similar to: Voicemail app. not working...

Hi Everybody, as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but VoiceMail application stated that there were no entries in voicemail.conf, so it didn't work. Installed again 1.2.0 and voil? the VoiceMail app. was working again. I asked to the group, but it seems I'm the only one with this issue! In Italy we say: "Chi lascia la via vecchia per la nuova, sa

Hi everybody, I'm in a middle of a Asterisk learning period. I am at a very good point except I'm not able to use VoiceMailMain. This Is my simple dialplan regarding VoiceMail ;Number that the IP Phones dial to access voice mail exten => 22999,1,VoiceMailMain (s${CALLERIDNUM}) exten => 22999,2,Wait(3) exten => 22999,3,Hangup Why do I get Forbidden 403 and one console display

Hi everybody, I have corrected this line in extensions.conf by stripping spaces off and now it executes: exten => 22999,1,VoiceMailMain(s${CALLERIDNUM}) when it runs, the mail box number is asked and password too. I expected no question were made, because I inserted CALLERIDNUMBER and s in front of box number. Anybody knows why? Thank to you all, very kind members of this list! Ciao Mauro

Hi dear group members, I have finally an Asterisk box working, capable of receiving and making calls. I have this issue while calling mobiles from our SIP softphones: -------------------------- linux*CLI> -- Executing NoOp("SIP/2000-6850", "3487024125") in new stack -- Executing Dial("SIP/2000-6850", "ZAP/g1/3487024125") in new stack -- Called

Hi everybody I'm trying to make Asterisk 1.2.1 run under VMWARE and Suse 9.2. I use ZTDUMMY module for timing and ZTTEST gets an average precision of 98,4 %. Is there any way to improve it? Best regards Mauro Zanin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060117/945d544a/attachment.htm

> > Hi All, > > First I've heard of this feature not working from a customer. I did some > digging and this is a common bug in several older Asterisk versions, it has > more than a few patches in the bug tracker. I've tried a few of them but > none will apply to a specific version I'm currently running for a customer, > 1.6.0.28. > > Does anyone have a

Hi everybody, I'm strugling between two devices: the both TE but one was set up as a TN. I have no current on that interface. I have tried to find some circuit over the net to power the connection, both commercial and home made. Can anybody give some hint? Ciao Mauro

Hi everybody, I have this issue: I need to automatically park an incoming call, play a welcome prompt and then connect to some extension but under extension user's command. I was thinking to use a small database to comunicate between asterisk and the main application. Has anybody had this kind of experience? Best regards Mauro

Hi Everybody, I'm experiencing this kind of issue. One ASTERISK 1.2.17 is connected to a Bristuffed HFC single ISDN channel card. Everything seems to work but sometimes the third party caller when listening to MOH listens some "SSHHHHH!" instead of MOH, this is not continuos, MOH plays ok for, say, 20 seconds then the sound and then another 30 seconds of good MOH. We have some

Hi List! I am running 'Asterisk 1.4.22 built by root' I have an issue with voicemails. In the var/spool/asterisk/voicemail/default/extnnnn/inbox the msgnnn.txt files are flagged as readable, this causes asterisk to just skip over the voicemails when listening. drwx-w---- 2 asterisk 4096 2009-12-03 15:17 . drwx-w---- 7 asterisk 4096 2009-12-02 10:55 .. -r--rw-rw- 1 asterisk 267

Hi List, Apologies if this appears more than once.. Apple mail seemed to post a follow up last time that isn't appearing so I've moved to webmail to send.. I am running 'Asterisk 1.4.22 built by root' I have an issue with voicemails. In the var/spool/asterisk/voicemail/default/extnnnn/inbox the msgnnn.txt files are flagged as readable, this causes asterisk to just skip over the

Hello list, My system behaves in an odd manner, and I can't find why. When users leave a message on the voicemail, once the message is recorded and the user hangs up, Asterisk crashes. I can't figure out when it started to behave like this. Here is the extract of the dialplan where it occurs : exten => s,1,Answer() exten => s,n,Playback(radiom-misenrelation) exten =>

I current have a working Asterisk 1.2.12 server with ODBC voicemail storage, realtime static maps for voicemail, sip and iax configuration files. Realtime extensions, etc. All works great. I have verified that this configuration works on my test server as well. Now I am trying to test the 1.4B3 version on the same test server, and all works well except for ODBC voicemail. I am using the same

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

Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten => 2,1,Playback(/media/asterisk/answerphone-en) exten => 2,n,VoiceMail(2000,s) exten =>

HI folks, I figured out where in the source code to hack the .wav file permissions which were set too restrictive for me, but I can't figure out how to do the same for the .txt file. Looks like the voicemail.c file sets it nicely for asterisk1.4beta3 using a #define statement early on, but msg????.txt comes out with permissions 0600 and there are no umask entries that affect how

Which version? Similar issues parsing callback number in 1.2.12 > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Kristian Kielhofner > Sent: Thursday, September 28, 2006 10:27 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Voicemail callback

Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP!

Try different entry in this parameter. In Italy mobiles start with 3, while public services with 1 and normal user numbers with 0. Using pridialplan=none, every number different from 0 was resulting in termination code 1, normally used for number never seen on the network. Ciao Mauro

Hello, I have modified the VoiceMailMain application to satisfy the request of the "local users", i.e., my wife. She lost patience with too many options (we have one mailbox, so we don't need to forward messages, or reply to messages, or file them in 6 different folders...) So the modified app says "Message 1", reads the message, "Message 2", reads...