similar to: RE: [Asterisk-Dev] Hardware details for the Digium TDM400P

I'm considering that board as a mail and voip gateway for home use. In view of all those statements about how little resources asterisk needs, did anybody already try running asterisk on it? Thanks, Bruno.

Hello folks Anyone know if this would be enough to decode oggs? AMD Elan SC520 133 MHz If not, what seems to be the lower limit required for decoding oggs? Many thanks, Kerry. <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only

Hi, all I''ve got problems with tc qdisc ingress. I''m using vanillia kernel 2.6.14.4 patched with http://www.ssi.bg/~ja/routes-2.6.14-12.diff, and iproute2-2.6.14-051107. i am using ingress to limit incoming traffic : (DEV is eth1 / DOWNLINK is 7700) # attach ingress policer: tc qdisc add dev $DEV handle ffff: ingress # filter *everything* to it (0.0.0.0/0), drop everything

For what it's worth... 1. You cannot do a recursive rcp to a 4.9-RC1 machine. For example, "rcp -r somedir testbox:/tmp" fails. The attached patch fixes this. 2. Only the dial picobsd build will compile. The others (bridge, isp, net and router) all fail at various points. _________________________________________________________________ Get MSN 8 Dial-up Internet Service FREE

Hello everyone, A few days ago on *-dev I proposed the idea of making AstLinux images on a routine basis as a test platform for Asterisk. The ultimate goal is to have a web driven interface (accessible to the public) where users can download the latest and greatest versions of Asterisk (HEAD, STABLE, and recently 1.2.0-beta*), apply optional patches from Mantis (or elsewhere), compile

hi, i facing a problem here. in my sip.conf, i specify a account like this, [1234] type=friend context=from-sip username=1234 secret=1234 nat=no canreinvite=yes dtmfmode=info mailbox=1234@default disallow=all allow=ulaw so i am able to login with username 1234 and password 1234 but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

is where anyone who knows what is needed to get the pickupexten (*8) running ? gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff I've activated it in features.conf (default *8) and also tested other extensions res_features.so is loaded show features says: Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind

Hi group. I have huge problem. My pickup exten #8 isn't working. This is what I have configured. pbx*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 #8 In sip.conf I have callgroup=2 pickupgroup=2 For called party and same for person that is trying to pick up the call. The person that is trying

Hi, I have several queues configured at my call center for different support levels. Today, something weird happened: - A client called queue 1 and was answered by an agent - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf - The user transferred the client to another Queue, by using the second channel and the XFer key of her

This evening I posted a new set of Digium G.729 codec modules to our FTP server and web site, for Linux x86 and x86-64 processors. They were built using GCC 4.0.1, and they now report the processor they were optimized for when they are loaded. The previous x86-64 module required a non-standard Asterisk binary configuration, so this was corrected. In addition, there was only a generic version

Greetings everyone, I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy allows you to communicate with multiple Asterisk boxes from a single point of contact using a variety of I/O formats, now including support for XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format. Astmanproxy is

] Hey ho, A few days ago we released the linux version of the phone, today we are very happy to have the mac version ready for a little field test. Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php At the same time, we also put a newer version of the windows and linux versions online. Let us know how you feel about it, a more mac look (brushed metal) is coming.

Hi All, I have a custom x86 board (using the AMD SC520 Elan CPU) which I am trying to get Linux running on. This is a true embedded board, not a mini PC / PC-on-a-chip arrangement. It has no disk drives (only DRAM , NOR Flash, some battery backed SRAM, a couple of Realtek 8100B Ethernet chips, LEDs, Hex Switches etc) I chose to use U-Boot as the bootloader as it claimed to already have x86

Hi, Just update my 8.x kernel sources last weekend, and newly built kernel did not boot for me: link_elf: symbol mem_range_softc undefined KLD file acpi.ko - could not finalize loading kernel trap 12 with interrupts disabled This is stripped down kernel with everything possible loaded from modules. Any ideas? Did not see any warnings in UPDATING... ./danfe

hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. we plan on the following call flows: 1. incoming call from exten 1111 is sent to our UAC with Dial() 2. our UAC makes

i have quake v1.06 installed and running fine under windows. however, running it in wine gives a bunch of errors. see below: prophet% wine --winver win98 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD. Try --winver nt40 or win31 ! prophet% wine --winver nt40 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD.

Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing