I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is any registerable incoming volume from these lines. I've been running them at rxgain = 25 (zapata.conf) to make the audio audible, however this creates poor call quality issues (static and distortion) on most calls, and audio garble in voicemails. Fxotune fails for every line with "Could not fill input buffer" I've tried changing PCI slots, played with echo settings, and done everything else I can think of to make this card play nice to no avial. Anyone with solutions or ideas, your input will be greatfully appreciated. Thank you in advance. -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key At http://www.2mbit.com/~trelane/trelane.key Key fingerprint = B4C2 8083 648B 37A2 4CCE 61D3 16D6 995D 026F 20CF
On 6/4/06, Andrew D Kirch <trelane@trelane.net> wrote:> I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N > Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x > quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is > any registerable incoming volume from these lines. I've been running them > at rxgain = 25 (zapata.conf) to make the audio audible, however this > creates poor call quality issues (static and distortion) on most calls, > and audio garble in voicemails. Fxotune fails for every line with "Could > not fill input buffer" > I've tried changing PCI slots, played with echo settings, and done > everything else I can think of to make this card play nice to no avial. > > Anyone with solutions or ideas, your input will be greatfully appreciated.Hi, I'm using one of these cards with 3 quad modules (4 x FXO, 8 x FXS) and I didn't have to touch the rxgain/txgain (0.0 for both). This is how it should be set. If you need to adjust these settings, then you have another problem. I know you tried changing PCI slots, but have you looked at /proc/interrupts to see if the card is sharing an IRQ with another device? Regards, Gonzalo.
Andrew D Kirch wrote:> I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N > Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x > quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is > any registerable incoming volume from these lines. I've been running them > at rxgain = 25 (zapata.conf) to make the audio audible, however this > creates poor call quality issues (static and distortion) on most calls, > and audio garble in voicemails. Fxotune fails for every line with "Could > not fill input buffer" > I've tried changing PCI slots, played with echo settings, and done > everything else I can think of to make this card play nice to no avial. > > Anyone with solutions or ideas, your input will be greatfully appreciated.If you plug a normal telephone into the wall socket is the call super quiet? -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
I'm having a related issue with "could not fill input buffer" and submitted this bug report: http://bugs.digium.com/view.php?id=7264 Rather then fixing volume issues, though, I'm trying to fix major echo issues. [The card used to work fine. fxotune even fixed the echo the first time I tried it, but gets the error you've been getting since then (granted though I'm now using zaptel SVN instead of a patched 1.2.5).] I'm at the point now where I either get fxotune working or buy a Sangoma card with hardware echo cancellation. -Barry King Andrew D Kirch wrote:> I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N > Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x > quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is > any registerable incoming volume from these lines. I've been running them > at rxgain = 25 (zapata.conf) to make the audio audible, however this > creates poor call quality issues (static and distortion) on most calls, > and audio garble in voicemails. Fxotune fails for every line with "Could > not fill input buffer" > I've tried changing PCI slots, played with echo settings, and done > everything else I can think of to make this card play nice to no avial. > > Anyone with solutions or ideas, your input will be greatfully appreciated. > > Thank you in advance. > >
Andrew D Kirch wrote:> I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N > Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x > quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is > any registerable incoming volume from these lines. I've been running them > at rxgain = 25 (zapata.conf) to make the audio audible, however this > creates poor call quality issues (static and distortion) on most calls, > and audio garble in voicemails. Fxotune fails for every line with "Could > not fill input buffer" > I've tried changing PCI slots, played with echo settings, and done > everything else I can think of to make this card play nice to no avial. > > Anyone with solutions or ideas, your input will be greatfully appreciated.As others have already noted, start with rxgain & txgain set to 0. And, check the lines with an analog phone to ensure you don't have weak audio from the pstn line to start with. If you do have weak audio (as in a very long pstn line), you're not likely to get the TDM400 or TDM2400 card to compensate for the loss without seriously impacting the s/w echo canceler. For high loss pstn lines, ztmonitor will not provide any useful indication. If you approach the problem from a professional perspective, you would use a transmission test set to measure the loss of the pstn line by dialing into the central office milliwatt generator (no asterisk involvement). If that measured pstn loss is anything greater then about 7db to 10db, I'd contact your pstn provider to see if there is anything they can do to improve it. (Most US telco's can install repeaters in the central office that will boast the audio levels. Repeaters have been in use by telco's for at least 20 years, and are typically used on long rural pstn lines.) If the analog audio is reasonable (or your measured pstn loss is less then about 7db), then you've got something wrong with the TDM2400 installation. That "could" be something like a mis-wired TDM2400-to-pstn line connections, etc. It will have nothing at all to do with the pci bus, interrupts, etc.
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