search for: exitcontext

Displaying 10 results from an estimated 10 matches for "exitcontext".

2006 Oct 27
0
Voicemail 'exitcontext'
This seems to be a bug. I can get exitcontext to work on a per mailbox basis in voicemail.conf. However, for realtime mailboxes, I added a new column called 'exitcontext' to my table, and the thing simply doesn't work. I can see asterisk selecting * from the table, but pressing 0 while in voicemail has no effect. If I press 0 while...
2006 Oct 10
10
Voicemail Press '0'
Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says "the context for the voicemail box that we're looking for in the dialplan for the jump to the
2009 Apr 14
2
Exit Dial Application
...-BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I' try to implement an automatic callback mechanism, just for local SIP calls.. Callback on busy and on no answer. If the other party doen't answer, it should be possible to press 5 to place an callback. Here is my dial: exten => _X.,1,Set(EXITCONTEXT=callback) exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT) And here the script for callback. [callback] exten => 5,1,agi(str_concat.sh) exten => 5,n,Hangup If I call someone and press 5, nothing happens. What could be a problem? DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmi...
2004 Jul 17
1
voicemail broadcast feature
Using CVS from 7/12/04 and trying to get the voicemail broadcast feature to work. Voicemail.conf has [mycontext] 3722 => 1234,BroadCast Test,,,cc=*@mycontext . then many other voicemail boxes. ----- whenever I leave voicemail at box 3722, only box 3722 gets the voicemail. It is not expanding it to other voicemail boxes in the [mycontext] context. Even if I replace the cc= line with
2008 Nov 18
1
Incoming Transfer
I have incoming analog and SIP DIDs that all ring multiple sip extensions with a Dial command as the first exten. I am curious to know if it's possible for the incoming caller to transfer out of the Dial command while in progress and dial a single extension? Thanks! jlc
2014 Aug 17
1
Overriding global voicemail options on a per-mailbox basis
...ilbox settings, unless listed otherwise". Now the book Asterisk, The Definitive Guide, 3rd ed. lists 39 voicemail advanced options starting on page 163. Yet on page 167, it states that "There are nine valid options: attach, serveremail, tz, saycid, review, operator, callback, dialout, and exitcontext." And the book Asterisk, The Definitive Guide, 4th ed. lists the same 39 voicemail advanced options starting on page 192 yet on page 197 it lists only 26 advanced options. So my question is this: Does anyone know which options can and can't be overridden on a per-mailbox basis? Are t...
2011 Jun 15
2
change destination on digit
Is there an easy way to setup diaplan so when someone pushes a digit such as * during a call, they will be transferred to another destination. For example, a caller is hearing ringing while calling a UA, but instead of waiting for the UA to pick up, they can push * and go directly to that UA's voicemail. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2015 Jan 20
1
Mailbox password change problem on realtime engine
...DEFAULT NULL, `saydurationm` int(3) DEFAULT NULL, `forcename` char(3) COLLATE utf8_unicode_ci DEFAULT NULL, `forcegreetings` char(3) COLLATE utf8_unicode_ci DEFAULT NULL, `callback` char(80) COLLATE utf8_unicode_ci DEFAULT NULL, `dialout` char(80) COLLATE utf8_unicode_ci DEFAULT NULL, `exitcontext` char(80) COLLATE utf8_unicode_ci DEFAULT NULL, `maxmsg` int(5) DEFAULT NULL, `volgain` decimal(5,2) DEFAULT NULL, `imapuser` varchar(80) COLLATE utf8_unicode_ci DEFAULT NULL, `imappassword` varchar(80) COLLATE utf8_unicode_ci DEFAULT NULL, `stamp` timestamp NOT NULL DEFAULT CURRENT_TIMES...
2006 Nov 22
1
DTMF detection during Call
Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian
2018 Dec 04
2
asterisk is not seeing my queues in database
I enabled the logs on the mysql database and ran : realtime load queues name cou0002-test in the mysql log I can see that the proper select statement is being executed: 2018-12-04T16:29:27.253094Z 229 Query SET SESSION TRANSACTION ISOLATION LEVEL READ COMMITTED 2018-12-04T16:29:27.254384Z 229 Prepare SELECT * FROM queues WHERE name = ? 2018-12-04T16:29:27.254902Z 229