similar to: PSTN calls do NOT hang up

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 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

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

Hi Neofita, Doug and All. I think I've the same problem but I don't know if it's related to the bug suggested below. I try to explain my behavior: - I dial the voicemail extension. - I hear: "You have 1 new message. Press 1 for new messages, press 2 for... or # to exit" (I listen the complete message or most part of it) - I press 1 - I can hear the first recorded message.

While I know this is not a true asterisk problem, I figure someone where may know. When you click on Messages and it gives you the count of Urgent, New, etc. How can you make the phone gather that information? For example, my phone shows me there is an e-mail. It also sends an e-mail. Yet, when I click on message before I connect to the contact center, it doesn't have any counts. Here is

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

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, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read

Hi, We're experiencing a problem with basic VM distribution groups where messages won't be delivered. VM is called with a command like: Voicemail(20&30&39&... CLI Shows: Oct 1 20:54:33 NOTICE[26943]: app_voicemail.c:1990 copy_message: Copying message from 60@default to 50@default Oct 1 20:54:38 WARNING[26943]: app.c:1125 ast_lock_path: Failed to lock path

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

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

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn?t work. What can I do, thaks Pedro -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: mi?rcoles, 31 de marzo de 2004 12:00 Para: asterisk-users@lists.digium.com

Hi I have recently upgraded from Ubuntu 15.04 to 16.04 and now everytime I shutdown my PC I get a black screen where it says "libvirt-guests.sh [6917] - running guests under URI address default: no running guests" and the PC just hangs forever and will not shutdown. I need to manually kill it in order to shut it down which is a pain in the ass. I have done some investigation and here is

I'm new to Asterisk, but, Managed to get it working for outound calls from my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing

Hello, Am I addressing this issue to the right address? Please respond. 2017-07-04 9:31 GMT+02:00 Jędrek Domański <jedrek.domanski@gmail.com>: > Hello? What is the status of this issue? > > 2017-06-30 22:24 GMT+02:00 Jędrek Domański <jedrek.domanski@gmail.com>: > >> Hi >> >> I have recently upgraded from Ubuntu 15.04 to 16.04 and now everytime I

Dear asterisk users, I am presently playing with an asterisk@home. I am trying to find the best codec solution for my voicemail records. I want to use ARI (Asterisk Recording Interface) to read the messages. I first used the default wav encoding that was not appropriate because my navigator does not handle wav mime types correctly and I have difficulties playing wav files in my basic linux

$ uname -a Linux mugen 4.4.0-83-generic #106-Ubuntu SMP Mon Jun 26 17:54:43 UTC 2017 x86_64 x86_64 x86_64 GNU/Linux $ lsb_release -a No LSB modules are available. Distributor ID: Ubuntu Description: Ubuntu 16.04.2 LTS Release: 16.04 Codename: xenial 2017-07-07 9:18 GMT+02:00 Dominik Psenner <dpsenner@gmail.com>: > Hey > > FWIW, I have virsh running on my laptop too and no virtual

Hi, This is included in the release now; any feedback? Privilege separation, or privsep, is method in OpenSSH by which operations that require root privilege are performed by a separate privileged monitor process. Its purpose is to prevent privilege escalation by containing corruption to an unprivileged process. More information is available at:

I can get vtpm_manager to run if I delete /var/vtpm/VTPM. However, when I kill it with control-c, and try to run it again, I get: INFO[VTPM]: Starting VTPM. INFO[TCS]: Constructing new TCS: INFO[TCS]: Calling TCS_OpenContext: INFO[VTSP]: OIAP. INFO[VTSP]: Loading Key into TPM. INFO[VTSP]: Unbinding 256 bytes of data. ERROR in VTSP_Unbind at vtsp.c:720 code: TPM_BAD_PARAMETER. ERROR in