search for: voicemenu

...e files to /var/lib/asterisk/sounds/record .. when I go to the Voice Menu Prompts selection down the left side of the Asterisk-GUI, I see my four files with the options to record again, play and delete. If I then go to the Voice Menu option to configure a Voice Menu, and click on the Create New VoiceMenu, I get nothing .. nada .. kaput .. how the heck do I create a menu for an incoming call on a Trunk? When I started this project I knew it would be fun .. I would learn a lot! The problem is that one of our administrators is absolutely a newbi to Linux, so I have to make this work with the...

...e files to /var/lib/asterisk/sounds/record .. when I go to the Voice Menu Prompts selection down the left side of the Asterisk-GUI, I see my four files with the options to record again, play and delete. If I then go to the Voice Menu option to configure a Voice Menu, and click on the Create New VoiceMenu, I get nothing .. nada .. kaput .. how the heck do I create a menu for an incoming call on a Trunk? When I started this project I knew it would be fun .. I would learn a lot! The problem is that one of our administrators is absolutely a newbi to Linux, so I have to make this work with the...

I currently have in extensions.conf: exten => 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)}) exten => 106,n,Monitor(wav,${CALLFILENAME},m) exten => 106,hint,SIP/106 exten => 106,Macro(stdexten,106,${HINT}) When I called x106 this was logged: -- Executing [106 at voicemenu-custom-4:1] Set("DAHDI/7-1", "CALLFILENAME=_xxxxxxx") in new stack -- Executing [106 at voicemenu-custom-4:2] Monitor("DAHDI/7-1", "wav|_xxx-xxx- xxxx|m") in new stack == Auto fallthrough, channel 'DAHDI/7-1' status is 'UNKNOWN' --...

...ack [Oct 20 18:37:17] VERBOSE[10611] logger.c: -- Executing [s at DID_trunk_1:2] ExecIf("DAHDI/1-1", "0|Set|CALLERID(all)=unknown <0000000>") in new stack [Oct 20 18:37:17] VERBOSE[10611] logger.c: -- Executing [s at DID_trunk_1:3] Goto("DAHDI/1-1", "voicemenu-custom-3|s|1") in new stack [Oct 20 18:37:17] VERBOSE[10611] logger.c: -- Goto (voicemenu-custom-3,s,1) [Oct 20 18:37:17] VERBOSE[10611] logger.c: -- Executing [s at voicemenu-custom-3:2] Wait("DAHDI/1-1", "2") in new stack [Oct 20 18:37:17] DEBUG[10611] chan_dahdi....

...ar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)}) exten => 106,2,Monitor(wav,${CALLFILENAME},m) exten => 106,3,hint,SIP/106 exten => 106,4,Macro(stdexten,106,${HINT}) I received this warning: WARNING[31463]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for extension (voicemenu-custom-4, 106, 1) I'm running Asterisk/1.4.22. Does anyone have any idea what I need to do to either make SetVar work or replace it with something else? Thanks you, Gary

...autoanswer),Set(_ALERT_INFO="RA") exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) exten => s,n,NoOp() exten => s,n,Dial(${ARG1}||) exten => s,n(fail),Hangup [page] exten => _X.,1,Macro(page,SIP/${EXTEN}) [default] exten => 6050,1,VoiceMailMain exten = 7000,1,Goto(voicemenu-custom-1|s|1) exten => 6000,1,MeetMe(${EXTEN}|MI) exten = 3010,1,Goto(ringroups-custom-1|s|1) exten = 3020,1,Goto(ringroups-custom-2|s|1) exten = 6005,1,Queue(${EXTEN}) [voicemenu-custom-1] include = default comment = Welcome alias_exten = 7000 exten = s,1,Answer exten = s,2,Wait(1) exten = s,3...

...102,busy() ;This is where incoming calls go to if I'm awake. [DID_trunk_2_timeinterval_Awake] exten = _NXXNXXXXXX,1,Set(GROUP()=SIPGROUP) exten = _NXXNXXXXXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)}) exten = _NXXNXXXXXX,n,Set(CALLERID(num)=1${CALLERID(num)}) exten = _NXXNXXXXXX,n,Goto(voicemenu-custom-1|s|1) Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081010/467c3002/attachment.htm

...nterface seems quite user friendly. Although the phone is pricey. HOWEVER the only problem I am having when doing interop testing is voice mail pickup / indication doesn't appear to work. Config: Message Waiting IP address or DNS name: <asterisk ip> Voicemail number: 8500 *8500 = voicemenu extension. I'm guessing that the phone just doesn't support the voice mail indication protocol that Asterisk uses. Can anyone confirm this for me or better yet tell me where I am going wrong. Thanks for any help as I have limited time to chose a SIP hardphone to use in a small office...

Hello, is anyone on the list using a normal cell/mobile phone which is able to act as a SIP client over WLAN? Or has anyone heard of a SIP client for cell/mobile phones running windows mobile 6.x? The phone should use SIP, when the asterisk server is reachable and should automatically switch to a German telco if it is not reachable. Thanks for any hints, Stefan --

...n => s,3,Dial(Zap/1,40,tr) exten => s,4,Playback(vm-isunavail) ;exten => s,5,Dial(SIP/clienta,10,tr) exten => s,5,Background(vm-enter-num-to-call) exten => s,6,NoOp,${CALLERID} ;exten => s,8,Dial(Zap/1,20,tr) ;exten =>s,9,Hangup include =>sip ;This context is used to record voicemenu and its recording properly. ; used to record prompts [playback] exten =>20,1,Playback(vm-sorry) exten =>20,2,Hangup [record] exten => 205,1,Wait(2) exten => 205,2,Record(/tmp/asterisk-recording1:gsm) exten => 205,3,Wait(2) exten => 205,4,Playback(/tmp/asterisk-recording1) e...

...his sample extension.conf uses's the ast DB to store a users current extension, in a db family of CallFWD and the unique Key is based on the current channel the user is assigned. In the globals var section each key is hardcoded EXT1, EXT2 this is used in the [incoming] context to associate each voicemenu option with a user/extension. A macro call is made for each extension which does a simple lookup in the ast DB based on the family/key of CallFWD/${EXT?}, this returns the extension the user is currently forwarded to. The dial command just use's what ever ext the user has assigned. The current...

Is there any way to have users be able to turn on or off call forwarding at the asterisk server, so they can configure their own forwarding number and enable/disable it? Hopefully, with the added benefit that it will remain on between server reloads and restarts? I have written a hack -- a AGI script to do various checking, and if the destination is "ok" set a database variable

Can multiple FWD accounts be registered? I have the following output in my sip.conf file: register=74928:xxx@fwd.pulver.com/74928 register=75160:xxx@fwd.pulver.com/75160 register=74573:xxx@fwd.pulver.com/74573 [fwd-74928] type=friend secret=xxx username=74928 host=fwd.pulver.com [fwd-75160] type=friend secret=xxx username=75160 host=fwd.pulver.com [fwd-74573] type=friend secret=xxx

...[dial_go] ; exten => s,1,Answer exten => s,n,Wait(2) exten => s,n,Goto(${Q_NAME},1) ; If none of above happen, send to queue exten => _s-.,1,Goto(${Q_NAME},1) ; exten => fax,1,QueueLog(${Q_NAME}|${UNIQUEID}|NONE|FAXDETECTED|${ANI}) exten => fax,n,Hangup() ; exten => _512,1,Goto(voicemenu-custom-1,s,1) ; exten => _5[01]X,1,Queue(${EXTEN}||||${Q_TIMEOUT}) exten => _5[01]X,n,Hangup() ; chan_dahdi.conf faxdetect = incoming I have successfully detected and Answer Machine with AMD application, but as I experience some dificulties to detect fax I have removed on AMD structure for...