similar to: Putting a call on hold

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,

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 was looking for a GSM gateway, but I didn't find any prices... Anybody knows how much one of these costs? And, is it possible to use an amateur radio with asterisk? Thanks, Marconi.

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

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,

Hi all. I am trying following scenerio for call park & pickup. voice is flowing established between B & C, after call-pickup ( instead of A & B ). can anyone please clarify why it is happening like this, ( or ) do i need some more configuration for park&pickup ? A B

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

Hi, anyone out there with the same problems and a possible solution to the following? The functions callparking and hold use the same transferdigittimeout in features.conf. While I think 3 to 5 seconds are enough to let the user "find" their keys on the phone, the double ammount of time ( 2 x 5 secs) you have to wait before a call is parked and the parkposition is announced, is

Hey guys, Im about to sign up for VoicePulse Connect. Of course, I plan on using my asterisk server to "register =>" with the service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use

Hey guys, How much disk space is used by Asterisk to store voice mail for about 10 - 20 users/mailboxes Thank you, Steve Maroney

> -----Urspr?ngliche Nachricht----- > Von: Mojo with Horan & Company, LLC [mailto:mojo at horanappraisals.com] > Gesendet: Dienstag, 25. M?rz 2008 23:23 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [asterisk-users] Asterisk parking hold and > transferdigittimeout > > It seems that the dialplan comes into play. If your parking >

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

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.

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

We are going to replace a BizFone with Asterisk and one of the features that I haven't been able to emulate is the PBX hold. We can put a call into hold with the * key and every 30 sec the phone gets a beep to remind us it is there and the caller on hold can hit # for other options. The closest I've come with Asterisk is call parking, and that doesn't quite do it. Using the

I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext => 8540 parkpos => 8541-8555 context => parkedcalls parkingtime => 45 /etc/asterisk/extensions.conf include => parkedcalls include => default [default] exten => 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and

I have the standard parking.conf but extension 700 doesn't show up in my dialplan.... Why? I can dial 701 which tells me that I don't have any calls parked there. 700 just gives me invalid extension noise.... Should I have extension 700 defined elsewhere? Thanks parking.conf [general] parkext =a 700 ; What ext. to dial to park parkpos => 701-705