similar to: Problem with Unavailable Message Creation

I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error opening text file for o utput -- Recording the message Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

Hi, I am fairly new to asterisk. I am currently testing my first setup. I've been able to debug most of the problems to make asterisk work with my hardware setup until this time. Currently I have the following issue: Voicemail is running but when I test to leave a voicemail thru my incoming PSTN channel (voicetronix / vpb), asterisk will not detect sound (according to the log) on that

Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server.

I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like to use with asterisk, I have set them up using chan_skinny. The phones work well, except the only problem is that it is like the cisco phones are muted. When I talk on the cisco phones I can hear my self through the ear peice, but the person who I am calling can not hear me at all. I have tried various cisco phones from various

I'm having some issues getting app_voicemail_imapstorage to talk to my IMAP server. From imapstorage.txt, I've got the voicemail.conf configured properly, but if I leave a voicemail for extension 9999, I see no indication that the module is trying to reach the IMAP server. What am I missing? # voicemail.conf [general] imapserver=172.16.17.2 [default] 9999 =>

Regarding this message, I've actually been told one caller who has consistently had this problem was using Vonage, but calling from his Verizon line, it worked. This skewed my survey. Therefore I do believe it's the same callers having the issue, and in which case, I think Vonage is to blame. I found this thread:

Hi, I'm having a bit of a problem with storing voicemail messages in an odbc database. I *think* I've got everything configured correctly but messages are stored on the asterisk server instread of in the database. System info 64 bit redhat RHEL 5.1 Asterisk 1.4.26 unixODBC installed used makemenuselect to instal res_odbc and cdr_odbc Back end database DB2 Database name voiceml

Hello, I have setup a test server with asterisk/AMP and have several 7960's connected to it. The asterisk server has a public ip and all the 7960's are behind nat'd routers. When I try to call from extension to extension I get directed straight to voicemail. I do not have any cards installed and instead direct everything to an Ondo server. I have been told it's not an AMP

Hello all, I've installed AAH 1.1 on my VIA C3 powered mini PC. I've made the necessary changes to the * makefile, so the compilation went well. The first thing I did was configuring two extensions from AMP, namely 200 and 201. Then I installed X-lite on two PC's and configured them with one of the extensions: System settings - SIP proxy - Default: Username: 200 Authorisation user:

I would like to convert a file to WAV49 for use with Asterisk using the linux command line. Specifically I would like to upload sounds to use for unavail.wav and busy.wav, but I'd like them to be compressed so that space is not wasted. I tried using SOX but havent found the correct command-line options to make it or Asterisk happy. Ideas?

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.

Dear Folks, [Test_Context] exten => _911.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _911.,2,Set(CALLERID(num)=xxxxxxx) exten => _911.,3,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)}) exten => _911.,4,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)}) exten => _911.,5,Set(${CALLERID}=${CALLERID(num)}) exten =>

Built asterisk 11.8.1 on a Debian VPS. Testing using ekiga. If h261 is checked in ekiga's video format list I have video, and mouse over the video window shows it to be using h261. But then I get the following lines a dozen or more times in the CLI: [Mar 21 16:25:32] WARNING[31818][C-00000010]: file.c:1241 ast_writefile: No such format 'h261' The problem is that I can't seem to

Hello all! I googled lists.digium.com and ser mailing list, but did not find any working configuration of asterisk used as voicemail for SER. This is my config if (uri==myself) { if (method=="REGISTER") { save("location"); log (1, "Registered\n"); break; };

I have an issue getting any incomming calls When the phone rings something picks up and gives it a fast busy. There is no one using Zap/2 it does the same thing with voicemail and voicemail 2 you can see the console output below, I would love any help anyone could shead on this issue, Michael NOTICE[1192484144]: File chan_zap.c, Line 4270 (ss_thread): Got event 2 (Ring/Answered)... --

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Extensions No Problems Panasonic Ext -> SIP Extensions No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1]

I am having problems with * voicemail emails. It will only mail out the gsm encoded messages. I would really like it to use the wav49 format to make it easier on my user's. I have tried specifying it with just wav49, and like wav49|gsm. It emails nothing, but if I set it to gsm|wav49 then sure enough it emails it out. According to the wiki it should send whatever is specified first. Anyone

Hi All, When our users receive a voicemail we send it attached to an email. It used to work fine, encoded in wav49 and read by Windows media player. Recently the default player in the company has become VLC which is unable to read wav49. I am trying to use OGG/VORBIS instead of wav49. I can't get it working: In voicemail.conf: format = ogg The result is as follow: [Mar 11 09:42:17]