search for: aa_1

...;, "0?from-pstn-reghours-nofax|s|1:2") in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer("Zap/2-1", "") in new stack -- Executing Wait("Zap/2-1", "1") in new stack -- Executing SetVar("Zap/2-1", "intype=aa_1") in new stack -- Executing Cut("Zap/2-1", "intype=intype|-|1") in new stack -- Executing GotoIf("Zap/2-1", "0?7:9") in new stack -- Goto (from-pstn-reghours,s,9) -- Executing GotoIf("Zap/2-1", "0?10:12") in new stack...

...e from A@H and some lines which shows the disconnect. Should DISA be loaded as a module in modules.conf? When I do a 'show applications' i see that DISA is there. Help! -------------------------------------- ;Asterisk CLI as I placed a call from cell into the system. Playing 'custom/aa_1' (language 'en') ==CDR updated on Zap/3-1 Executing Goto("Zap/3-1", "disa_custom|s|1") in new stack Goto (disa_custom,s,1) Hungup 'Zap/3-1' ------------------------------------- ; MY EXTENSIONS.CONF [aa_1] exten => 8888,1,Goto(disa_custom,s,1) [disa-cus...

...9; (language 'en') -- Executing Hangup("Zap/3-1", "") in new stack == Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'Zap/3-1' in m == Spawn extension (macro-exten-vm, s, 8) exited non-zero on 'Zap/3-1' in macr == Spawn extension (aa_1, 240, 1) exited non-zero on 'Zap/3-1' -- Executing Hangup("Zap/3-1", "") in new stack == Spawn extension (aa_1, h, 1) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' -- Hungup 'Zap/1-1' --------------- Any pointers/suggestions wil...

...new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP. I can make out going calls. The problem I'm having now is the digital receptionist greeting (aa_1). If I set it to automatically forward to an extension it works. But, if I have it play a message (press 200 for Joe, etc), you can't here the message at all. I can dial the extension number and * will accept and forward me to that extension. I can see on the CLI that it is suppose to b...

I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

...ns,${EXTEN}) exten => 1234,1,Dial(Sip/phone1,20) ;Aastra 480iCT [closed] exten => s,1,Goto(ivr,s,1) [ivr] exten => s,1,Answer exten => s,n,Set(LOOPCOUNT=0) exten => s,n(begin),Set(TIMEOUT(digit)=3) exten => s,n,Set(TIMEOUT(response)=10) exten => s,n,Background(aa_1) exten => s,n,WaitExten(10) exten => s,n,Goto(loop,1) exten => #,1,Directory(default|internal) exten => i,1,Playback(invalid) exten => i,n,Goto(loop,1) exten => t,1,Goto(loop,1) exten => loop,1,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1]) exten => loop,n,GotoIf($[${LO...

This is probably a stupid question.. How do I login to voicemail from the PSTN? I can dial *98 from within the system, but when dialing from the PSTN I have it set up to ring a dial group, then to an extensions vmail. During the extensions vmail prompts, I dial *98 and it sends me to the directory. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Nitesh, check out my config that I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, February 23, 2005 4:12 PM To:

...have it narrowed down to the following. When the call is parked from the auto attendant the ring back is set to <snip from pastebin.ca/2452 > ******* Works Correctly and send call back to the extension that parked the call ****** 069 == Parked Zap/3-1 on 701. Will timeout back to aa_1,2204,1 in 45 seconds 070 -- Playing 'digits/7' (language 'en') 071 -- Playing 'digits/0' (language 'en') 072 -- Playing 'digits/1' (language 'en') 073 -- Stopped music on hold on Zap/3-1 074 -- Started music on hold, class 'def...

...ge 'en') -- Playing 'digits/7' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'digits/0' (language 'en') -- Executing Hangup("CAPI[contr1/7104370]/3", "") in new stack == Spawn extension (aa_1, h, 1) exited non-zero on 'CAPI[contr1/7104370]/3' -- CAPI Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI = 0x201

...dged=yes echocancel=128 echotraining=800 rxgain=5 ; 0 txgain=-4.5 ; 0 busydetect=no pridialplan=local internationalprefix=0011 nationalprefix=0 usecallerid=yes hidecallerid=no callprogress=no group=0 channel => 1-15,17 /etc/asterisk/extensions: [from-onramp] ;exten => s,1,Playback(custom/aa_1) exten => s,1,Dial(SIP/116) exten => h,1,Hangup and here's some log info: asterisk*CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 asterisk*CLI> asterisk*CLI> asterisk*CLI> &&&&&&&&&&&&&&&&&a...

...WARNING[12167]: MFC/R2 UniCall/2 0101? ->????? [1/????? 20/Group B?????? /Accepted Paid] Sep? 7 10:29:59 WARNING[12167]: Unicall/2 event Answered Sep? 7 10:29:59 WARNING[12167]: MFC/R2 UniCall/2 Channel echo cancel Sep? 7 10:33:13 WARNING[12167]: Timeout, but no rule 't' in context 'aa_1' Sep? 7 10:33:13 WARNING[12167]: MFC/R2 UniCall/2 Channel gains Sep? 7 10:33:13 WARNING[12167]: MFC/R2 UniCall/2 Channel switching Sep? 7 10:33:13 WARNING[12167]: MFC/R2 UniCall/2 Call control(6) Sep? 7 10:33:13 WARNING[12167]: MFC/R2 UniCall/2 Drop call(cause=Normal Clearing [16]) Sep? 7 10:33...

...e channels ; default usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes faxdetect=both busydetect=yes busycount=4 busypattern=500,500 callprogress=no ; define channels context=from-pstn ; signalling=fxs_ls ; channel=>1-3 ; context=aa_1 ; signalling=fxs_ls ; channel=>4 ; -------------------------------------------------------------------------------------------------------------------------------------------------- Output from asterisk log : Mar 25 13:45:16 VERBOSE[17282] logger.c: -- Starting simple switch on 'Zap/1-...