Tag: asterisk

  • [Asterisk/FreePBX]Set Inbound DID as Caller-ID for Forwarded Calls

    This is a snippet developed by a colleague of mine. This is for people who’s supplier’s doesn’t allow passing original Caller ID for forwarded calls. What this does is it will show the DID where the calling party called in as the caller ID instead of showing up your default trunk number for the caller ID.

    First add the below code to extensions_override_freepbx.conf, next restart/reload asterisk and you’re good to go.

    [macro-outbound-callerid]
    include => macro-outbound-callerid-custom
    exten => s,1,ExecIf($["${CHANNEL(channeltype)}" = "Local"]?Set(CALLERID(all)=${MASTER_CHANNEL(DEXTEN)}))
    exten => s,n,ExecIf($["${CHANNEL(channeltype)}" = "Local"]?Set(AMPUSER=${MASTER_CHANNEL(DEXTEN)}))
    exten => s,n,ExecIf($["${CALLINGNAMEPRES_SV}" != ""]?Set(CALLERPRES(name-pres)=${CALLINGNAMEPRES_SV}))
    exten => s,n,ExecIf($["${CALLINGNUMPRES_SV}" != ""]?Set(CALLERPRES(num-pres)=${CALLINGNUMPRES_SV}))
    exten => s,n,ExecIf($["${REALCALLERIDNUM:1:2}" = ""]?Set(REALCALLERIDNUM=${CALLERID(number)}))
    exten => s,n(start),GotoIf($[ $["${REALCALLERIDNUM}" = ""] | $["${KEEPCID}" != "TRUE"] | $["${OUTKEEPCID_${ARG1}}" = "on"] ]?normcid)
    exten => s,n,Set(USEROUTCID=${REALCALLERIDNUM})
    exten => s,n,GotoIf($["foo${DB(AMPUSER/${REALCALLERIDNUM}/device)}" = "foo"]?bypass)
    exten => s,n(normcid),Set(USEROUTCID=${DB(AMPUSER/${AMPUSER}/outboundcid)})
    exten => s,n(bypass),Set(EMERGENCYCID=${DB(DEVICE/${REALCALLERIDNUM}/emergency_cid)})
    exten => s,n,Set(TRUNKOUTCID=${OUTCID_${ARG1}})
    exten => s,n,GotoIf($["${EMERGENCYROUTE:1:2}" = "" | "${EMERGENCYCID:1:2}" = ""]?trunkcid)
    exten => s,n,Set(CALLERID(all)=${EMERGENCYCID})
    exten => s,n,Set(CDR(outbound_cnum)=${CALLERID(num)})
    exten => s,n,Set(CDR(outbound_cnam)=${CALLERID(name)})
    exten => s,n(exit),MacroExit()
    exten => s,n(trunkcid),ExecIf($[${LEN(${TRUNKOUTCID})} != 0]?Set(CALLERID(all)=${TRUNKOUTCID}))
    exten => s,n(usercid),ExecIf($[${LEN(${USEROUTCID})} != 0]?Set(CALLERID(all)=${USEROUTCID}))
    exten => s,n,ExecIf($[${LEN(${TRUNKCIDOVERRIDE})} != 0 | ${LEN(${FORCEDOUTCID_${ARG1}})} != 0]?Set(CALLERID(all)=${IF($[${LEN(${FORCEDOUTCID_${ARG1}})}=0]?${TRUNKCIDOVERRIDE}:${FORCEDOUTCID_${ARG1}})}))
    exten => s,n(hidecid),ExecIf($["${CALLERID(name)}"="hidden"]?Set(CALLERPRES(name-pres)=prohib_passed_screen))
    exten => s,n,ExecIf($["${CALLERID(name)}"="hidden"]?Set(CALLERPRES(num-pres)=prohib_passed_screen))
    exten => s,n,Set(CDR(outbound_cnum)=${CALLERID(num)})
    exten => s,n,Set(CDR(outbound_cnam)=${CALLERID(name)})
    

     

  • Connecting two Asterisk/FreePBX using SIP Trunks

    This was a project that I’ve been working on and off for some time and always ended up with failure. There are many documentations available on the net however the one that worked for me is using IP trunks and here’s how it is done.

    Lets take an example so it’s easy to identify. Let’s assume the the two PBXes are at Sydney & Melbourne.

    In Sydney PABX, create a new trunk, name it to-Melbourne. For the peer details set it up as follows:

    Trunk Name: melbPABX

    type=friend
    context=from-trunk
    host=Melbourne-PABX-IP-OR-FQDN
    username=sydPABX
    password=superstrongsecretpassword
    qualify=yes

    In the Melbourne PABX, create a new trunk with a name to-Sydney. For the peer details set it up as follows:

    Trunk Name: sydPABX

    type=friend
    context=from-trunk
    host=Sydney-PABX-IP-OR-FQDN
    username=melbPABX
    password=superstrongsecretpassword
    qualify=yes

    Points to remember:

    • Password common for both trunks.
    • Trunk names used under Peer Details acts as the usernames.
    • Extension numbers should differ on each PABX otherwise it would not be possible to route calls correctly.
  • [Asterisk/FreePBX] Blocking incoming calls to an extension

    You may have a scenario where you’ve a particular extension assigned to a special service and you want only a select group of people/extensions to be able to make calls to this extension.

    This can be easily achieved with the dial plan below. There’s just one file to edit : extensions_custom.conf

    [kick]
    exten => _X.,1,Playback(invalid)
    exten => _X.,n,Hangup

    [from-internal-custom]
    exten => 7572/7520,1,Answer()
    exten => 7572/7507,1,Answer()
    exten => 7572/7584,1,Answer()
    exten => 7572/4914,1,Answer()
    exten => 7572/7500,1,Answer()
    exten => 7572/_XXXX,1,Goto(kick,${EXTEN},1)
    exten => 7572/_X.,1,Answer()

    Let’s analyse what’s happening here.

    The first section [kick] tells Asterisk to play a message saying the dialed destination is invalid and then to hang up.

    The next section [from-internal-custom] defines what extension can connect/dial to this particular extension (in this example ext 7572 is the one needing incoming restrictions).
    The part before the “/” defines the destination extension and the part after the “/” defines the source extensions.
    So in this scenario extensions 7500, 7507, 7520, 7584 and 4914 will be able to call 7572. All other 4 digit extensions will be sent to [kick] which will terminate their calls.

    The last part will allow any other call to be picked up by the extension (for example, an external call).

     

    Original idea from : source

  • Asterisk/FreePBX Time Conditions with Multiple Time Zones / Daylight Saving

    This solution is for older Asterisk/FreePBX installations (v2.11 and below) where the Time Condition module doesn’t have the Time Zone dropdown.

    TZThis is based on a bash script which alters MySQL DB values to represent the change in DST.

    This script will help you to automate DST changes easily and and have peace of mind about not having to manually change every year.

    Scenario:

    PABX is for an Australian company which has offices in multiple states. Main office is in NSW which observes daylight saving whereas a branch office is in QLD which does not observe daylight saving. Server time is set for NSW timezone. (more…)

  • Asterisk Phone Inventory / Useragent List

    Create batch file with below and run the script. It will list down the Useragent, Contact & Username.

    asterisk -rx “sip show peers” | cut -f1 -d/ | grep -P ‘\d\d\d\d’ | grep -vP ‘(UNKNOWN|Unmonitored)’ |
    while read PEER
    do
    asterisk -rx “sip show peer ${PEER}” |
    grep -P “(Useragent|Contact|Username)”
    echo “====”
    done

  • Asterisk/FreePBX – How to restrict an extension to call certain extension only

    There may come a time that you want a public access phone that can only dial out a certain set of extensions. To accomplish this, a custom context needs to be created and applied to that extension.

    Here’s the scenario.
    Extension 312 is the public access phone. This should only be able to dial x530 & x555. All other calls needs to be terminated.

    Define the context in extensions_additional.conf

    [from-intercom]
    exten => 530,1,goto(from-internal,${EXTEN},1)
    exten => 555,1,goto(from-internal,${EXTEN},1)
    exten => _.,1,hangup()

    If FreePBX, go inside the extension and the context from from-internal->from-intercom. Apply the settings and you’re good to go.

  • Asterisk/FreePBX Blind Transfer Return Call to Origin

    Another customer of ours requested this feature. By default, with blind transfers if the internal destination doesn’t answer the call, it goes to vociemail or terminates if vociemail is not configured. This customer wanted to return the call to the originator so that they can direct the call elsewhere.

    You need to modify two places. They’re listed below.

    globals_custom.conf:
    TRANSFER_CONTEXT = custom-test_transfer

    extensions_custom.conf:
    [custom-test_transfer]
    exten => _X.,1,NOOP(entering custom-test_transfer)
    exten => _X.,n,set(timeoutd=25) ; set timeout in seconds
    exten => _X.,n,set(extLeng=${LEN(${EXTEN})})
    exten => _X.,n,noop(the extenlength is ${extLeng})
    exten => _x.,n,dial(local/${EXTEN}@from-internal,${timeoutd})
    exten => _X.,n,Gotoif($[“$\{DIALSTATUS}”=”ANSWER”]?hangup,callback)
    exten => _X.,n,Set(CALLERID(name)=RB:${CALLERID(name)})
    exten => _X.,n(callback),dial(local/${BLINDTRANSFER:4:${extLeng}}@from-internal)
    exten => _X.,n(hangup),hangup()
    exten => _*X.,1,dial(local/${EXTEN}@from-internal,15)
    exten => _*X.,n,hangup()

    Thanks to Alexander for fixing my earlier code. I am yet to test this on a live site 🙂

    extensions_custom.conf:
    [custom-test_transfer]
    exten => _X.,1,NoOp(Entering custom-test_transfer)
    exten => _X.,n,Set(timeoutd=25) ; set timeout in seconds
    exten => _X.,n,Set(extLeng=${LEN(${EXTEN})})
    exten => _X.,n,NoOp(The extenlength is ${extLeng})
    exten => _X.,n,Dial(Local/${EXTEN}@from-internal,${timeoutd})
    exten => _X.,n,Set(CALLERID(name)=RB:${CALLERID(name)})
    exten => _X.,n,Dial(Local/${BLINDTRANSFER:4:${extLeng}}@from-internal)
    exten => _X.,n,Hangup()

    Once setup, reload asterisk and test this out.

    There are several other scripts out there, but this is the one that worked correctly for me.

  • Common Issues with MOH – Asterisk/FreePBX

    When working with MOH (Music on Hold) files in Asterisk/FreePBX I’ve come across a few issues that can occur regularly mainly due to human error. Best way to identify the issue is to check CLI entries when putting on hold as they give distinct clues as to what the problem is.

    First one is that you get an error message stating cannot open or find the file. This is a permission issue. Make sure the MOH files are awned by asterisk user and has correct file permissions. This has happened to me more than one occasion when I’ve uploaded the file via SSH instead of the GUI or after doing a volume change using sox.

    Second one you’d get is something on the line of unknown stream or unsupported format. Which simply means that asterisk doesn’t have the required tools to play the file.
    With my customers, what I normally do is convert the files to wav files, on 8000 Hz, 16 bit, Mono settings using “Nero Wave Editor” and upload the files. This resolves many problems with different formats as well as reduces CPU load.

    Final issue can be rectified using one of my earlier post.  : http://www.asanka.me/?p=19

  • A fix for new Asterisk MOH file not found issue

    The new Asterisk PBX system for some odd reason have an issue with MOH not working by default. Digging deeper I found out that there’s a mismatch in directory names where asterisk looks for these files.

    It looks like the GUI portion of asterisk uploads files to /var/lib/asterisk/mohmp3 but config files set to look in /var/lib/asterisk/moh. the easiest fix is as below.

    mv /var/lib/asterisk/mohmp3/*.* /var/lib/asterisk/moh/
    rm -f /var/lib/asterisk/mohmp3
    ln -s /var/lib/asterisk/moh /var/lib/asterisk/mohmp3

    This will move all content on mohmp3 directory to moh, delete mohmp3 folder and then create a symlink named mohmp3 targeting moh. Now you don’t have to mess about with editing any config files.

  • Setting up a Cisco router for auto-provisioning of Cisco handsets

    Here is a sample config to setup auto-provisioning for Cisco handsets. This only covers the part of distributing the config files to the handsets and not the config file creation part.

    This assumes that VLAN100 is the VOICE VLAN and PCs are connecting via the PC port at the back of the phones.

    !
    ip dhcp pool VOICE
    network 192.168.100.0 255.255.255.0
    default-router 192.168.100.1
    dns-server 192.168.100.1
    option 66 ascii "http://auto.provisioning.url/"
    lease infinite
    !
    interface Vlan100
    description VoiceVLAN
    ip address 192.168.100.1 255.255.255.0
    ip nat inside
    ip virtual-reassembly in
    !
    interface FastEthernet3
    description ToVoiceSwitch
    switchport mode trunk
    switchport voice vlan 100
    !