similar to: Asterisk 1.8.7 and VoiceMailMain

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like

I have a user, running CVS a/o 11/23/03, who has complained about "phantom" messages showing up days or even weeks after she has deleted them. So I asked her to let me know when it happened again, and she called a few minutes ago. The directory listing below shows a listing of the /var/spool/asterisk/voicemail/default/XXXX/Old directory, and to my surprise the messages are indeed

Does anyone have a quick and dirty script for defragmenting mailboxes? i.e.: -rwx------ 1 root root 80553 Oct 20 16:27 msg0000.gsm -rw-r--r-- 1 root root 218 Oct 20 16:27 msg0000.txt -rwx------ 1 root root 781164 Oct 20 16:27 msg0000.wav -rwx------ 1 root root 79360 Oct 20 16:27 msg0000.WAV -rwx------ 1 root root 7260 Oct

Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL

I have a customer running Asterisk 1.4.22 and FreePBX 2.5.0 that is having problems with Voicemail. They can listen to their voicemail but on the weekend it stopped delivering messages via email. The only thing I can notice is that the permissions for the files on teh voicemail directories are created with no permissions at all. Here is the listing on one of the mailboxes: 4 -------rw- 1

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

Hello everybody, I was trying to install a web interface to my Voice Mail, Vmail.cgi I can log on it, list messages, but no play with the following error msg; "Hrm, can't seem to open /var/spool/asterisk/voicemail/default/234/INBOX/msg0001.WAV" Remark: playing the message msg0001.WAV directly OK Any smart guy up there could help ? Thanks, --------------------------------- Do

Hi, I know I am using SVN, but I was wondering if anybody ever came across this error. I can't read my voicemails because files seems to be corrupted, for lack of a better word. When I do access my messages, I get those errors: [Dec 2 19:45:05] NOTICE[25993]: app_voicemail.c:7432 open_mailbox: Mailbox: /var/spool/asterisk/voicemail/xxx/709/INBOX, expected 0 but found 3

Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email

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

SOFTWARE FreePBX 2.1.3 CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 Sangoma Wanpipe 2.3.4.5 I had an Asterisk server lock up on me today after 95 days of up time. Had to manually kill the Asterisk process and then restart. Nothing out of the ordinary in terms of memory use as far as I could tell. Seems to be running fine now. Here is the log file. I deleted the stuff in the middle to keep it

I am having a problem with the Vmail Web app and was hoping someone has some advice. I have been able to install vmail.cgi. I am able to bring up the vmail web page and login with out problems. However, I never see any vmails listed even when there are vmails present in the directory [root@linux-home INBOX]# ls -l total 40 -rwxrwxrwx 1 root root 255 Dec 2 14:45 msg0000.txt

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'm using asterisk 1.2.14 When asterisk stores voicemail messages in /var/spool/asterisk/voicemail/default/EXTENSION/INBOX files are created with: -rwx------ 1 asterisk web-aster 6690 Apr 17 16:08 msg0002.WAV -rwx------ 1 asterisk web-aster 6732 Apr 17 16:08 msg0002.gsm -rw------- 1 asterisk web-aster 274 Apr 17 16:08 msg0002.txt -rwx------ 1 asterisk web-aster 65324 Apr 17 16:08

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)... --

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 installed asterisk in a 4.11 RELEASE FreeBSD, and we are using two Zoom X5v with SIP and works fine, we can call each other and this is OK ----------extensions.conf---------- [general] static=yes writeprotect=no [sip] exten => 123,1,Dial(SIP/123,20) exten => 123,2,Voicemail(u${EXTEN}) exten => 123,3,Hangup exten => 123,103,Voicemail(b${EXTEN}) exten => 123,104,Hangup

Hi all, In my Asterisk setup, incoming calls through Cisco PSTN gateway to Asterisk extensions sounds work fine. All calls can be terminated properly after hangup. However, when calls were forwarded to voicemail, after recording & hangup the PSTN calls and cisco FXO port remained connected unless cisco port was manually shut/no shut. # key used to hang up the call did NOT help. Did anyone

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 have installed my first Asterisk implementation using the Asterisk@home ISO. I am using the SJPhone software. Using the setup page, I have been able to configure two extensions. Whne I dial from one to the other, the other does not answer even though it is registered. Watching the log in the CLI, I can see that recorded messages are being played;: == No one is available to answer at this time