similar to: How to customize voicemail greeting

Hi there, I am configuring a small voicemail server and I am facing the following problem. Executing this command: exten => 1234,1,VoiceMail(${NUMBER}@test) When a user does not have a customized temporary greeting vm-intro message is played asking for the message to the user but when the user has already a temporary greeting both the temporary greeting and vm-intro are played. Basically

I'm using Asterisk 1.4.13, the latest released version. The linux platform is FC7. I setup my Asterisk server to use IMAP storage. Dovecot is the IMAP server. Its storing messages perfectly--no problems. I should also mention that I'm using MySQL for real-time configuration. That must be working (at least partially), because I can authenticate v. the voicemail table. However, the

Hi, I have a wav file recording that i want to use on my voicemail, how can i set this up? thanks!

Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the "The person at extension..." message, not the greetings I have recorded. Thanks -- asterisk*CLI> show dialplan macro-stdexten [

Making custom "voicemail greetings" seems fairly straight forward, and I've done it. However, I'm looking for a way to make the actual extension answer with "You've reached my Jim Dandy voice mailbox, go take a flying . . .". (OK, so maybe not), instead of "The person at extension xxxx, is unavailable" Possible? Easy? Under my nose? joe a.

I'm struggle to answer a simple question. One user at extension 4000 wants a custom .gsm file to play for their mailbox. I can't figure where to put it/what to set in voicemail.conf to achieve this: voicemail.conf 4000 => 4000,system,voicemail at ....net Relevant extensions.conf line: exten => 2,n,VoiceMail(4000 at voicemail) It all works fine, playing the system VM greating, but

I have voicemail configured to store messaging in an odbc database. Does anyone have any thoughts on how best to play back someones greeting from the db? Thanks, ~jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080212/09c64a88/attachment.htm

I'm using Asterisk to read voicemail users out of a SQL database. I am assigning users real phone numbers as their voicemail box. The problem is that if I re-assign a phone number (say, 972-245-0001), the new user is stuck with the old user's greeting and saved messages. What is the best way to resolve this? I don't want to use unique mailbox ids because my dialplan looks like this

I'm trying to apply the following patch to CVS 1.07 and I after running patch -p0 < vm_tempgreet.patch.txt I get the following error: patching file apps/app_voicemail.c Hunk #1 FAILED at 176. Hunk #2 FAILED at 1263. Hunk #3 FAILED at 1293. Hunk #4 FAILED at 1331. Hunk #5 FAILED at 3303. Hunk #6 succeeded at 3562 with fuzz 2 (offset 206 lines). 5 out of 6 hunks FAILED -- saving rejects to

Hi, I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN -> SPA 3102 -> SIP Proxy -> Asterisk The problem I am having is that when the phone hangs up, SPA 3102 can't detect it and relay the CANCEL message. Is this problem with my SPA 3102 config or it just works like that by default? Thanks in advance for your help. Regards, Mark -------------- next

Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. Best Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081119/e74ef6b1/attachment.htm

Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked

Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears?

Hello; I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac Pro intel box. When I try to record a message from an incoming call or a greeting message from internal phone using voicemail, It's like something is not doing well. I can heard anything, only a distorsion sound that is equal to lenght of the message left. First I thoug that could be something with format=gsm|wav.

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

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

I can not figure out what the heck is going on. I went back to my old version and I still get errors when the voicemail system tries to load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11

Hi I am working on deploying voip for my company and would like to seek some advice on the number of E1 lines we need to rent. Our telco told us that there can be at most 30 concurrent channels on an E1 line. Typically, what is the maximum number of DIDs that we can allocate to that E1 line before users get frequent "all lines are busy"? We are running a support center with mostly

Hello Sir, I want some help from you. I have my configuration like this. I have three machines. PC1,PC2,PC3. hub hub PC1--------------------PC2-----------------------PC3 eth0 eth0,eth1 eth0 I want to work my configuration like diffserv. I am generating traffic from PC3 and sending it to PC1 via PC2(as a router)and from

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to obtain the Caller ID if the calls are from the phone line. exten => s,1,Answer() exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN} routing to ${phonenum} ) exten => s,n, Verbose(1|callid is ${CALLID(num)}) exten