Asterisk HowTo's
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Asterisk Voicemail spool
If you do not want to store your voicemails on your Asterisk PBX server, you can edit voicemail.conf to delete them. However, using a program called Sendmail, you can make Asterisk email your PBX user the new voicemail, and then the voicemail gets deleted off the server. The following option needs to be yes in voicemail.conf
/etc/asterisk/voicemail.conf
delete=yes ; After notification, the voicemail is deleted from the server. [per-mailbox only]
; This is intended for use with users who wish to receive their voicemail ONLY by email.
This script to relieve this /bin/delvm, delete every 5 min. voicemail in the vm-spool queue (I'm not sure what this script does, other than maybe check voicemailboxes every 5 minutes, and delete periodically. Please edit in English)
#!/bin/bash # # build-in use crontab -e */5 * * * * /bin/delvm # Autor: Don.K at 6th Nov 2006
# define here your context, mainly is default cx=default
vmbox=`asterisk -rx "show voicemail users" | grep $cx | awk '{print $2}'`
if [ -e /var/spool/asterisk/voicemail/$cx/$vmbox/INBOX/msg0000.wav ]; then rm /var/spool/asterisk/voicemail/$cx/$vmbox/INBOX/* else exit 0 fi
Call Pickup
In order to be able to take over a call on another telephone, must be registered than first in the configuration features.conf as a pickup extension.
pickupexten = *8
Now we can implement extension the call pickup are with the option callgroup and pickupgroup and the value for the group the calls to take over are in sip.conf for the appropriate telephone registering.
[GXP2000] type = friend username = 201 secret = 1234 context = default dtmfmode = rfc2833 callgroup = 2 pickupgroup = 2 disallow = all allow=ulaw allow=alaw allow=g723.1 allow=g729
Feature Application
An Feature Application will most there used where no appropriate button exist on the phone. For instance the Call Transfer on Siemens Gigaset 450C or often other DECT Telephone without Transfer (TRNF) key.
features.conf
[general]
#include parking_additional.inc #include features_general_custom.conf
featuredigittimeout => 1000 ; Max time (ms) between digits for
[applicationmap]
#include features_applicationmap_additional.conf
[featuremap]
blindxfer => ## ; Blind Transfer atxfer => ** ; Attended Xfer automon => *1 ; One Touch Record
Re-dialing on Asterisk
If your telephone does not offer a redial feature, the following example can be used to re-dial the last number by dialing *5. The following example fits into extensions.conf
[default] include => macro-recall
exten => _X.,1,Macro(recall,${EXTEN}) exten => *5,1,DBget(toCall=redial/${CALLERID}) exten => *5,2,Macro(recall,${toCall}) exten => *5,102,Hangup()
[macro-recall] exten => s,1,DBput(redial/${CALLERID}=${ARG1}) exten => s,1,Dial(SIP/${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-BUSY,1,Voicemail(b${ARG1}) exten => _s-.,1,Goto(s-NOANSWER,1)
Asterisk GSM Soundformat
Sound files made on Windows are usually wave files. These wave files cannot be read and processed by Asterisk, so we will have to change these wave files into an Asterisk-compatible format, such as GSM. Using a software called SOX, we can do so. The following example shows how to convert foo.wav into foo.gsm using SOX.
sox foo.wav -r 8000 -c1 foo.gsm resample -ql
The converted gsm sound files after copy /var/lib/asterisk/sounds/custom there are asterisk compatible wave formats
sox wav_file.wav gsm_file.gsm sox gsm_file.gsm wav_file.wav sox unnow_wav_file.wav asterisk_wav_file.wav
Call forwarding
This implementation allow Call forwarding by using dial *72*nnnnnn#, n is the number where the call to forward
exten => _*72*.,1,Answer()
exten => _*72*.,2,Set(DB(CFI/${CALLERID(NUM)})=${EXTEN:4})
exten => _*72*.,n,Playback(vm-saved)
exten => _*72*.,n,SayDigits(${EXTEN:4})
exten => _*72*.,n,Hangup()
Disallow Call forwarding by using dial #73#
exten => #73#,1,Answer()
exten => #73#,n,DBdel(CFI/${CALLERID(num)})
exten => #73#,n,Playback(call-fwd-cancelled)
exten => #73#,n,Hangup()
This will need the macro insert into the appropriate extension, like extension 200
[200]
exten => 200,1,Macro,callforwarding|${EXTEN}
exten => 200,n,Dial(SIP/202,30,mt)
exten => 200,n,Hangup()
insert macro-callforwarding into extensions.conf above the default context and below general section
[macro-callforwarding]
exten => s,1,Set(temp=${DB(CFI/${ARG1})})
exten => s,n,GotoIf(${temp}?cfi:nocfi)
exten => s,n(cfi),Dial(SIP/${temp})
exten => s,n(nocfi),NoOp
Note, if your phone may don't understood the pound key (#) like Grandstream, then apply another char for instance (*) or only numbers
Asterisk Ringtone identification
An Asterisk macro using ringtone identification. In companies, it makes sense to detect the distinction between an internal and an external call.
That provide the following Macro, which in is entered in the configuration file extensions.conf, on FreePBX this is the file extensions_custom.conf.
In this example with SIPAddHeader, this is here for "global" settings, for all external calls now sounds a different tone, the first command generates an additional 0 (Zero) to add automatically a valid prefix for outgoing dial number while call externally.
[macro-exten_did_prefix]
exten => s,1,Set(CALLERID(num)=0${CALLERID(num)}) exten => s,1,Set(CALLERID(num)=${CALLERID(num)}) exten => s,n,SIPAddHeader("Alert-Info: <http://127.0.0.1>\;info=alert-external\;x-line-id=0;")
