I have Asterisk 1.4.15 installed on a Fedora Core 8 machine. Asterisk is re-starting intermediately i.e sometime it does not re-start the whole day, and sometime it just re-start after few calls. There is no log of error in /var/log/asterisk/messages but if I see the /var/log/messages I can see lines similar to below for Asterisk Process almost the same time asterisk re-start segfault at fff10098 eip 0809b0b5 esp b793e370 error 4 segfault at 00002dfc eip 0809b0c5 esp b781f370 error 4 In the dialplan we have used MixMonitor() to record the calls. Can anyone help me on getting to the root of the problem or fixing it? Thanks in advance. Regards, Sanjay Rajdev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080505/4553d358/attachment.htm
Hi, i will try to setup 3 * box, 1 ser. if none, let's say i have 4 extensions 101, 102,103 and 104, 101 registered on * 1, 102 on * 2, 103 on * 3 and 104 on * 1 also. i will define this dial plan: [dial-extension] exten => _1XX,1,Dial(SIP/${EXTEN}) <- look it up on the local first exten => _1XX,1,Dial(SIP/${EXTEn}@ser) <- if not on local, check in ser when ser receives a request, it will ask the other 2 * servers where the request did not come from, e.g if call originated from ask * 1 first if not then ask * 2. Do you think there will be issues? i also plan on installing heartbeat so users will only register on a single host., or can i simply use DNS SRV? is my setup possible? TIA regards, nhadie --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080505/1e0597e9/attachment.htm
Sanjay Rajdev wrote:> I have Asterisk 1.4.15 installed on a Fedora Core 8 machine. Asterisk is<snip>> In the dialplan we have used MixMonitor() to record the calls. > > Can anyone help me on getting to the root of the problem or fixing it?We have fixed a _lot_ of issues in that area of the code since 1.4.15. I would suggest trying the latest version. If it still gives you trouble, please let us know on http://bugs.digium.com so that we can fix it up for you. Thanks, -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc.
On Wed, May 7, 2008 at 11:06 PM, Russell Bryant <russell at digium.com> wrote:> Sanjay Rajdev wrote: > > I have Asterisk 1.4.15 installed on a Fedora Core 8 machine. Asterisk is > > <snip> > > > > In the dialplan we have used MixMonitor() to record the calls. > > > > Can anyone help me on getting to the root of the problem or fixing it? > > We have fixed a _lot_ of issues in that area of the code since 1.4.15. I would > suggest trying the latest version. If it still gives you trouble, please let us > know on http://bugs.digium.com so that we can fix it up for you. > > Thanks, > > -- > Russell Bryant > Senior Software Engineer > Open Source Team Lead > Digium, Inc. >I suggest upgrading to 1.2.X unless you NEED the features in 1.4.x. Give it another six months to a year. It just works and new releases in 1.4 such as Zaptel obviously have had zero testing before release. It is bug after bug. Thanks, Steve Totaro
I had a problem in the dictate app, which I have fixed. Thanks for the help. By the way here is a description of what was happening. app_dictate does not close the file descriptor after the call hangs or a new dictation starts, as and when the dictation increased the count of open file descriptor increased and forced the asterisk process to reach the limit of allowed maximum number of open file descriptor. S o I added ast_closestream(fs), where ever I thought it was necessary and at the end I checked for if(fs){ ast_closestream(fs) ; } this line was causing the problem, in case the file descriptor was already closed it was still going into the if and trying to close a closed descriptor. I have made change to set fs = NULL everywhere after ast_closestream(fs) I am not a developer for Asterisk and even cannot make changes in the SVN as I do not know lot about the branches in it, but if someone from your side can take the effort to change this It would be great help for others. Regards, Sanjay Rajdev ----- Original Message ----- From: "Russell Bryant" <russell at digium.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Sent: Thursday, May 8, 2008 8:36:14 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Asterisk Restarting due to segfault Sanjay Rajdev wrote:> I have Asterisk 1.4.15 installed on a Fedora Core 8 machine. Asterisk is<snip>> In the dialplan we have used MixMonitor() to record the calls. > > Can anyone help me on getting to the root of the problem or fixing it?We have fixed a _lot_ of issues in that area of the code since 1.4.15. I would suggest trying the latest version. If it still gives you trouble, please let us know on http://bugs.digium.com so that we can fix it up for you. Thanks, -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080508/c7394d2c/attachment.htm
Sanjay Rajdev wrote:> I am not a developer for Asterisk and even cannot make changes in the SVN as I do not know lot about the branches in it, but if someone from your side can take the effort to change this It would be great help for others.Please open a report on http://bugs.digium.com that describes what you have found. We will help you get the fix into svn. Thanks, -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc.