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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, 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

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

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'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. Ideas, anyone? Leif

...is a very low volume service as you can imagine so i would prefer something with low or no minimums and a pay-as-you-go scheme. An API would be a plus, although not necessary (i.e we can handle web forms). Anyone using or know a service that would like to recommend ? Thanks in advance Stelios S. Koroneos P.S Sorry for the cross-post

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:

Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the

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 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 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

...st make sure phones that need specific > > features fail to the same server (i.e. hinting and such) as those > > features don't work cross server. > > > > Aaron > > > > On Tue, 2006-11-14 at 08:16 -0700, David Thomas wrote: > > > On 11/14/06, Stelios Koroneos <skoroneos@digital-opsis.com> wrote: > > > > JR Richardson gave a very nice presentation at Astricon on how to do > that with DUNDI > > > > > > As I understand it JR Richardson's DUNDi solution does not support > > > IAX. It uses regcontex which...

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:

All, I'm trying to send an SMS directly from asterisk but it doesn't seem to be working. The SMS() function does create an outgoing file but doesn't deliver the SMS. Can anyone help me to understand how SMS() works. Thanks. extensions.conf example: same => n,SMS(hello,a,17654307001,"hello nick") - nick

On 29 Jan 2015, at 11:07, Administrator TOOTAI wrote: > Le 28/01/2015 22:03, Steven McCann a ?crit : >> Hello, > > Hi > >> >> I'm investigating a situation where there was a hundreds of minutes >> of >> calls from an internal SIP extension to an 855 number in Cambodia, >> resulting in a crazy ($25,000+) bill from the phone company. I'm

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

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

Greetings to all. I am not sure of this is a "user" question or a "business" so apologies in advance if it should be asked in the business list. A client of mine has a UK branch that is served by a provider that uses the Broadsoft solution. I want to create a sip trunk from a remote asterisk pbx to the client. The provider claims that Broadsoft has a strict