similar to: Log Error

Guys. What does this error mean? -- Playing '/var/spool/asterisk/voicemail/intruder/201/unavail' (language 'sp') -- Playing 'vm-intro' (language 'sp') -- Playing 'beep' (language 'sp') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/intruder/201/INBOX/msg0000 format: wav, 0x812b4f0 -- User ended

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

I'm running Asterisk HEAD from March 4. I've Googled a bit but I can't figure out what causes this error: app_queue.c:374 in changethread: Can't change device '**Unknown**' with no technology! It doesn't seem to be causing any problems, but I'm curious what causes it. I did a few Google searches and found a lot of people asking about it, but no real answers.

hi all, i have an asterisk install with a digium 4 port fxo card and cisco 7960 sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz 256KB cache and 1GB of ram. when a call comes in on zap/1-1 for example, the delay between when zap sees the line going to ring state, and when the desktop telephone rings can be as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece).

After I leave a voicemail for an extension and hangup, my asterisk console (with debug turned up quite high) shows two error messages like: WARNING[7664]: app_queue.c:341 changethread: Can't change device with no technology! WARNING[7668]: app_queue.c:341 changethread: Can't change device with no technology! Clues? Thanks! -Dorn

Before I go and try to write something myself, I'm curious if anyone has a script that they're using for setting and clearing the MWI on a legacy PBX. I need to pick up a Zap channel and dial #63XXX to set the MWI, or #64XXX to clear it, where XXX is the extension number. One complication is that I've got a couple dozen extensions to handle the MWI for, and only four channels to work

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

Hello, After upgrading to the latest development CVS Head, I am now getting regular errors as follows: Dec 17 17:07:30 WARNING[8092]: app_queue.c:340 changethread: Can't change device with no technology! Also, my ability to answer calls with XTen Pro softphone seems to be a bit flaky now. Any ideas? ===== Jason Goecke www.goecke.net Ph: +31.707.504.634 Mb: +31.707.504.634 Fx:

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press "#" to end the recording, at which point I am told "Your message has been

Asterisk 1.2, Fedora Core 4: When I leave a voicemail message, it writes the necessary files to the INBOX: msg0000.gsm msg0000.txt msg0000.wav msg0000.WAV When I hang up, the files are erased. There is no indication of anything untoward in the logs: -- x=0, open writing: [...]/INBOX/msg0000 format: wav49, 0x99ce778 -- x=1, open writing: [...]/INBOX/msg0000 format:

I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on

Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-root&diff2=25). When I issue the following: chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk The above command results in the following rights on messages: msg0000.gsm

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

An outside caller started to leave voice mail. The CLI shows: Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg0000 format: gsm, 0x8295d40 -- x=1, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg0000 format: wav, 0x829e2c0 -- User cancelled by pressing 0 -- Playing 'vm-saveoper' (language 'en') Later on,

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

Hello, I am attempting to write a script to launch a callback based on a dial-in service. I have created this call file: --------------------------- channel: IAX2/user@voipjet/011_valid_number maxretries: 3 retrytime: 5 waittime: 5 context: dialtone extension: 912125551212 priority: 1 --------------------------- Where I first attempt to dial the callback user (channel) and then connect the

Ever since I started using asterisk I see often this error message, can sombody tell me what it means? Feb 26 09:20:40 WARNING[29371]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! __________________________________________________________________ Anton Krall

Is there a way to make the asterisk voicemail app play back messages in NEWEST FIRST order, instead of OLDEST FIRST? I see the situation repeatedly where someone needs to dip into their voicemail archive to get something from a recently saved voicemail message, and they have to slog through lots of irrelevant stuff to get there. I have seen this question come up previously on this list without

Anybody knows why ARI gives this error message when I enter extension number and password. *Warning*: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg0000.txt): failed to open stream: Permission denied in * /var/www/html/recordings/modules/voicemail.module* on line *525* It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A