similar to: command like break ore exit in the dialpan

Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR

Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] ??? exten => 99,1,ANSWER() ??? exten => 99,2,PLAYBACK(tt-monkeys) ??? exten => 99,3,HANGUP() The phone

Hi, i am having problem on playing asterisk sound file on my new installed asterisk.. i have the following extension , if i call from any SIP / IAX phone playback or voicemail doesnt play anything .... but when i dial 102, I hear the MP3 music .. exten => 99,1,Answer() exten => 99,2,Playback(prepaid-welcome) exten => 99,3,Hangup() exten => 101,1,VoiceMailMain() exten =>

Hi i just want to let you know that is available a new release of ccmanager. I've added the possibility to import queue_log information in a mysql database and to generate reports using this information. The software is in a beta state and provides this functionality: - users management - call generation (making a GET or POST request on a certain URL) - queue management (LOGIN / LOGOUT /

what is the support in asterisk for ssl voip protocols? I am looking for a solutions to grant the possibility to some users to use an asterisk server as a proxy voice, for talking each them in a safe and secure mode on internet. Is it possible? thanks

Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks

Hi i am upgrading an asterisk server from 1.2.4 to 1.4. i've installed libpri 1.4 i've installed zaptel 1.4 I've installed the new version of misdn with the script of beronet. i use this configure script: ./configure --with-postgres=/usr/local/pgsql then: make menuselect [*] 1. cdr_csv [*] 2. cdr_custom [*] 3. cdr_manager XXX 4. cdr_odbc [*] 5. cdr_pgsql XXX 6. cdr_radius

Hi i want to put an AGI script in a queue, to serve once at time the callers. Example: Queue (8 callers waiting) Agi script / IVR (serving the caller) can i do that? Thanks

i am developing a web application to manage callcenter, i will shortly release it on sf.net this is my problem: i will grant to users the possibility to transfer calls to other users using a web interface, for example if SIP/200 is talking with SIP/400 who wants to transfer the call to SIP/500 i use this commands with manager API: Action: Redirect\r\n Channel: SIP/200-sads\r\n ExtraChannel:

I am experiencing this: 1 - A,B,C are SIP users logged on QUEUEA with ringall strategy 2 - I call QUEUEA 3 - A,B,C start ringing 4 - nobody answer.... 5 - D logs on the QUEUEA 6 - D doen's receive any call, but A,B,C are still ringing How can i avoid that? I'd like that when D joins the QUEUEA it will immediately receive the call that is still ringing on other users... Thanks in

Hi i am planning to develop a php script that will be called from AGI for the management of an IVR application. I'd like to be able to do the following things from php: - retrive callerid - play some audio files to the caller - wait for some DTMF digits - retrive the DTMF - stop the call the php have to collect some information from the user and after some check on a database inster some

Why does voicemail prompt me for an extension instead of just asking my password? [voice-mail] exten => 99,1,VoicemailMain(${EXTEN}@inside) exten => 99,2,Hangup Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

Please, try exten => 99,1,Wait,1 exten => 99,2,Record,/tmp/pcmfile:pcm exten => 99,3,Wait,1 exten => 99,4,Playback,/tmp/pcmfile exten => 99,5,Wait,1 exten => 99,6,Record,/tmp/voxfile:vox exten => 99,7,Wait,1 exten => 99,8,Playback,/tmp/voxfile (put your own extension). Pcm recording is OK, playback is OK. Adpcm recording is noticeably worse. Adpcm playback is very

Barton Hodges wrote: > I've been messing around with a "free line" notification > where an extension is dialed for a second when a line becomes > available. I can't seem to get the "h" extension to continue > when the local party hangs up. I've seen references to other > people having the same problem in the list archives, and the > solution

I am trying to implement a Centralized Call Waiting System. I have red some document about asterisk group features to manage group and category of a sip channel. I have done a lot of test about it but always it doesn't work correctly if I transfer the call. This is the macro code I use for inbound calls. [macro-test] ; ${ARG1} - technology something like SIP ; ${ARG2} - resource.

Hi, I have Asterisk running with the VoiceMail. Using the latest CVS. I have my extensions.conf setup so that if a SIP caller dials *99 the VoicemailMain() as follows: exten => *99,1,Wait(1) exten => *99,2,VoicemailMain() A couple days ago I installed the MySQL/Voicemail support described at http://www.voip-info.org/wiki-Asterisk+voicemail+database Now for some reason

Hi can you help me to build a asterisk manager command event to join a conference? i've seen that there is the event Event: MeetmeJoin Channel: <channel> Uniqueid: <uniqueid> Meetme: <meetme> Usernum: <usernum> Can you explain me how it works? Can i use it to join an existing conference? Thanks

thanks John yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones. it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected: -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new

Dear all i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is possible to configure a scenario like this: 1) receive a call and put it on-hold in a queue (OK) 2) monitor the queue and trigger an outbound call to a remote number using AMI, setting the channel of the on-hold on a specific var named channel2Link (OK) 3) when the remote number answer, trigger an

hi i'm trying to configure a TE205P on asterisk@home i've edited /etc/sysconfig/zaptel adding this line: MODULES="$MODULES wct2xxp" now, when the system is loading, i can see that the wct2xxp module is loaded correctly but if i try the command: /usr/local/sbin/genzaptelconf i get: STOPPING ASTERISK STOPPING FOP SERVER Generating '/etc/zaptel.conf' Generating