similar to: Howto disable adsi in app_voicemail.c so I can noload *adsi*.so

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

I have searched through the archives to no avail. Can someone give a quick overview or starting point for ADSI. I just received a 390 and was told that it is unlocked. But am not sure of the next step... I have tried a limited test using the adsi.conf and received: -- ADSI Available on CPE. Attempting Upload. WARNING[106510]: File app_adsiprog.c, Line 1211 (adsi_process): Invalid or

I sent this yesterday from my wrong email account. yeah, Thank you. I screwed up by only looking at adsi.conf, and not asterisk.adsi. Once I saw that, things started falling into place. Are you able to pass arguments to the phone so that for every call there is unique data on the screen. I see that works for caller id with $Call1p and $Call1s, but what If I obtain an OrderID from my

I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03

I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

Hi all, I have a demo machine I'm running up on Lenny - it has the packaged Asterisk version installed (1.4.21.2+stuff). I'm trying to add an extension to leave a voicemail message, just with Voicemail(1234), which I've done before (on 1.2 at least), but it's saying "no application 'Voicemail' ". "module show like voi" shows

If anyone is experiencing this type of registration error: Registration from '<sip:40852@192.168.70.2>' failed for '192.168.70.25' try adding the following line to your modules.conf file noload => app_adsiprog.so This error is clearly asterisk trying to register with the phone and not the other way around. the app_adsiprog.so is only used to download adsi scripts to a

Hi, I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following, WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue

Hi I was trying to have realtime voicemail working with ODBC Driver. Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page ) But I do not manage to make it work with ODBC. Outside Asterisk, ODBC works fine, I can access my databases & tables ! Asterisk fails to

Hi, Would anyone know of a way to set the time automatically on an ADSI capable phone from *? The phone in question is a Aastra 480e. While I am at it, does anyone have any helpful docs on the ADSI script programming? I have managed to do basic functions by modifying the asterisk.adsi file using stuff gleaned from the app_adsiprog.c file, but docs would be really helpful at this point. Tia,

Hi I am trying to make asterisk load config from database, so far I get the sip, extension working, but voicemail seems still looking for config file, not from the database. the extconfig.conf looks like ... sipusers => mysql,asterisk,sip_buddies sippeers => mysql,asterisk,sip_buddies extensions => mysql,asterisk,extensions_table voicemail => mysql,asterisk,voicemail_users .. the

Hi Looking at the source of app_voicemail.c there are many statements like: ast_debug(1, "%s doesn't exist, doing what we can\n", prefile); Where do I have to enably this to be showed in the console or logged to a file by logger. core set debug does not seem to help here. Well, my actual problem is, that if a customer has recorded his own greeting, he

Hi *, I have set up a very simple asterisk configuration where I intend to be redirected to the voicemail whenever I dial 100 with my kphone SIP client. The problem is that asterisk can not find the stream 'vm-theperson'. I have made a non-standard installation (since I am just testing), and that file is located in /mnt/tr2/fake_root/installed/usr/local/var/lib/asterisk/sounds. 1. How

Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish prompts that can handle for example, instead of saying "trabajo mensjes" would say "mensajes de trabajo o mensajes trabajo" (inverse)? Also can handle singular and plural (mensaje vs. mensajes)? Anton

Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Hello, >From Astricon 2019 notes [1], you can read "[a]ll 3 app_voicemail variants can now be built". What does it mean ? Is this change tied with a specific Asterisk version ? Is possible to change from ODBC to IMAP without re-compilation ? Is it also possible to mix mailbox types on a single system ? [1] https://wiki.asterisk.org/wiki/display/AST/Agenda+2019 Best regards

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing