similar to: SOLVED - 1.4 svn voicemail bug / crash

I cannot access my voicemail and get the following warning in my console: [Nov 25 10:26:43] WARNING[5628]: app.c:935 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/8900/Old': File exists I have also noticed that Asterisk will crash several minutes later after this warning message. I am using the latest SVN 1.4 branch of Asterisk (Revision 48007) and

Thanks for all your help Martin, Guys, This is a good find and hopefully could help someone else. I've been having a problem with forwarding voicemail from one mailbox to another. I ran down the sendmail and soundcard path and came up goose eggs. With intuitive guidance from Martin Pycko (Digium), I switched from Redhat 9 Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it

1.6.2.0-rc1. I am having trouble with voice mail intermittently not working correctly on CHANUNAVAIL. (it may happen for other statuses too, haven't checked). Basically here's what happens: -- Executing [1651xxxxxx at mydids:1] Macro("SIP/ipkall-trunk-14838bc8", "phone,1651xxxxxx") in new stack -- Executing [s at macro-phone:1]

Is there a zaptel mailing list? Here's the error: CC [M] zaptel-1.4/xpp/xbus-core.o zaptel-1.4/xpp/xbus-core.c: In function ?debugfs_open?: zaptel-1.4/xpp/xbus-core.c:171: error: ?struct inode? has no member named ?u?

Asterisk Project Security Advisory - AST-2007-021 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Crash from invalid/corrupted MIME bodies when | | | using

Asterisk Project Security Advisory - AST-2007-021 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Crash from invalid/corrupted MIME bodies when | | | using

I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the

thanks to those who replied. I have managed to get the functionality I was looking for working with a series of Macros. However, it doesn't work as simply as I would like. There are two issues I've run into: (1)Goto provides no way to pass variables between one context and another. (2)I can't find any way to Goto a specific point within a Macro when calling it. Mostly this is a

I got dragged away from Asterisk (somebody made me an offer I couldn't refuse for system administration), but I'm thinking about seeing if I can't get it deployed at my new employer. Regardless, there are two things about older voicemail that used to annoy me: - Dial by name. Has anyone made it so it can be first or last? - Jump to voicemail; you used to have to actually dial the

Used the Directory application instead of the Directory AGI. -----Original Message----- From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Tuesday, January 23, 2007 11:29 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] * 1.0.9 Voicemail record name does not playback in Directory() On * 1.0.9 User logs into voicemail, dials Option Zero, then Option

hey all, I'm getting this warning on the console when I leave a voicemail on my test server: [Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6552 vm_exec: Prefixing the mailbox with an option is deprecated ('u250@local-vm-users'). [Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6553 vm_exec: Please move all leading options to the second argument. This is what voicemail.conf

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've noticed a few differences in my recent upgrade from 1.4 to 1.6.2 that have me baffled. I thought I'd write to the list and see if anyone has any ideas. - In 1.4, app Directory matched users based on the name listed in their voicemail.conf entry. Now it appears that 1.6 matches only on the full name in the users.conf file and ignores the definition in voicemail.conf - In

Hi, I'm a Polycom user who have just been trying out the Aastra phones. I`ve run into a bit of an issue. I cannot configure the voicemail dialout correctly on one of the two phones I am using, even if the documentation seems to state that it should work in the same manner on both phones. I have both a 9143i and a 57i CT. I`ve set both up with provisioning from config files. Both

Colin, Thanks very much for your feedback. I have achieved success. I appreciate it. Thank you and thnaks to the List. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Colin Anderson Sent: Thursday, December 09, 2004 4:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE:

2008/12/9 Olivier <oza-4h07 at myamail.com> > > > 2008/12/9 Tilghman Lesher <tilghman at mail.jeffandtilghman.com> > > On Tuesday 09 December 2008 09:14:11 Olivier wrote: >> > Hi, >> > >> > In voicemail.conf: >> > ; Supported values: >> > ; 'filename' filename of a soundfile (single ticks around the >>

> > Hi All, > > First I've heard of this feature not working from a customer. I did some > digging and this is a common bug in several older Asterisk versions, it has > more than a few patches in the bug tracker. I've tried a few of them but > none will apply to a specific version I'm currently running for a customer, > 1.6.0.28. > > Does anyone have a

Sean and WipeOut...that's great....exactly what I'm after. Works great. Daryl

Hi, Just wanted to pass on a fix that I found with the ARI recordings interface (www.littlejohnconsulting.com) for using a browser to access voice mail. It turned out to be a rights issue and group membership issue. I was planning on moving Asterisk to a non-root (http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-root&diff2=25) user but I had not done this prior to

I have encountered the following problem with the latest Asterisk source (as of 4/23/2006): Someone calls me on my PSTN line, it then dials my analog extension (I have both SIP and analog phones where all analog phones are a shared extension.) After a while, I get a busy signal. How can I further diagnose this? What could be the problem?