similar to: R: Sound Quality Problems

Thanks for reading this. I've been pulling hair for days trying to resolve this, and any help someone can give me would be very much appreciated. I have an Asterisk box that is basically a P4-3GHz, a Digium-recommended SuperMicro X5SSE-GM motherboard, 2GB RAM, 250GB IDE hard-drive with UDMA, a SoundBlaster Live! 24 sound card, a Digium Wildcard TE110P, and a Digium Wildcard TDM400P with 4

Hi, I have this error, I have a digium card TE110P Tiger3xx When I'm load the dirvers by this command modprobe wcte11xp I got this error "Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for wct1xxp " when I'm Lunch the command ztcfg -vv I got this error.

Hi, This error I got it just when I gonfigure zaptel support isdneuro 31 channels. But if I configure zaptel to support T1 and just 24 channels I have no problem. ######################################################################## #### # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = it span=1,1,0,ccs,hdb3,crc4 bchan=1-23 dchan=16

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

I've been scratching my head for a while and I expect it is my mediocre knowledge of Asterisk which is holding me back. If anyone can assist me with some pointers I'd be grateful. Basically, I've hooked up a Viking intercom at the front door. It hooks into an fxs as a "phone". Up till now I've just played back a "go away" message if any internal phones are

Hi, I a new user of asterisk, I'm trying to in install zaptel drivers on my ISDN card Digium Tiger 3xx TE110P. And my configuration is # # Zaptel Configuration File # span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = it ; ; ; Zapata Configuration file ; [channels] immediate=no switchtype=EuroISDN signalling=pri_cpe pridialplan=unknown context=incoming usecallerid=yes group=1 channel

Hi everybody, I have a Asterisk installation working with a Digi card (HFC-S PCI A ISDN card) and another Asterisk installation with a Digium (Digium Wildcard TE110P T1/E1 Card). For Digium I used zaptel drivers but Digi works with bristuff drivers. Both installations work fine and are connected to two different phone lines. What I want to do is let calls incoming on the Digi installation be

I had a problem with this card and 2.6.11 kernel. I am using FC3 but sticking with the 2.6.9 kernel. I had a lot of make warnings on the zaptel build and the card played up. It also wouldn't do a modprobe -r without crashing the system. With 2.6.9 zaptel compiles fine and I can unload the mod as and when. Also stay well away from the 2.6.12 FC3 kernel as it didn't work at all and

Hi all, I have asterisk 1.4.26 (and I tried 1.4.29) connected PRI all 23 lines and for the most part everything works. Dialing out on 23 lines to phones works fine. I have to use the Local channel to call the intercom system (from call files). If I only call 1 intercom system at a time so it uses DAHDI/1 everything seems to work as I can call all 8 intercom systems and play a message. The

Hello all, I'm having a problem with my intercom function that I use for under-chin paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's for our general phones. I have a global defined which has all the SIP channels concatenated together - this is ${ALL-PAGE-EXTS}. The problem comes when a user is on the line, and someone else uses the intercom function to page

So, after playing with 6.0 on the Cisco 7960 and 7940 platforms, I have the following gripes, which I've sent to a very clueful Cisco person already. Mind you, I love the Cisco 79xx series phones, and currently they are what I recommend to anyone who wants a 'real' IP phone. I just cringe - Speed dials. It's nice to now have speed dials in the line appearances that

Environment: Asterisk 1.2.14, FreePBX 2.2.0, Aastra 480i IP telephones firmware version 1.4.1.1077. Problem: Intercom feature: the dialed phone does not play the splash tone when auto-answering an intercom call. Otherwise, intercom works perfectly. Questions: What is the extensions.conf syntax to trigger a splash tone in Asterisk 1.2.14 (from the documentation and posts I've found, it has

Hi. I'm currently setting up a particular conference: 3 members (a,b,c), a can speak with b and c, b and c can speak only with a and not between them. I found my possible solution with paging/intercom using option "d" (full-duplex), but I need to make ringing the phone in intercom. Now I set auto-answer=6 but after first ring the phone hangup the call. There is a way to using

A friend pointed out this url http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists intercom/auto-answer as being a feature in Cisco Call Manager (which as I understand it, uses SIP predominately for handsets). I've come across comment somewhere that intercom isn't supported in the SIP spec. Does anyone know if the apparent capability of Intercom being available in SIP

There was a thread a few months ago that tossed around some ideas for using a cisco phone for intercom or paging. I don't have any ip phones, and wondered if anyone had any luck getting intercom or paging to work on the cisco units. Do any of the (cheaper) ip phones have a way to support intercom or paging? I presume that it's not part of the SIP or IAX protocols. Chris.

I'm having hard times with paging intercom Heres my dialplan exten => 777,1,Goto(intercom,777,1) [intercom] exten => 777,1,SIPAddHeader(Call-Info: <sip:192.168.16.105>\;answer-after=0) exten => 777,2,Page(Local/308 at page& Local/309 at page& Local/310 at page) [page] ; Paging context exten => _X.,1,Macro(page,SIP/${EXTEN}) [macro-page] ;

Hey all, Has anyone had any issues with phones having multiple lines that are in different contexts? We've got a couple phones that we're testing intercom functionality for, and I'm noticing that for some strange reason, no matter what line we use, the phones tend to be completely in one context or another, not segregated like I would expect. Our contexts look like this: context

Hi All... I'm trying to get the intercom feature working on some snom 190 phones but having no luck... As you can see from the SIP trace below (from the called phone), intercom=true is being appended to the To: header as per requirements. I've email'd snom a few days ago but have yet to get a response. On my 190s, im running snom190-SIP 3.57v. I am pulling the config for the

a phone system, and plug it into a SIP Adapter like the PAP2T. Never done it myself, so I can't recommend a suitable intercom. Hopefully s= omeone else can. Dan Journo Kesher Communications (UK) Business Phone Systems<http://www.keshercommunications.com/> | Hosted PBX<h= ttp://www.keshercommunications.com/hostedpbx.html>

Today I released Enhance Me 1.004 for AMP 1.10 and Asterisk @ Home 2.2. These utilities allow for speed dialing, a revised version which uses AMP to store speed dial numbers and NAMES so that only operators of the web interface can add and delete speed dials. Data is stored in mysql instead of Asterisks' DB. This stops the waste of the 300 series extension numbers. Instructions on