similar to: IAX dns lookups

5-nodes server and 1 node client are connected by gigabits Ethernet. #] iozone -r 32k -r 512k -s 8G KB reclen write rewrite read reread read write read rewrite read fwrite frewrite fread freread 8388608 32 10559 9792 62435 62260 8388608 512 63012 63409 63409 63138 It seems 32k write/rewrite performance are very

hallo all could anyone tell me how to get the * to send keepalive packets over a registration "trunk" or how to increase the amount I'm having natting issues, (the machine is siting behind 2 nat firewalls) thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML

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Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all I been looking into the whole code strugture of chan_iax and i see there is a option to specify the refresh rate of registrations: But there is no code to actually load this from the config file thus i changed the setting in chan_so.h, and recompiled. But still my refresh rate is 60 sec. I need to get this down to 15 sec (nat /pat firewall issue) any ideas? thanks Liaan

Hi, I have successfully configured an IAX trunk between 2 asterisks, calls can go through both ways without any problems, NAT in the middle of course (iptables) Now , leave them for a while , and make a call from the external server , it doesn't go through, Dial from the internal one, everything works fine again.. Now , it is clearly a problem in the NAT engine,

Does anyone know of a AGI script that takes advantage of the weather sound files that's included with the extra sound files available from www.loligo.com/asterisk/sounds/ <http://www.loligo.com/asterisk/sounds/> ? Thank, Jeramie -------------- next part -------------- An HTML attachment was scrubbed... URL:

I know this has come up before but... Has anyone done anything to get an IAX client built on Windows? I thought someone had started one, but I haven't heard anything about it since - and that was months ago? Anyone have any idea what the status is? -- Ron Gage - Saginaw, Michigan I am looking for work - resume at http://www.rongage.org/resume.doc Electrical Engineering, Linux Programming,

Hi all I'm being really stupid today. i simply want asterisk to answer a incomming call, then wait for digits dialed. and then dial that extenstions but i keep on getting: WARNING[3314]: pbx.c:2017 ast_pbx_run: Invalid extension '5', but no rule 'i' in context 'zap-in' my config: exten => 0,1,answer() exten => 0,2,digittimeout,5 exten =>

Hi Everyone, My name is Joe and I need some help from the Samba community. I have a need to be able to backup both Windows NT and AIX files using one IBM 3590 tape drive attached to a 7017-s80 running AIX 4.3.3. The files that I need to back up on the NT side are on a box that cannot be NFS mounted. It is essentially a "dumb box" with minimal configuration abilities. I am able to back

I have a Digicom TE110p card installed in our exchange. I have compiled and installed libpri, zaptel and recompiled and installed asterisk. I have configured udev as I am running Fedora Core 3. The problem that I have is that when zaptel is not running everything works fine. However when I start zaptel (service start zaptel) then I can make normal calls ok but the 'Playback()'

Hi all, We currently have 2 Asterisk boxes and we pass calls to a fro. All works great except we now need to pass variables between them. For example now on box 1 we have: exten => _23XX,1,SetVar(Foo=1234) exten => _23XX,2,Dial(SIP/${EXTEN:0}@Box2) When the call dials into Box 2 the variable Foo does not get passed... Does anyone have any clever ideas? -------------- next part

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29

Apologies for the double post in re my SPEEX issues, but can anyone report successfully installing libOgg and getting SPEEX to work on Asterisk? Details: Fedora Core 3, Asterisk 1.0.4, VoicePulse Connect. Here's my original query with the CLI log http://lists.digium.com/pipermail/asterisk-users/2005-February/ 088225.html Thanks. /rg

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan. ? sip.conf ? [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=outside secret=1234 host=dynamic [2001] type=friend context=outside secret=1234 host=dynamic [2002] type=friend context=outside

Dear all I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan I have setup asterisk with avaya system i have 5 avaya system on 5 location i use 16XX,22XX,33XX,44XX,20XX to reach avaya extentions but when

I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule: exten => _21XX,1,Dial(Zap/g1/${EXTEN:2},20) where 21 is just a prefix to indicate it?s an analog extension and XX matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on

hi there, whenever i try the clusterwise regression from the fpc package, there occurs the following problem: the first cluster is always designed in a way, that when i run a normal linear regression on the independent variables to describe the dependent variable (only on those respondents from the first cluster) - then the regression uses only one independent variable that describes the whole

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

Hi, all I am doing "prrof of concept" system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration. At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be