I recently started playing around with Asterisk@Home. I bought a Digium OEM board FXO card on eBay for 6.99. It works okay, but are there any boards that provide a better sound quality then the Digium Wildcard X100P / X100P FXO PCI Card Asterisk PBX (cgi.ebay.com/ws/eBayISAPI.dll?Vi … eName=WDVW)
First off congrats on entering the asterisk community, I am sure you will find lots of help and very cool things to do with asterisk. However I do want you to realize the digium “oem” card you bought is not a Digium product, and has never had any stamp of approval, it is a clone card, a card that shares many properties and loads under the same driver, if you are looking for better quality, I would reccomend a tdm01b.
But anyways, your bad sound could be caused by shared interrupts, running x windows or running asterisk with out hdparm on your drives.
Try this store.yahoo.com/asteriskpbx/newitastdmde.html
The X100P was discontinued, not sure why but it could have been for quality problems.
Thanks…I am pretty impressed so far. I am already saving about $60 bucks on month in charges from Accessline (I had two business “virtual offices” from them).
I moved my home line (Zap line) over to the asterisk box this weekend. I have a couple polycom IP 500s and two Sipura 1001s. I need to add a Polycom IP600 though for my home office.
The tdm10b is worth the $125 bucks? In terms of quality?
Also had another question:
I want to split up my extensions for directory and to allow outbound calling through specific trunks.
For all of my “200” extensions (200, 201, 202, etc), I want to be part of my “home” context…and the primary outbound trunk should be the ZAP line. For all of my “400” extensions, I want to be a part of a context called 3g and the primary outbound trunk should be one of my SIP lines. And finally all of my “600” extensions should be a part of a context called perce and the primary outbound trunk should be the other SIP line.
How hard is this to setup?
[quote=“mogorman”]First off congrats on entering the asterisk community, I am sure you will find lots of help and very cool things to do with asterisk. However I do want you to realize the digium “oem” card you bought is not a Digium product, and has never had any stamp of approval, it is a clone card, a card that shares many properties and loads under the same driver, if you are looking for better quality, I would reccomend a tdm01b.
But anyways, your bad sound could be caused by shared interrupts, running x windows or running asterisk with out hdparm on your drives.[/quote]
I have only used the T100P but I am sure the quality on the tdm10b is good. Another alternative is a Handytone 488 store.yahoo.com/grandstream-netw … nphad.html
Regarding the outbound dialing you should read up on context and include. An example is below
[home]
include => ZapOut
[ZapOut]
exten => _9X.,1,Dial(ZAP/1/${EXTEN:1})
[3g]
include => SIPOut
[SIPOut]
exten => _9X.,1,Dial(SIP/${EXTEN}@sipprovider)
I have the cheap 100p ($6.99) and am not happy with the quality. Between rx and tx volume and echo there is no sufficient compromise.
I am not surprised, the wiki pretty much warns you away, but it was a low cost way to test the waters.
Thanks for bringing the ata-488 to my attention, this seems exactly what I want. The power fail-over is a killer feature.
Only question: Do you like the voice quality?
Anything else I should know?
Thanks,
Remco
The GS 488 may not be for prime time yet.
The sound quality was good but it had some quarks.
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For dialing out the FXO from a different extension, you dial it’s extension and get a dail-tone. That’s easy. Dialing the analog number isn’t. You have to configure your originating extension for inline DTMF which screws up voice mail if your not using the ulaw codec. In my case, I need to use a more compressed codec like ILBC for bandwith reasons.
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It rings the FXS phone four times before forwarding to an assigned extension and theres no way that I’ve found so far to change the number of rings. The caller hangs up before it gets forwarded to the desired extension. I need to change it to zero rings for my purposes.
Everyones configuration is different and it may work fine in yours, but for me, they need to work on the DTMS and and inward dialing.
I’m also using a GS 286 and two GS 101’s with complete satisfaction so the 488 limitations will probably get fixed with the next firmware release.
I pointed that out to their customer support. Maybe they need to hear this more often.
The 4 ring setup is really stupid.
I live with it for now, hoping for a firmware update.
the x100p and clones are no good, buy a tdm card. Or if you can get digital lines where you live, pick those.