similar to: Hangup detection failed

Hello there, I tried to found documentation about openloop disconnect on Asterisk/Zaptel. And up to now, I didn't find anything. Is openloop disconnect supported by zaptel/wcfxo drivers? Other question: maxsilence and silencethreshold don't seem to have any effect on voicemail. Could it be possible that in certain situation it just wont work? I tried adding some verbose in

hi, i am trying to do autoattendant but failing. as in the manual i inserted the background(welcome-mainmenu) file so that after the sound the caller can dial the extension he wants to call. i figured that the background sound wasn't coming in the asterisk. how do we do this without first loading the welcome message? for example after certain rings the caller can dial the extension no to

I'm running the a very recent CVS version of asterisk on an RH9 machine. My problem is that my x100p takes about 10 seconds to detect a hangup. After that it takes about 10 more seconds for the the zaptel device to release the line. Here's an example of my console report: == Parsing '/var/spool/asterisk/voicemail/default/101/INBOX/msg0000.txt': == Parsing

Where are user's voicemail passwords stored and how does the asterisk administrator change them? TIA, Jeff Heath

Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten => 83086921,1,Answer exten => 83086921,2,Dial(SIP/stefan,5,r) exten => 83086921,3,VoiceMail,u111 exten => 83086921,4,Hangup exten => 83086921,103,VoiceMail,b111 exten => 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 =>

Hi, Try as I might, I can't get hangups detected on a Zap channel with loop start lines. So, after someone leaves a voicemail and then hangs up, Asterisk doesn't know it, exits VoicemailMain2, and loops back to the corporate greeting, tying up the line even though the outside caller has hung up. Therefore, I've added the following hideous hack - er, code - to voicemail2.c. It

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

I know this has been moved to contrib, but is anybody using it successfully? We are looking at using Asterisk for the fine IVR features it has, tied in with another platform. Calls are getting routed to it, but the following is happening: (I've redacted phone#s and passwords) asterisk log: Sep 17 11:56:24 WARNING[17423]: No entry in voicemail config file for '+13609XX2000' Sep 17

What I'm trying to achieve is that a voicemail message should be at least 3 seconds long for it to be saved, but *after that* a prolonged silence (e.g. 10 seconds) should terminate the call and recording. My current settings (Asterisk 10.7.0 and 11.2.1) are: ; Minimum length of a voicemail message in seconds for the message to be kept ; The default is no minimum. minsecs=3 ;

Ok its probabally something really eaisy im missing. I've searched the archives and voip-info. Asterisk is trying to send the email notification for voice mail. But the log says Invalid sender. Sender = asterisk@localhost.localdomain and not asterisk@somedomain.com as assigned in conf file. VM Config: [general] format=gsm|wav49|wav serveremail=asterisk@somedomain.com <----Actual

The advanced voice mail options in RC1 is not providing any functionality except "4 - place outgoing call" and "* - return to mail menu". Below is a sample line from voice mail.conf: 210 => 234,test1,test@dotality.com,,|attach=no|cid=yes|review=yes The is a rew install from the tgz file. No patched have been made to the source code. I did try to apply the advanced10

I am having a bit of a problem getting AMD to work on a new server. On my regular office server it works like a charm. I am running Asterisk 1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and I am using a SIP trunk to send out calls (the same one on both servers). Here is the output of a call on my office server: -- Attempting call on Local/0445540881644 at CC2 for

Hi everybody, I'm trying my Asterisk to send emails when a new message arribes to a voicemail user but no email arribes. my voicemail configuration is the following: VOICEMAIL.CONF: [general] format=wav serveremail=asterisk at mydomain.com attach=yes maxmsg=20 maxsecs=180 minsecs=3 maxsilence=10 silencethreshold=128 maxlogins=3 fromstring="My Asterisk" When I look at maillog file,

Hi, I've placed a demo search engine of the asterisk users archive here: http://asterisk.nextrieve.com/cgi-bin/asterisk I know there are a number of other ways to search this list that have been suggested and one person suggested that another wasn't necessary but this engine will do some things that the others can't. Specifically you can search within selected months and from

Here is what happens on startup: Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime: Static SQL: SELECT category, var_name, var_val, cat_metric FROM settings WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime: Everything is fine. Sep

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

Asterisk Users, I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System 2.9.18-4-amd64. A TDM03B is installed on the Debian System. Every time, I try to change my voicemail pin via the Sip phone, the voicemail.conf does not get modify and I see this warning message on the Asterisk command line: [Aug 29 00:12:23] WARNING[19142]: app_voicemail.c:799 vm_change_password:

Hello, I have moved from Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k (debian pkg) to CVS-HEAD, and realtime. Compiled no problem and now running, with realtime extensions and sip users in postgres (ODBC connection) database, trunking also works. I have looked on google, wiki, and this mailing list, along with talking to some peers, but to no avail. My problem revolves around voicemail. I have looked

I've been using Asterisk 1.2.6 with a 4 port FXO Sangoma A200 card for the past month without many problems (other than the fact that the Sangoma card doesn't disconnect hung up calls immediately, which I posted about in another thread, and has still not been fixed), however, I had a call from one of our clients today and they complained that our phone system kept ringing through when they

I'm having a bit of a problem here: I have a * box with a fritz isdn card (running capi 2.0 and chan_capi) and a x100p card for testing purposes. As a proof of concept, I wanted to be able to dial into the * using the isdn line, listen to a message, and enter a 3 digit extension number. If this happens, I wanted the * box to dial out using the x100p card, into our PBX (Nortel Meridian). If