similar to: User on PC?

Hello I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? Thank you.

Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description "Asterisk daemon" start on runlevel-2 stop on shutdown respawn exec

Hi, I'm very new to asterisk and managed to set one up in debian, I installed via apt-get the asterisk and asterisk-bristuff packages. I downloaded the bristuff source as well. I managed to get as far as loading the following modules: zaptel 199144 4 cwain,zaphfc,zttranscode,ztdummy But the problem now is that all channels that I try to configure gives me an error that

Hi List, Does anyone have experiences to relate on the various Asterisk-based PBX appliances out there? Like the Aastra 160, Digium S844i, etc. Do the Epygi Quadro and Grandstream GXE also use Asterisk? Thanks, Femi -------------- next part -------------- An HTML attachment was scrubbed... URL:

According to http://kb.digium.com/entry/12/ The Iaxy will respond to pings on port 9999. You can ping your broadcast IP on your network and listen with tcpdump on your network on port 9999 which will show the Iaxy responding and what IP address it is coming from. Ex. ping 192.168.1.255 tcpdump -i eth0 udp port 9999" Before I get my karma whacked again, does this work for

Hi All, I thought I read a post a while back of a system call or something in the dialplan whereby a call count can be incremented and spit out to a text file. Not like a group count of active channels. What I would like to accomplish is have an incremental count of a specific dialplan routine that gets called, so after a week or month, I can see how many times a specific dilaplan action has

I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Leif

Hi, I'm using gotoiftime on asterisk, but it seems  there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Regards, nhadie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! As the upstream of my DSL-connection is very slow, I'd like my sip-phones to use iLBC to connect to my *. My gateway provider only allows ulaw. Hence, I'd like to use the follwing setup: SIP-phone <--iLBC--> Asterisk <---ulaw----> PSTN-Gateway I get the following error: "Unable to find a codec translation path from ilbc to ulaw" Setup SIP-phone: disallow=all

Hi We have a number offices accommodating 4-6 people each hence it is very important for PBX to be fanless and silent. We have been looking at using IDE flash disks also called DOM. The performance tests we have done so far satisfy our requirements, however we are concerned with DOM durability. We have installed debian and vanilla asterisk on 1GB DOM. All seems to work fine at the moment however

Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? Heck, it doesn't even need to be able to do calls :-) Leif

I'm running asterisk on a Freebsd with 2 Nic's. Inside NIC is 192.168.5.x where the phones are. Outside NIC used to be a public IP with the ISP's device set to bridging, but the new WiMAX router only offers me the public ip 94.18.x.x on the outside, and forwarding everything to 192.168.1.50 on the "Outside NIC" Some of the phones are being disconnected with Asterisk

Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call

I made a patch to app_dial.c to make Dial(ext1&ext2&ext3,tumeout,B) return busy when just one extension is busy. http://www.neland.dk/app_dial.c.diff It works, but... I can't figure out setting/reading an option. It looks fairly easy, but the flag is always set. *** app_dial.c.org 2009-11-04 22:15:50.000000000 +0100 --- app_dial.c 2009-12-01 09:29:19.000000000 +0100

When phone A registers, I want phone B to ring, when picked up, it should call phone A and connect the phones. Translated: When GF in Mexico powers up laptop where soft iax-phone registers automatically, I want to talk to her asap :-) How to? Leif

Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service....)

Had my Asterisk instance stop responding to incoming/outgoing calls today. Had to kill -9 the asterisk process and restart it to get it back. Not really looking for help on that as the instance is version 1.6 and is due to be replaced with an upgraded version shortly. However, this does make me wonder, do you restart periodically to try to avoid issues or do you just let things run until

Just a Question. I would like to know if TDMoE follows specifiaciones of TDMoIP RAD protocol that says that there is a compression of 16/1 when you do TDMoIP. Manuel Marin Garcia TRANSTELCO S.A. DE C.V. Campos Eliseos 9050 B4 – Cd. Juárez, Chih. 32452 - México Oficina: +52 656 692 11 09 – Fax: +52 656 692 1112 - Celular: 915 727 6141 http://www.transtelco.com.mx

In relation to our CRM-system I'd like to send a query to asterisk who is extension xxx talking to. When the operator enters the page with customer data, the crm should send a query to asterisk, to get the cli of the call the operator is having. If the number is matching the customers number in crm, a record will be made, if it is not, a popup "Are you talking with this customer

Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I