similar to: Asterisk in VMWare, how does it perform and what is the limit?

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

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" > > > > .... > >

> > Its up to 5.8G of resident memory with 28321 calls processed. > The OOM killer is going to kill this soon at this rate (8GB RAM machine). > This seems like a pretty serious problem. > It looks like I'll need to restart asterisk every night.... Hi the number of cpu cores that you see with top times 512Mbyte is the level of ram that's needed e.g. a hp-gen8 with 2 octo

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

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. >

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?

Hello all. I am trying to use WINE to make a windoze-only development environment (very rudimentar, command line cross-compiler and stuff) under Linux, because GNU make is far superior from the M$'s 'nmake'. Anyway, I have some trouble with expirating licenses and stuff, and the solution is to revert the system date back to some valid thing and then compile. Since I am now running

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'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

Hi all, Disregard my previous posts, I've consolidated everything here. I'm having terrible performance issues with samba over a WAN (point-to-point T1 link). Doing a copy of a 2MB file from a samba server to a linux client running smbclient takes over 5 minutes. SCPing the same file takes seconds. The server is running samba version 3.0.25c with kernel 2.6.16.18. I've put up a set

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 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

Why? Because we have a mixture scattered about the codebase, but only a very few with `.TXT`. I like consistency. It makes my obsessive tendancies happy. Is this important? Nope. Why am I asking first? Because its possible someone, somewhere will be disrupted by this so I figured I'd ask first. -Chandler -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

On Thu, Jan 17, 2019 at 6:51 PM Hubert Tong < hubert.reinterpretcast at gmail.com> wrote: > I'm wondering if this would wreak havoc upon semi-case-sensitive systems > like Windows. I'd like to hear from people who use Git and SVN on Windows. > FWIW, I will of course not do this in any case where there might be a collision due to such file systems. I don't think there

> 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, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI

hello all! After having read http://www.xiph.org/ogg/vorbis/listen.html and listened to the examples on that very educational page, I decided to augment the info. This example is more simple, it involves the comparision between two streams of the same radio station, FranceInter (a station of Radio France in Paris). The ogg stream is running at around 30 kbps/11 kHz in stereo. The WMA stream

Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone

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