similar to: Emergency Asterisk Guru Help needed EMERGENCY

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

If anyone could tell me what this error is all about, I would be very grateful. Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not permitted Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not

Hello all, I HAD video working before I upgraded to 1.08 (latest stable with Gentoo) and now it won't work. I just see noise bars and not the video. I know the camera works as I can use it in other programs such as AIM & Yahoo. I have the following setup: sip.conf [general] videosupport = yes port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind

Strange things. When I run the RxFAX command through an internally dialed extension, I can *hear* fax tones, meaning, I presume, that the RxFAX application is running. In fact, doing a show application confirms that. So, I'm presuming RxFAX application is talking as it should. However, inbound fax calls (tones) are not being detected. I know that my extensions file is corect. I am

I have a setup which allows users to access my asterisk box via FWD. That is, a user in say, France can call into a local access number for FWD, then hit number 7 which dumps them into a DISA request for a password, which then dumps them into my internal extension so they can dial out through a VOIP provider. Everything works fine until they enter their tones for the number they are calling

Greetings list, Wondering if some kind soul can help me with an issue with chan_sip segfaulting as soon as it loads... Basically, if sip.conf contains any peers with "host=dynamic" in them, asterisk won't start. Doing -vvvdddc yields the following: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Segmentation fault As

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

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

I've recently upgraded *@home to CVS HEAD and in addition to losing the ability to use the MySQL database, I've noticed that my MOH has degraded significantly. I've tried all sorts of remedies -- removing the x100p card and loading asterisk without the zaptel drivers and such -- still have terrible MOH. I've used .raw. sln. mu. .gsm .mp3 files for the music on hold and it

Hi: This should be easy. I'm running 1.2.15. When a caller calls someone's voice mail, it goes straight to a beep, even though there is an unavail.wav file in that user's voice mail directory. Here is the relevant part of extensions.conf: [internal] exten => 2211,1,Dial(SIP/211,10) exten => 2211,2,VoiceMail(u211@default) exten => 2211,3,Hangup Here is the relevant part of

Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring

We are trying to set up some scripts to test asterisk under various loads. What we are doing is trying to load a bunch of calls in to various queues atuomatically from various numbers etc so we can see how it behaves. I think we can do this by loading files in to the var/spool/asterisk/qcall directory. However the format of this file has a field named identifier which appears to be a file

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

Hello, I?m trying to leverage R_ToplevelExec to implement C level try/catch. The way it is implemented allows for running a function in a top level context. This context gets an empty handler stack, so either the function runs correctly or it jumps. When it jumps, it does not find a handler, so it jumps to the top level context. R does not allow me to call begin context and end context

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

Is there a way for setting up dovecot in such a way that a user can "jump back in time" to an old mailbox "state" in order to retrieve his accidentially deleted mail? -- Ralf Hildebrandt Gesch?ftsbereich IT | Abteilung Netzwerk Charit? - Universit?tsmedizin Berlin Campus Benjamin Franklin Hindenburgdamm 30 | D-12203 Berlin Tel. +49 30 450 570 155 | Fax: +49 30 450

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,