similar to: Dead FXO Module on TDM400P?

Hello folks, I'm very interested in the Digium/Asterisk combination but need some clarification. I would like to setup a SOHO for business and home use. Scenario One: I have one analog line, 4 analog telephones. Do I need a TDM400P + 4 FXS modules (Green) + X100P? Scenario Two: 2 analog lines, 1 selective ring number for fax, 8 analog phones. Is this what I need? 2 TDM400Ps and 8 FXS

Hello list... 1) Is there an IAX softphone that supports any kind of voicemail indicator? 2) I have 2 TDM400Ps installed in a system. I need the audio on the analog phone (FXS modules) to be amplified somewhere between 10 and 15db so I set rxgain and txgain to 15 for the FXO interfaces and FXS interfaces. When a call comes in on the FXO at this setting, the call sometimes has about 20 seconds of

Hi all, Theoretically, I know it's possible, but is any using multiple tdm400ps (fxo) in single * box? In a production environment? Any gotchas aside form irq sharing? Thanks Ryan

I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot

Hello, I have found a "bug", I think in the way TDM400P cards handle FXO interface disconnect/re-connect problems. Normally I do keep all the wires connected from my CO / PABX quite securely, but I had a need to re-route the cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO

List members, Is it possible to record outbound analog calls using an X100P? I was asked if I knew how to record all calls for a shop with 4 analog phones transparently to the end users. I thought Asterisk was a good fit for this and I envisioned using either Digium TDM400Ps or Sangoma A200s with 4 FXO and 4 FXS modules. The FXO modules would be connected to the existing PBX and the FXS

Hi all, haven't seen many posts about asterisk in Singapore... Getting a server going there and was wondering if TDM400Ps will be fine in FCC mode, and if there are indications / cadence values that I should be putting on there as in other international locations. Seen an unsettling post on voip-info about Singapore issues with Call Polarity/Hangup Detection -- crossing my fingers I

Someone mentioned earlier (I can't find the message now) that they had a motherboard that allowed you to change IRQ assignments in BIOS. Does anyone happen to know how to identify motherboards that can do this? I'm going to put together a new machine now and I'm having trouble picking a motherboard for it (ordering from Dell or other online vendor is not an option, since I need

Hi, it took me 2 days to get my asterisk box running, so now I completed and I am disappointed of the sound quality. When I call other people their voices sound somewhat scratchy. First I thought it might be a codec problem, but I also recognized it during the ring tone or even the DISA connect tone. Sometimes it is better quality and sometimes more scratchy. Where might be the problem? I am

Hello Asterisk Community! There have been some complaints made by those customers that purchased the TDM400P board and it didn't work properly in their boxes. Digium promised to swap such boards for the new - revised version and will keep the promise. However since we were backordered we're currently shipping boards to the customers that paid and are waiting. But if you want to receive

asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were "scratchy" and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. Is this a problem at the carrier? I'm trying to call them now, but it's Sunday morning in the sticks, and my chances of

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to

Hey all - we have two TDM400P cards in an SMP Redhat9 box, with 4 FXO ports each running thel latest Asterisk CVS. Users connect to this to share POTS lines using analog phones connected to Sipura SPA-2000 boxes. All works reasonably well, except every day (or more) a line will get "locked up". No other way to describe it really -- the line just stops working, so if someone tries

Hello. I've seen several posts talking about line quality using Digium cards that are sharing IRQs or on machines where X is running but after trying all of those fixes I am still having a problem with line static on outoing calls. BTW, calls that are from one extension to another extension have no static, however, they have occasional clicks and pops. At any rate, I was wondering if

Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of

Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? Thanks, - Darren

Hi all! It's great to start with "for dummies" question, but hey, we all have been human infants also =) Problem is, that I can not log on to this channel and I haven't found anything helpful during the past few days either. 1. The irc.freenode.net server gives me "Couldn't look up your hostname" and "No identd (auth) response" followed with

Hello Folks, I noticed that all of the IaxClient based softphones with exception of Firefly only seem to have support for GSM but what about iLBC? The quality is excellent with iLBC even on a dialup connection! Meanwhile while the audio on GSM often sounds scratchy. Is anyone looking to implement iLBC in an IaxClient based softphone soon? Errol Biz4Web Solutions Limited

For those that are interested, here is my report back to Sayson on the 480i ---------- Forwarded message ---------- Date: Sat, 31 Jul 2004 22:03:31 -0400 (EDT) From: xxx@bgcfreedom.com To: xxx@sayson.com Subject: 480i User Feedback With Asterisk Seshu, I am using a 480i, and I am impressed with the phone on a whole, but obviously the firmware is lacking. Details follow. Hold button works, but

Hi, I'm sorry to bring this up again, but I have been googling forever and whatever solutions are offered don't work for me. I am using x-lite (the latest build) and trying to use Speex. When I do call from the x-lite to another SIP phone or PSTN (through Cisco gateway) My asterisk fills up with this message: WARNING[1007]: codec_speex.c:196 speextolin_framein: Out of buffer space The