similar to: Serious memory leak in asterisk (manager)

hi all seems there's a memory leak in asterisk somewhere. probably in chan_mgcp. It seems like 2048kB is allocated but not released each time I lift the handset. This is, however, never released. So... oslpbx:~/debug/20030801112939# ps axfv|grep asterisk | tail -1 3206 pts/0 S 0:00 0 410 35601 6024 1.1 \_ /usr/sbin/asterisk -gvvvvcp (lift handset)

hi Just installed astping in Nagios (former netsaint), and it works fine - for some time. After several hours, the Asterisk manager module just stopped responding, and I had to restart to make it work Any idea what might cause this? Sorry - I don't have the log now. Is it possible to have asterisk log console output somewhere? roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS -

Hello, I pulled the latest from cvs today and ran several tests and added more options to the CFLAGS in the Makefile. To start with, I ran valgrind against sshd & it comes up with this: ==24959== 112 bytes in 1 blocks are definitely lost in loss record 297 of 310 ==24959== at 0x40164650: malloc (vg_clientfuncs.c:100) ==24959== by 0x807A0D1: compat_init_setproctitle (setproctitle.c:236)

Hi All, I am trying to get Asterisk up and running on my new Mandrake 9.1 install. I've installed Linux in the "standard" mandrake security mode, and "su" to do my attempts at install. I managed to obtain the source from CVS, and have been able to compile Zaptel. I then ran insmod zaptel, and also make config. I think I have compiled and loaded Zaptel successfully as

I have spent some time digging through the archives for comments concerning Asterisk and monitoring systems, and I have found few results. check_asterisk.pl.gz (http://www.dynx.net/ASTERISK/misc-progs/) which gives an error on download, and has no further Google references astping.tar (http://www.dynx.net/ASTERISK/misc-progs/ and also in the mailing list archives) supposedly sends a query to

I am running 2.2.6 on a customer's site and everything has been perfect. However I noticed the other day that one of the two nmbd processes (it is a wins server) had grown to 13Mb over a two week period. I needed to update to the latest CVS version anyway and had hoped that something in that might fix things. Five days later and things are looking the same: nmbd started on Nov 9th: root

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to

Hi, I'm trying to write a tool, which shows me the state of the current calls. For this purpose I'm reading from Pipe the Asterisk output and parse it... asterisk -vr | mytool However, the problem ist how to get the information about who got this call in the group. The Zap channels are assigned dynamical. Only thing I can see which channel is connect to the caller but not who is

For doumentation purposes I run a batch R program that does reshaping etc. of data and finally prints things. When I paste the code into an R-window everything runs in a few seconds, but in Batch mode i get in the output file the result below. Any clues why R uses 4 min to print a 1129\times 6 dataframe in batch mode? The scrit I use to run R is: c:\stat\r\%R_VERS%\bin\Rterm --no-restore

Hi, This morning I had some failed calls. On the console (and in the log) I saw the error "Unable to allocate channel structure". Before I restarted the process, I checked it's memory usage in ps and glanced at my free memory in top. Asterisk was using a normal ammount of memory, about 40M. I don't think this was a system limit. This was running Asterisk v1.0.2. Below is

I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB of memory. This is serving about 75 sip clients, Polycom500's and 600's. We are running into problems with the memory. Asterisk, right now, is using about 1.8GB of system memory. I am using Asterisk 1.0.7, Zaptel 1.0.7 with Digiums TE410 1xT1 RBS and 1xT1 PRI, Libpri 1.0.7 on Fedora Core 3. My question; is this

Hi All: I am trying to compile asterisk with oh323 but I can't compile it. I am using instruction provided at http://www.oinko.net/astrecipes/index.php?from=1&q=astrecipes/compiling+asterisk+with+oh323. The compile error I am getting is as follows. Quite a few other people are getting exactly same error but no one has posted a fix for this error yet. Any help would greatly be

n active project, than a dead one. Otherwise who is going to patch vulnerab= ilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

I have a pretty strange problem. I have about 100 DID's that come down a PRI from SBC in the United States. On Friday afternoon, one of my DID's flipped out. When you call it, the SBC operator comes on and says that the line has been disconnected. I contacted them and they ran test and they are telling me the problem has to be on my end. My problem is that the CLI never shows the

Hello Philipp and All, My scenario is a bit different than the one I had explained before. I'm sorry. Let's suppose I have someone calling one of my Asterisk clients. This asterisk client has CFB (Call Forward Busy) activated. The forward number is a Voice Mail System, however is not a Asterisk's Voice Mail. It is a third party Voice Mail System, that has a SIP Trunk with my

Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, _________________________________________________________________ Expand your wine savvy — and get some great new recipes — at MSN Wine.

I've looked all around, and I can't find an answer to this. I apologize if this has been discussed already or is buried somewhere in voip-info.org. I have an asterisk setup on linux 2.6.11.11 kernel, a revision E/F TDM400P, and Polycom IP501 phones. As soon as I load the zaptel module into the kernel, the voice prompts and voicemail system ceases to work. The asterisk logs

Dear all, I'm wanting to do a series of comparisons among 4 categorical variables: a <- aggregate(y, list(var1, var2, var3, var4), sum) This gets me a very nice 2-dimensional data frame with one column per variable, BUT, as help for aggregate says, <<empty subsets are removed>>. I don't see in help(aggregate) how I can change this. In contrast, a <- tapply(y,

I have a number of Polycom phones to setup with my * server. For my initial few phones I hand wrote configs. Does anyone here who uses Polycom phones have some form of management utility for automating their setup? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.