similar to: Voicemail storage question

Displaying 20 results from an estimated 70000 matches similar to: "Voicemail storage question"

2003 Nov 24
2
Pressing 0 in Voicemail causes * to hangup
I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227
2004 Apr 19
0
strange problem with SIP/voicemail
I'm having a very strange problem I've been fighting with all day. It's 2:30am, and I'm stuck. I think the problem may lie with one of my SIP providers, but I'm not sure. I have two ways to call into my test Grandstream. I can call a PSTN 360 area code number that will forward to my FWD number, which in turn is registered with my * box on extension 2030. If I call the 360
2003 Nov 06
2
Voicemail2 vs Voicemail
>> Wouldn't that break everybody's dialplans where they would have to >> replace all occurrences of Voicemail2 with Voicemail and all >> occurrences of Voicemailmain2 with Voicemailmain? > > No, we would register with both names. Is it necessary (with reasonably current cvs) to make any changes in the *.conf files to use Voicemail2, or is that happening
2004 Apr 15
0
external voicemail access - solved (mostly)
thanks to those who replied. I have managed to get the functionality I was looking for working with a series of Macros. However, it doesn't work as simply as I would like. There are two issues I've run into: (1)Goto provides no way to pass variables between one context and another. (2)I can't find any way to Goto a specific point within a Macro when calling it. Mostly this is a
2004 May 12
0
Problems Retrieving Voicemail Remotely
I am having problems retrieving voicemail from outside the asterisk system. My extensions.conf is configured as follows: exten => 7900,1,VoiceMailMain2(s${CALLERIDNUM}) exten => 7900,2,hangup exten => 7902,1,VoiceMailMain2 exten => 7902,2,hangup exten => 7999,1,dial(sip/7999,20) exten => 7999,2,voicemail2(u7999@incoming-pri) exten => 7999,102,voicemail2(b7999@incoming-pri)
2004 Apr 12
2
Voicemail storage in DB
Hey all, Quick Question. I have heard mention that Asterisk has the capability to store voicemail inside a database, instead of storing each voicemail in a separate file under a spool directory. Is this true? If so, does it (or can it) use MySQL? Is there any documentation available showing how to do this? The problem that we are having is that we need redundant voicemail servers
2003 Jun 06
0
sendmail invocation in voicemail and voicemail2 applications
The voicemail and voicemail2 applications both invoke sendmail as /usr/sbin/sendmail -t. However sendmail is invoked using popen(3), the input being terminated when the pipe is closed and not when a . is entered on a new line. While it looks like the code at the moment can't generate a single line with a '.', it might be prudent to apply the following patch which tells sendmail to
2003 Jul 10
1
Voicemail answers, but drops SIP call after about 3 seconds.
I am calling from my laptop to an asterisk box which answers the call and I can hear the voicemail prompts, but the problem is that after so many seconds, MSN Messenger drops the call because it thinks it hasn't been answered by the remote machine. I'm not sure if this is an asterisk problem, or if it is Messenger not knowing the call was answered. Has anyone else run into this sort of
2004 May 02
1
Voicemail or voicemail2?
I'm using the stable branch. Is voicemail or voicemail2 deprecated? TKS Paul pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040502/1b0ab572/attachment.htm
2005 Mar 27
2
Comedian Voicemail Issues
Hello, I have set up Comedian Mail on my Asterisk system. I am using Voicemail not Voicemail2 in my extensions.conf file. The system works great except for 1 thing...It is not possible to create custom unavailable or greeting messages for 3/4 voicemail boxes. For some odd reason 3/4 users are unable to modify the default voicemail prompt with their own custom greeting. The greeting gets
2003 Jul 16
4
voicemail instructions
Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user
2005 Jul 01
0
Voicemail storage
Hi all, Currently when someone leaves a voicemail the message is stored in the /var/spool/asterisk/voicemail/default/(user's ext)/INBOX , as it should. However I've noticed that a copy is placed in the /tmp directory. Once a message is heard and deleted, the copy in /tmp remains. My question is why?, for how long? And is there a way to modify a config file to send it somewhere else?.
2003 Oct 25
1
Voicemail.conf in MySQL is not functioning
Voicemail.conf in MySQL is not functioning where I get the following error from Asterisk messages log file: CLI debug output is as follows: Executing VoiceMailMain2("SIP/2205-3df0", "") in new stack -- Playing 'vm-login' -- Playing 'vm-password' -- Incorrect password '1234' for user '0' (context = <any>) -- Playing
2003 Oct 18
1
Creating new voicemail accounts
I have googled this one to death, and can't find anything. I added a number of new users to my asterisk (current CVS) system. I am using the "Voicemail2" family. I added entries in extensions.conf and voicemail.conf for my new users, and I have tested leaving and retrieving new voicemails for them. All of this works fine. But if one of the new users tries to "Administer
2004 Jan 04
2
Voicemail Out call
There was a post in the 'wiki' for an application to provide an outcall when there is a voicemail is left on asterisk. I am having a problem that this application will only work if the caller presses the pound sign at the end of recording. As most people just hang up, this application isn't working. Can any offer suggestions to accomplish this out call?
2003 Oct 27
3
passing digits for voicemail from sip gateway
I am seeing strange behavior that I don't understand. Voicemail2 and voicemailmain2 work fine if I call from a sip phone directly connected to *, but if I call either of them from an analog line on the other side of a sip gateway, voicemail seems to ignore digits. If I am recording a message and press #, nothing happens except that it records the tone onto the message and I can't specify
2003 Sep 24
1
Voicemail doesn't hangup
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
2005 Feb 03
1
Multiple mailbox on the same SIP extension
I'm wondering if there's a way it will show on the phone when there's a new message. Here's what I'm trying to do : in my extensions.conf when someone call from a PSTN line on my TDM04B card they have a choice. When someone press 1 for sales then I have 3 phones ringing at the same time. Each phone as already there own mailbox because if someone know there extension
2003 Nov 03
1
new voicemail notification by calling #?
Hi guys, This is a two part question about the Voicemail application: Firstly, is there a way built in to the current app which would allow me to have Asterisk call a phone number everytime a certain mailbox receives a new voicemail? I know about the email and pager notifications that are already built in to voicemail.conf, and I've used those but also have a need to do what I describe
2010 Aug 30
0
Voicemail prompts fuzzy and quiet
Strange issue that I can't figure out and I am hoping someone may have some ideas. Two Asterisk boxes running 1.2.34 (yeah I know it is old, but it runs like a top and I am not going to mess with it). *B rsyncs config from *A. *A dies. I bring up *B and it all works fine, except for one issue. Calls to voicemail are garbled and low. Phone to phone and phone to gateway work perfect. If I