search for: jlamanna

...abilizes. at that level. You can disables hypertreading and cut your ram usage to half of that. I can't see what hardware you are using but I think you need to check that the rule above fits your hardware. b.r. Freddi > > > > On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna <jlamanna at gmail.com > <mailto:jlamanna at gmail.com>> wrote: > 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...

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" > > > > .... > > 1780466242 bytes (1780181594 cache) in 2352909 al...

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

...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. > 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...

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

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? >...

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,

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, 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?

...es hypertreading and cut your ram usage to half of that. > > I can't see what hardware you are using but I think you need to check that > the rule above fits your hardware. > > b.r. > Freddi > > > > > > > On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna <jlamanna at gmail.com> > wrote: > 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 &g...

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 >...

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, 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

Hi, I have a PRI problem where it appears that my system is not responding to SETUP messages on a channel. It seems to be retransmitting a significant number of RELEASE messages to clear a call that is most likely to be long gone. This causes a huge issue because I get a bunch of hangup cause 102s (timeout). I'm using a TE410P (1st Gen) as my PRI card. Has anyone seen this at all? Thanks.

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, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Since SetMusicOnHold() is being deprecated, how do we set the channel musicclass from an AGI script? Last time I checked you can't call dialplan functions from AGI. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141005/03df3f3b/attachment.html>

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

...nected to a pri) I know it's not reliable, but it should work at leaste, what should i do on Asterisk and Mediant to make this work? Im quite confuse with all these fax issues :S Thanks in advance > > Message: 11 > Date: Fri, 9 Apr 2010 17:30:23 -0700 > From: James Lamanna <jlamanna at gmail.com> > Subject: Re: [asterisk-users] Fax Over PRI connected to a Sangoma card > - Fax machines connected to Sip Mediant AudioCodes > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: >...