search for: vmessage

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

I'm working on setting up a macro that will allow users to call their own DID number, and when they hear their voicemail greeting hit the * key and be prompted for their password to check vmail. For some reason though the background command isn't working as I'd expect it to: [macro-vmessage] exten => s,1,Answer exten => s,2,Background(/var/spool/asterisk/voicemail/sixthree/${ARG1}/unavail) exten => 1,*,Macro(checkmessage,${ARG1}) exten => 2,*,Hangup exten => s,3,VoiceMail2(s${ARG1}) exten => s,4,Hangup [macro-checkmessage] exten => s,1,VoiceMailMain2(${ARG1}) ex...

...apateller(answer|nocallerid) exten => s,3,NoOp exten => s,4,PrivacyManager exten => s,5,LookupCIDName exten => s,6,LookupBlacklist exten => s,7,Background(pls-wait-connect-call) ;exten => s,8,Dial(Zap/4,15,Ttr) exten => s,9,Answer exten => s,10,Wait(1) exten => s,11,Macro(vmessage,2101) exten => s,105,Macro(vmessage,6000) exten => s,107,Macro(vmessage,6000) Has anyone else had this issue and can offer a solution to my problem? I am using a current download of asterisk as of today 6/4/2004

...; Master context for local, toll-free, and iaxtel calls only ; ignorepat => 9 include => default include => parkedcalls include => trunklocal include => iaxtel700 include => trunktollfree include => iaxprovider ;This will create a macro we will use in the dialling plan [macro-vmessage] exten => s,1,VoiceMail2(u${ARG1}) exten => s,2,Playback(groovy) exten => s,3,Playback(goodbye) exten => s,4,Hangup [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device(s) to ring ; exten =>...

...; Master context for local, toll-free, and iaxtel calls only ; ignorepat => 9 include => default include => parkedcalls include => trunklocal include => iaxtel700 include => trunktollfree include => iaxprovider ;This will create a macro we will use in the dialling plan [macro-vmessage] exten => s,1,VoiceMail2(u${ARG1}) exten => s,2,Playback(groovy) exten => s,3,Playback(goodbye) exten => s,4,Hangup [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device(s) to ring ; exten =>...

...9;m able to receive but not make calls with zaptel using an X101P connecting to Asterisk with an Xlite client. My client has context = flat in sip.conf and extensions number 8919 In extensions.conf I've got: [home] ; Line 1 ; exten => 8919,1,Dial(${PHONES1},20,Ttm) exten => 8919,2,Macro(vmessage,${PHONES1VM}) exten => 8919,3,Hangup [outgoing] exten => _9.,1,Dial(Zap/1/$EXTEN:1) [flat] include => home include => outgoing zapata.conf contains the following - I have 2 x101p cards installed [channels] language=en group=1 context=from-analog signalling=fxs_ks usecallerid=no ech...

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using

...onf [general] static=yes writeprotect=no [globals] TRUNK=Gradwell TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) PHONES1=SIP/2201 [flat] include => home include => outgoing [home] exten => 2201,1,Dial(${PHONES1},20,Ttm) exten => 2201,2,Macro(vmessage,${PHONES1VM}) exten => 2201,3,Hangup [outgoing] ignorepat => 9 ignorepat => 8 exten => _9.,1,Dial(SIP/${EXTEN:1}@Talklite) exten => _8.,1,Dial(SIP/${EXTEN:1}@Gradwell) =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- linux:/etc/asterisk # tethereal -R "sip&quo...

...RG4},s,1) ; Callee chose to send this call to a telemarketer torture script. exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain [macro-vmessage] ;This will create a macro we will use in the dialling plan exten => s,1,VoiceMail2(u${ARG1}) exten => s,2,Playback(groovy) exten => s,3,Playback(goodbye) exten => s,4,Hangup [demo] ; ; We start with what to do when a call first comes in. ; exten => s,1,Wait,1 ; Wait a se...

Mateo, Dialing the extension to your softphone is the same as any hardware extension. Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty exten => 1000,2,Macro(vmessage,1000) exten => 1000,3,Hangup Change [mateo01] to [1000] in your sip and you will be saying that ext. 1000 is registered with the specifics you are using. Update the settings in your softphone to register the name and number as 1000 Then any attempt to dial 1000 should come to that phone. Wil...

I am attempting to write in incoming context for calls. 1. If the caller id is given and it is not black listed it will Playback a greeting and then right the phone or go to voicemail under busy or unavailable conditions 2. If no caller id is given, then Privacy Manager will ask for the number. I am testing 6145551212 to see if the black list will work 3. If a caller id is given, and it is

Hi My head hurts... Can anyone help out here, my remote IAX can see my local IAX and visa versa, conversation starts, I can dial my remote (POTS) landline number, remote end answers, trys to route to local iax2, I see it start the conversation here, the extension (SIP) rings once and then it dies... Both ends are defined with accept IPADDRESS to keep it in the family and simple.. Debug info

The following series of patches fill in most of the gaps in the OCaml bindings to libxl, to make them useful for clients such as xapi/xenopsd (from XCP). There are a number of bugfixes to the existing bindings as well. I have an experimental version of xenopsd that successfully uses the new bindings. An earlier version of the first half of the series was submitted to the last by Ian Campbell on

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten