similar to: How to retrieve a phone number from call forwarding?

Hi, I'm very new to Asterisk. I tried VoiceMail() application. I'm able to modify extensions.conf and voicemail.conf to send a voicemail audio file to my email. It works great so far. Now, I'm looking into publishing those voicemail files on a web page. According to http://www.voip-info.org/wiki-Asterisk+VoiceMail, it mentions that Asterisk VoiceMail supports "Web interface for

I have many "edit" actions in my project. If the url is like http://localhost/project/3/edit and someone enters http://localhost/project/3333333/edit the edit action will fails because it cannot find a project with id of 3333333. To combat this I am adding to every action like edit: rescue redirect_to :action => ''index'' flash[''errors''] =

i know dovecot can act as IMAP and POP3 proxy ..... but i'm having a hard time configuring it. Actually i'm using a simple dovecot configuration with virtual users stored on MySQL. My dovecot-sql.conf is pretty simple: [root at correio dovecot]# cat dovecot-sql.conf driver = mysql connect = host=localhost dbname=DATABASE user=USERNAME password=PASSWORD default_pass_scheme = PLAIN #

Dear the following is the asterisk's dbase(Mysql5). if the extension =17171000 asterisk run appdata=222222, but I prefer to run appdata=3333333. let me know how I can run the appdata=33333 best Mani mysql> select * from ext; +----+---------+--------+----------+------+---------------------------------------+ | id | context | exten | priority | app | appdata

Hi Daniel, Am 27.03.20 um 09:24 schrieb Administrator: > Hangup is h extension. your macro will never be executed. Solution: > > same = n,Dial(whatever) > same = n,[...]) > same = n,Hangup > > exten  = h,1,1,DumpChan() >  same = n,System(/home/asterisk/bash_test) I don't really understand your code… I think I don't have to edit the first part of the conf file

Hi everybody, we use Asterisk to route all calls to a inbound phone number to a specific outbund mobile phone number, depending on time and date. I'd like to send a notification email to a specific email address, each time we receive a call. For this I used the tip of "dicko" here [1]. I'm a Asterisk newbie. Unfortunately it doesn't work. The System() command is not

Hi, On dovecot 1.2 i had the following configuration on my dovecot-sql.conf file: password_query = select endereco as user, password, '/var/spool/mail/%u' as userdb_home, 'maildir:/var/spool/mail/%u' as userdb_mail, 8 as userdb_uid, 12 as userdb_gid, concat('*:storage=', quota) as userdb_quota_rule, 'Trash:storage=100M' as userdb_quota_rule2 from emails

dovecot 1.0.15 Hello, i try to set quota settings for my users. currentyl i use a mysql table for auth process and now i want to add quotasettings for each individual user. at the moment i have the problem that only global quota is effective and no userquota which is stored in usertable. This is my mysql-usertable: login varchar(255) password varchar(64) home varchar(128) uid int(11) gid

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

Hi all, How are things going ? Is there a way for me to individually identify each BRI channel on the Sirrix quad BRI board. The reason I ask is because our client uses the "Asterisk Flash Operator Panel" to monitor its external lines and transfer calls from the lines to the various SIP phones. The "Flash Operator Panel" requires that we set a static value for each line or

Le 27/03/2020 à 20:19, Kai Herlemann a écrit : > Hi Daniel, > > Am 27.03.20 um 09:24 schrieb Administrator: >> Hangup is h extension. your macro will never be executed. Solution: >> >> same = n,Dial(whatever) >> same = n,[...]) >> same = n,Hangup >> >> exten  = h,1,1,DumpChan() >>  same = n,System(/home/asterisk/bash_test) > I don't

-- SIP/ vaso -e26c answered Zap/14-1 -- Executing DumpChan("SIP/ vaso -e26c", "") in new stack -- Executing DumpChan("SIP/vaso-e26c", "") in new stack Dumping Info For Channel: SIP/vaso-e26c: ============================================================================ ==== Info: Name= SIP/vaso-e26c Type=

On Wed, Aug 8, 2018 at 1:53 PM, Daniel Journo <dan at keshercommunications.com> wrote: > > Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. > > AgentA answers and is able to use that feature code. > > If AgentA performs an attended transfer of a call from a queue to > AgentB, the > > feature code no longer works. > > > > It only

I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File

Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at

Hi all First I want to wish for everone a happy new year... Well... I have run asterisk 1.4.16.1 in a server. I have this IVR, in extensions.conf: [ura] ;exten => s, 1, Wait,1 exten => s, 1, Answer() exten => s, n, Noop() exten => s, n(debug),DumpChan() exten => s, n, Set(LANGUAGE()=pt_BR) exten => s, n, Set(CALLFILENAME=/var/spool/asterisk/monitor/entrada/) exten => s,

Is there a way to list the names of the channel variables that are currently defined on a given channel via AMI? I know of GetVar and SetVar, but GetVar needs the name of the variable to get.

Hi all, with an (attended) transfer i see the following happening: 1) A calls B1 2) B2 calls C 3) B2 transfers call to A 4) A talks to C At step 3, the channel A is connected to channel C and B1 and B2 are hung up. In the h extension for channel B2, the channel is renamed to B2<ZOMBIE> and i see that the channel variables of A have been merged into B2<ZOMBIE>. If there were

Would someone be able to give an example of a working use of setvar from chan_dahdi.conf? I am trying to create a custom variable like I use in sip.conf but I have been completely unsuccessful getting any variable set using setvar to appear for a DAHDI channel. I am running 1.8.11-cert8 and am using the newer format (but I have tried using the older [channels] format). Here is an example:

I am trying to use ImportVar to get some information out of a SIP/ZAP channel. I cannot seem to find an example of the syntax, or what variables I can access. Basically, I would like to output which person is being called. i.e: SIP/25 calls SIP/21. 25 executes a macro, and the result is SIP/21. The info that I want is stored in the channel's "Direct Bridge" variable. I have