similar to: voicemail context. macro, and directory

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

I use the following macro for my extensions. It only works with Zap channels and assumes that any Caller*ID number that is 4 digits is an internal call and all other calls are external calls. Use like this: exten => 1234,1,Macro(std-exten,Zap/4,20) [macro-std-exten] ; ; Caller*ID is 4 digits (internal call) ; exten => s/_XXXX,1,Dial(${ARG1}r2,${ARG2}) exten =>

Has anyone had any problems with the voicemail app in the last weeks cvs? I was using a checkout from 10-31 and everything worked fine, but the last few days I get a hangup right after it calls the voicemail app. No sound, just an immediate disconnect. Nothing has changed in my extensions.conf, etc. Here is what I have in extensions.conf: [macro-oneline] exten => s,1,Dial(${ARG1},20,tr)

Hi List; I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten

Please see my extensions below. I will try to type you through this. In default, an extension 5149053538 is matched on. This fires a macro that determines what time is it, and resets a variable which is then used to call another macro to place a call. if the call is answered, and the far end hangs up, the dial macro exits, then the routing macro exits, and you are back to default. at this point

Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050812/1bff12e4/attachment.htm

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} -

Hi all, I'm sure this has been done before but I just can't figure it out. On my * box I have a simple IVR: [tolc_menu] ; Welcome and information to callers exten => s,1,Answer() exten => s,n,Wait(2) exten => s,n,Background(welcome-to-tolc) ; Say Hello exten => s,n,Wait(1) exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter extension number if known, or

I am configuring call forwarding in our * setup, but I am having trouble triggering the correct voicemail call. When I have an extension, e.g. 201, forwarded to another, e.g. 202, my macro will call: Dial(Local/${fwNum}@internal/n,30) and once into the internal context, the macro is called again but with MACRO_EXTEN set to 202 rather than 201. So the context flow of control will "do

I am using the sample (slightly modified) macro for dialing phones. My extensions are in the 2000 range. Each extension has it's own external DID. Is there any way to have the macro look up the DID to be used for the extension and set the DID as the callerid? Maybe from a flat file somewhere? Or is there a better way to do this??? I know I can use callerid in sip.conf, but I only want the

I'm having a few issues with the MySQL realtime configuration in CVS-HEAD. I tested it initially with realtime extensions (realtime_ext => mysql,asterisk,extensions) and a realtime switch in extensions.conf and that works fine, So I though I'd go back and test a static configuration mapping. I used the table structure from the asterisk guru postgres howto to create something

Hi All, I was made aware today that some of my calls coming in are not going to voicemail... Below are some logs, and the macro that should run on the incoming_pstn context for that extension. I can see that theres a non-zero exit before it gets to voicemail, but I've no idea why. In this case theres 2 SIP clients to sim-call. On other occasions it works fine. In the CDR logs, I can see

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

Hi, I have following one-line macro extension: ------------------------ [macro-oneline] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Device(s) to ring ; #exten => s,1,AGI(misterhouse.agi,"CallerID") exten => s,1,NoOp exten => s,2,DBget(temp=CFIM/${MACRO_EXTEN}) ; Get CFIM key, if not existing, goto 103 exten => s,3,Dial(Local/${temp}@default/n) ;

I have Spandsp working fine. Asterisk sees a fax on the zap port and redirects the call to the fax-in area. This works if I have a simple dialing rules that goes answers first and waits 10 secs then goes to the next item. If it hears a fax it goes to the right place. Here is a sample that works. [incoming] exten => 2019,1,Goto(test,s,1) [test] exten => s,1,answer exten => s,2,wait(5)

Dear All Can you please do me favor and let me know what is my problem with my Asterisk VoiceMail configuration as it doesn't work correctly in my case ? Please find below that part of my extensions.conf that I intend to make use of voice mail for No Answer reply : " [line-incoming] exten => _XXXXXXX,1,macro(dialuser,SIP/${EXTEN},${EXTEN}) [macro-dialuser] exten =>

I think you will find the functionality you are looking for is in * already. Here is an excerpt from the sample extensions.conf file that is included with the source: exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten

Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute

I am using realtime mysql for extensions, sip, and voicemail. Outbound call routing does not really perform well in realtime extensions due to the high number of rows in the database (300k), so I can not use it. It appears with my limited knowledge that the query method is not robust enough for large databases. Given the fact that I already have realtime and mysql configured, what are my options