similar to: GSM / Radio

Hi there, I''ve read the paper on Xen live migration, and it shows some very impressive figures, like 165ms downtime on a running web server, and 50ms for a quake3 server. I installed CentOS 5 on 2 servers, each with 2x Xeon E5335 (quad-core), 2x Intel 80003ES2LAN Gb NICs. Then I installed 2 DomUs, also with CentOS 5. One NIC is connected to the LAN (on the same switch and VLAN), the

I had this idea, and after looking for something like this already in progress, I found another guy who tried to start it... But I was unable to contact him, and his project seems to be dead. But, I believe it is possible, and I wanted to know the opinion of the experienced... So, let's go: I got an asterisk server setup to receive free calls from US to Brazil. The problem is that at my work,

Hi there, I got really tired of false hangups, specially when someone calls from SIP/IAX/whatever to PSTN, which makes no sense in using busydetect in the zap channel, as the caller will eventually hear the busy tones and hangup, causing the zap channel to be "freed" as well... After suggesting that the busy detect be enabled only for inbound calls, and finding other people that think

Hi, How do I put a call on hold? If i press # the music on hold plays to the other person, but asterisk asks for a number to transfer... I don't want to transfer, I simply want to put the person on hold, so he/she can hear the music while I do something, then get them off hold. Is it possible? The scenario: The person calls me from a SIP phone, and I receive the call in a regular PSTN phone,

Any one know of software that allows 2-way radios as VoIP(SIP) clients, besides dingotel's usb & mic cable trick ? http://www.dingotel.com/2way/requirements2way.asp They might be ok if the SIP client was not hardcode to their own SIP proxy Has anyone tried any hacks to get the 2-way radio SIP client to regsiter to a * box. hmm chan_frsgmfrs anyone? using the usb/mic cable under linux :)

Two days ago, I was talking on the phone from the FXO, to a SIP phone. After some time (like 1h30m), all of a sudden, there's a huge noise, like a buzz... Really loud. So I hungup, and called my asterisk box again... All I could hear was that sound. Someone called me from the internet, and as Asterisk dialed the FXO, all she heard was that noise too. So, I logged in my Asterisk server,

First, I am surprised at IAX... My asterisk server is behind a firewall, I am behind a firewall, and my iax client connected... Voodoo??? :) But, the firewall has one unblocked port, which I think I should set to the IAX, so the magic won't be necessary, and it makes sense to think that there will be less latency without the "magic layer". But, when I set port=myport in the

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

Hi, I tried to issue: xm save VMID FILE, and got the following error: Error: Error: [Failure instance: Traceback: twisted.internet.error.ConnectionRefusedError, Connection was refused by other side: 111: Connection refused. xm create works fine, xend is running on domain 0, I can log in the VM through xm console VMID and ssh without problems. I can also stop and restart the domain.

Hy, I have a network with WIFI communication and VHF/ HF channels. I have integrated asterisk in the network using SIP, ZAP and IAX2 channels for WIFI communications, but I don't Know How I could integrate the VHF/ HF channels. I have heard speaking about app_rpt project, but I don't Know very much about this. Could I integrate VHF/ HF channels with this application? if the answer is

Hello to the list, I have a question about Asterisk/app_rpt: When a signal is *not* being received and or transmitted by the radio system attached to Asterisk/app_rpt via its interface, is the incoming and or outgoing data suppressed (silence suppression)? In other words! At idle but still connected to some other device, what data will be transferred? I currently don't have access to a

Good morning, There is a new patch up on reviewboard[1] right now for the removal of app_rpt and chan_usbradio from Asterisk trunk. As it stands right now these two modules do not appear to be maintained in this repository and have out-of-date code. Russellb's patch will see these to modules removed from asterisk trunk (asterisk 11). If a large part of the community wishes to help

Hello All , In some ocasions i?m getting a problem with my X100P board. I?m trying to trace tre problem , but i didn?t find a possible answer. -> I get those messages when trying to use Zap Channel Sep 29 14:15:46 WARNING[-1094796368]: chan_sip.c:2107 sip_new: Unable to allocate channel structure Sep 29 14:15:46 NOTICE[-1094796368]: chan_sip.c:7283 handle_request: Unable to create/find

Pete VK2YX writes > I've been involved with IRLP for about 5 years and am one of the original > install team. I've gone through the emmotions of allowing "other" networks > connect to IRLP and I know its caused some lots of headache. > > As far as a closed network goes, yes there is LOTS of passion to keep it HAM > only and I'd have to support that notion.

Hi, I'm looking into support for SS7 and I found an article (http://www.openss7.com/news13022002.html) which says that OpenSS7 supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI interface cards. It also says that Linux Support Inc is the primary sponsor of Asterisk. However I cannot find these cards on the Asterisk hardware page

Hello, Could anybody point me where to find the internal structures of the COW devices in /dev/mapper? What I want to do, exactly, is: make a backup of the snapshot in an image file, reading all the used blocks, and leaving the "not-used" ones. Later, I want to get the image file, and restore the cow device. But I can''t find an LVM document describing exactly how the COW

Hi all, This my first mail in the community. My name is Manolo. I'm new in Asterisk. My objective is to control some radio using Asterisk via web. In google I could see that the application app_rpt had this goal, but currently is not in the new Asterisk 1.8 or higher. Thinking in future, I thought the easer way to do that is to buy some RoIP unit and do the communication with

Hi, all I have installed TDM400 with 1 FXS and 1 FXP ports. Now I am goig through documentation on how to configure it. It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one do I use? Can someone send me sample zaptel.conf file for Australia? This will save me some time and will be used as a working example. Thanks, Rudolf

Greetings, Where can I get the chan_echolink channel driver from? I've seen reference to it, but have yet to find a place to download/compile it. It is part of the app_rpt.so module... I am told, but do not see the source with app_rpt. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Whatever happened to the Zapatatelephony project? The last information there is from 2001 and a lot of new cards have been released since. Has GNU hardware development completely stopped?