similar to: Asterisk 1.6 and RFC4235

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 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 Thank you for your support. The server is actually compromised, I discovered that after making a deep trace using the audit daemon and looking for the kill signal (SIGKILL) that terminates asterisk. I discovered that there is an executable with a random name in the /boot folder that is killing and deleting asterisk !!! This executable is launched by a service in /etc/rc.d/ with the same

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

> 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> >

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. >

> > 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

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

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, I'm trying to use iaxmodem against a very old version of asterisk (1.0.7 - its a debian sarge embedded system), yet when asterisk gets a call from iaxmodem, it says that the "format for the call is unknown". Does anyone know if there is a minimum version of asterisk that is compatible with iaxmodem 1.1.0? 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, 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, 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, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

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

See the following SIP trace. Where in the world does Asterisk get port 1025 to respond to? This is asterisk 1.6.x. Thanks. -- James <--- SIP read from zzz.zzz.zzz.44:9363 ---> NOTIFY sip:pbx1.mydomain.com SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M From: "xxx-xxx-xxxx" <sip:xxxxxxxxxx at pbx1.mydomain.com>;tag=467525dd6fac949do0^M To:

Thanks James, What i need is to make the fax machines connected to the audiocodes mediant 1000 be able to send and receive fax throught Asterisk (connected 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

Hi, I have recently run into the problem with macro implementation in AEL in Asterisk 1.6. I have some older AEL dialplan which runs on 1.4 but it does not on 1.6 and I'm not sure how to solve this correctly. Let me explain... For example, in Asterisk 1.4 I have a macro like this: macro read_digits(digits) { Set(playlist=${SHELL(${PYTHON} ${SCRIPTS}/read_digits.py

Greetings Asterisk Enthusiasts, Last week, about 50 developers gathered in Atlanta, GA, USA at the Georgia Tech University's Information Security Center (GTISC) for a week of discussion about the future of Asterisk. One of the topics that came up was the future of the existing Asterisk release branches. Asterisk 1.2 was released in the Fall of 2005. At this time, Asterisk 1.0 was put into