search for: dhadi

Hi. I want to use the new asterisk 1.4 with dahdi, but I would like to know the svn branches for the dahdi, so I can use them that way -- much easier to keep up with bug fixes, etc. Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com

Hello guys, I was wondering if it's possible to assign a dahdi channel to two diferent groups. Thanks Adolphe Cher-aime From my Iphone

Hi all, I'm getting compilation error as trying to install latest version of dahdi on CentOS box 5.9 which I now updated from 5.6. I also installed the dependencies but still not getting the clue to get install the driver. Listing down the errors below; CC [M]

...6 almost everything seems to be working. All my SIP lines seem to work and calls come in as expected and all my DAHDI lines work with incoming calls, but I can't make an outgoing call on a DAHDI line. The simplest line is: exten=> s,1, dial(DAHDI/1) which says it can't create a DHADI channel and gives an unknown error 0. As far as I can tell everything is OK, but clearly it's not and I don't understand why incoming calls work and outgoing calls don't. Dahdi is using a TDM400 with 4 FXO modules. The messages at boot time seem correct and "dahdi show channels...

Hello I have upgraded our asterisk box from zaptel to dhadi two weeks ago... Since, there has been quite a significant amount of echo when making a call. Only for the local outgoing call, the person on the other side doesn't hear any echo. This is with a TE-110P ISDN PRI card .. I've pretty much took the original zaptel configuration and used it...

...ed, would only work for typical calls that were bridged. They would not work for things like voicemail. The voip-info wiki suggests using a meetme conference that listens for "D", but I think using the feature is cleaner - when it works. So, to make the call bridged, I send it across a dhadi "local" span. My DAHDI configuration - system.conf: fxsks=1-8 dynamic=loc,1:0,31,0 dynamic=loc,1:1,31,0 bchan=9-23,25-39 dchan=24 bchan=40-54,56-70 dchan=55 Channels 1-8 are the analog lines to the Toshiba. I then set up two "E1" spans (they aren't physical spans, but...

...time the shared extension is dialed. I have followed the documentation [1] and followed through other threads I saw regarding this feature [2,3,4]. Any advice would be appreciated. Sincerely, Dennison Williams ; === ; = sla.conf ; === [line1] type=trunk device=SIP/240 ; The example show using a DHADI trunk, but I can use a sip line right? autocontext=line1 [station1] type=station trunk=line1 autocontext=sla_stations [2272](station) device=SIP/2272 [2282](station) device=SIP/2282 ; === ; = sip.conf snippet ; === [2272] type=friend host=dynamic dtmfmode=rfc2833 secret=nottherealpasswd context...

...using redfone fonebridge to have Pstn connectivity on my asterisk box. I can receive in coming calls however outgoing calls don't go to provider. It's seems it's a span config problem. Because in systemconf when I try to config span as follow span=1,0,2,ccs,hdb3,crc4 And start dhadi I have the following message Unable to config ....... Bad argument(22). Please any help Adolphe Cher-aime From my Iphone

Hi, I'm using an half T1 line on a asterisk (obviously :)) 1.6.2.4 system, up to recently everything was fine but we are starting to experience the call limitation of the line (15). So as to warn user of the problem i attached a vocal notification to the CONGESTION status after a Dial(), but it looks like it also catch other congestion case (maybe on the receiver side). Should i / Could i

Hi, I have a tdm22b (2 fxs / 2 fxo) When Asterisk is just started, outbound calls routing to fxo port, do not working with error: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) Inbound calls to fxo port work fine. After first inbound call, the outbound calls starts working. CentOS 5.4 asterisk 1.6.0.21-1 dahdi 2.2.1.-1 Can anybody help me to identify what is the

Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten => fax,1,Dial(DAHDI/1) ; the fax machine exten =>

Hy guys i am having so much hard time to setup asterisk on a virtual machine that i got , i just want to know if i really need to use Dahdi and libpri on a complete Digital PBX i just gonna use sip and iax. I will never use any kind of analog line on this machine. Wait for a feed back. Daniel Abreu.

I'm having trouble getting a xorcom set up. A large part of the problem is that the box is a _long_ way away and I can't get to/at it easily, so while I could probably fix this in a few hours if the machine were in front of me, I'm struggling over a slow unreliable laggy link. Ok, enough whining from me. I have a new Xorcom plugged into the usb of a Suse 11 machine I built Dahdi

...mes the call gets answered for 1 second, but actually the phone has not rang, it?s just the CDR, and asterisk hangup automatically, I cought the log of 1 call like this, I hope you can help me with this. My setup is : <vendor> ----SIP--? <Asterisk> ?----IAX2---? <Asterisk with Dhadi channels> Here: -- Executing [966505103150 at from-internal:1] Macro("SIP/100-b609f9c0", "user-callerid|SKIPTTL|") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/100-b609f9c0", "AMPUSER=100") in new stack -- Executing...

...quot;Calibri","sans-serif"; color:#1F497D'>As far as my personal experience i reformat my desktops = /fully, semi annually, and all servers get a facelift every other month ( new = glib for new freeswitch updates, new ZAP hardware ? then you need new zaptel.. = wait zaptel aka dhadi needs X, X needs Y.. and so on.. <o:p></o:p></span></p> <p class=3DMsoNormal><span = style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"; color:#1F497D'><o:p>&nbsp;</o:p></span></p> <p class=3D...

...ew RAM/ NEW CPU/ etc all these need software upgrades eventually.. As far as my personal experience i reformat my desktops /fully, semi annually, and all servers get a facelift every other month ( new glib for new freeswitch updates, new ZAP hardware ? then you need new zaptel.. wait zaptel aka dhadi needs X, X needs Y.. and so on.. Mike ContacTel.COM From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan Thurman Sent: May-20-09 7:33 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Step-by...

I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;