similar to: High resident memory with 11.14.0 ?

Hi, I have an Asterisk server that's been running now for around 2 days. I've noticed that the resident memory seems to be very high for its current call load: PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 18321 asterisk 20 0 8050m 5.2g 6968 S 13 66.2 363:11.80 asterisk $ asterisk -rx "core show channels" 24 active channels

Also, how big does the cache in frame.c grow to? I've recompiled with MALLOC_DEBUG on that server: asterisk -rx "memory show summary" .... 1780466242 bytes (1780181594 cache) in 2352909 allocations in file frame.c ... Seems like a ridiculous cache. On Mon, Nov 24, 2014 at 9:02 AM, James Lamanna <jlamanna at gmail.com> wrote: > cat /proc/cpuinfo lists 4 cores. >

cat /proc/cpuinfo lists 4 cores. So even if that's not showing hyperthreading, maximum 8. By your rule, that would be 8 cores * 0.5GB = 4GB memory. I've seen resident memory be up over 6GB. On Sat, Nov 22, 2014 at 1:29 PM, Freddi Hansen <fh at danovation.dk> wrote: > > Its up to 5.8G of resident memory with 28321 calls processed. > The OOM killer is going to kill this soon

On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote: > > Also, how big does the cache in frame.c grow to? > > I've recompiled with MALLOC_DEBUG on that server: > > > > asterisk -rx "memory show summary" > > > > .... > >

On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna <jlamanna at gmail.com> wrote: > > On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> > wrote: > >> On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> >> wrote: >> > Also, how big does the cache in frame.c grow to? >> > I've recompiled with

On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote: > Also, how big does the cache in frame.c grow to? > I've recompiled with MALLOC_DEBUG on that server: > > asterisk -rx "memory show summary" > > .... > 1780466242 bytes (1780181594 cache) in 2352909 allocations in file > frame.c > ... > > Seems like a ridiculous

Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. Thanks. -- James

Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a

Hi, I've noticed that if I have a multi-line linksys (942 or 962) phone with the same sip registration mapped to each line key, that if all the lines are full the phone will accept another call. I would expect the phone to respond with "busy" so the call would to directly to voicemail. Has anyone else experienced this and know of a workaround? I know it seems like an

Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on "sip show peer" shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify?

Hi, Does anyone have any good empirical data suggesting what the maximum number of PRI calls (incoming and outgoing) without hardware echo cancellation can be handled on a single box is? I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of D-Channels going down and then coming back up (See below). I've looked at the number of simultaneous calls at each of these points,

Hey All, We are running a small SIP/IAX termination service at the moment (planning on growing it) with 2 asterisk machines. One terminates the SIP/IAX calls from our customers and one is our gateway to our upstream provider. Both machines are logging CDR data to the same postgres table using the cdr_psql module. The problem I am having is I'm trying to work out how to link the CDR records

Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not

Hi. Between 2 linux (2.6.11 client and 2.6.14 server) machines connected by a 100Mb link I get samba performance copying a file from the client to the server through a smbmounted share of around 4.2MB/s Is this to be expected? Or can it be improved (and if so, how?) I've tried tweaking SO_(SND/RCV)BUF (after reading numerous articles on samba performance...), but it doesn't seem to have

Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even

> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> >

Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a "sip show peer" on those extensions shows them as "OK". Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is

Hi, Does anyone have some good examples of a Kamalio or OpenSips configuration that integrates with Asterisk? Essentially I want to use the SIP router as the UA, but still run all the calls through Asterisk (for dialplan, etc..) I've looked for examples on the project web sites, but I haven't found anything decent yet. Thanks. -- James

Hi, Has anyone had any experience using DRBD to mirror an entire asterisk machine? If so, is there a performance issue at all when people are recording voicemails and the like? It seems like that could generate quite a bit of traffic. Also, do you bother to mirror the log files as well? Thanks. -- James

Hi, Some users have been reporting a peculiar problem. The are having an issue when they dial out to some multi-level IVRs where you make 2 or 3 touchtone choices and then are connected to a live operator. When the live operator connects the operator cannot hear them or sometimes it results in dead air. With the one-way audio issue, is it possible that something has locked the channel into some