debug) echo "VERBOSE \"read: $input\"\n";
  return $input;
}

function __write__($line) {
   global $debug;
   if ($debug) echo "VERBOSE \"write: $line\"\n";
   print $line."\n";
}

 //read the standard agi variables
 while (!feof($in)) {
       $temp = str_replace("\n","",fgets($in,4096));
       $s = split(":",$temp);
       $agi[str_replace("agi_","",$s[0])] = trim($s[1]);
       if (($temp == "") || ($temp == "\n")) {
               break;
       }
 }

 //get the variables and strip off all the extra stuff around
 __write__("GET VARIABLE ARG2");
 $arg2 = substr(strrchr(__read__(),"("),1,-1);
 __write__("GET VARIABLE ARG3");
 $arg3 = substr(strrchr(__read__(),"("),1,-1);

 $cf = fopen("/tmp/cb2_".$agi["callerid"],"w+");
 fputs($cf,"Channel: Local/8600".$arg3."@default/n\n");
 fputs($cf,"Set: CALLERID(name)=".$agi["calleridname"]."\n");
 fputs($cf,"Set: CALLERID(number)=".$agi["callerid"]."\n");
 fputs($cf,"Set: _ORIGCIDNAME=".$agi["calleridname"]."\n");
 fputs($cf,"Set: _ORIGCIDNUM=".$agi["callerid"]."\n");
 fputs($cf,"MaxRetries: 0\n");
 fputs($cf,"RetryTime: 10\n");
 fputs($cf,"Application: MeetMe\n");
 fputs($cf,"Data: 99".$arg2."|dpqx\n");

 //Now move (!) the file to the outgoing dir AFTER we closed it
 fclose($cf);
 exec("mv /tmp/cb2_".$agi["callerid"]." /var/spool/asterisk/outgoing/");

 fclose($in);
 ?>


See also


Go back to Asterisk tips and tricks

Created by JustRumours, Last modification by JustRumours on Wed 21 of Mar, 2007 [20:30 UTC]

Comments

Comments Filter

by rushowr on Wednesday 20 of September, 2006 [13:17:06 UTC]
I'll be posting my results once I get to this :) I've got a HUGE amount of stuff to finalize and then I'll be starting on enhancements :)

by Lacy Moore on Wednesday 20 of September, 2006 [08:04:10 UTC]
I've been playing with this for several hours tonight (more like 6), and finally have it working with a Polycom IP601. There are a few issues that will keep it from being deployed in the environment I had planned to. I'll probably keep playing with it, especially now that I have figured out what it is doing.

I think this is a great feature for a business environment, especially for those people who need to screen their calls. I've noticed that it leaves you hangin if the calling party hangs up. I think this is probably a limitation of MeetMe, and with the enhancements suggested this would no longer be a limitation.

by Mike on Thursday 10 of August, 2006 [21:44:02 UTC]
Anybody do any of enhancements that were recommended? Post!! :-)

by Lacy Moore on Thursday 06 of April, 2006 [16:07:09 UTC]
I'm having trouble following this. What is the SIP/myPeer? In this a context that needs to be added, or is this supposed to be replaced with something, or what? I've tried this, and it didn't seem to work, and the only thing is that I'm just not sure what the SIP/myPeer is doing. I did a saearch on mypeer and it looks to be a context in some installations, though what kind of context, I'm not sure.

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:
Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver