I have been weighing the pros and cons of internal card FXO gateways like the Wildcard TDM400P and external FXO gateways like the Sipura SPA-3000, Grandstream HT-488, and the ever growing list of companies that are starting to make them.
Can anyone comment on them on these points:
Audio quality: Is the audio quality the same for the TDM400P compared to the SPA-3000 or other gateways?
Power failover: My main beef about the TDM400P is that it has no power failover capability to connect the FXS port to the FXO when the power is off while the external gateways do. Can you recommend external boxes that connect to the TDM400P to do this kind of relay switching?
Pass-through Fax/T.38 Fax: The TDM400P has the great benefit of having a developmental fax engine and T.38 through SpanDSP while some external gateways will support both pass-through and T.38 and others will support only pass-through. Unfortunately, the SPA-3000 does not support T.38.
Scalability: The Wildcard family scales from FXO lines to T1/E1 while the low cost external gateways only offer FXO lines. (Of course the higher end gateways scale up as well.) You also have a spaghetti of power cords when you set up more than a couple of FXO lines.
Adjustible FXO impedance: The TDM400P seems to have adjustible impedance from the country setting of the Zaptel driver. I have only seen the SPA-3000 support adjustible impedance settings for external gateways. This is important to match the impedance standards of different countries and possibly variations with PSTN lines (?) otherwise you will get echo.
To me, the closest thing to a best solution would be for Digium to release a board that includes FXS-to-FXO power failover.
I find it hard to recommend the TDM4xxP FXO ports to anyone:
- They ‘blind dial’ whether there is a phone line actually present or not or whether the line is ‘in-use’ already. So, if your phone line is down/broken/disconnected, this card happily dials, waits a few seconds, and then declares “Connected”. Yeah, right.
This is an absolute deal breaker for me. If there is no phone line, don’t dial!
Even the lowly $9 X100P clone cards report CHANUNAVAIL if the PSTN line is disconnected or in use.
The volume is unusably low. yes, I’ve tried the ‘rxgain’. It makes a minor difference, and as I keep making the value larger, eventually no sound passes any more. I call the PSTN number connected to the FXO port and SHOUT a voicemail message. The message is barely audible when played back despite the VM prompts being perfectly fine volume.
The call progress on outgoing calls appears to be: dial. wait a few seconds, declare ‘connected’
They have no idea when to hang up if the remote end of the PSTN network hangs up.
But, don’t spend money on the Grandstream HT-488, either. That was my second attempt at a working FXO port for my Asterisk:
No caller ID on incoming PSTN phone calls! An absolute deal breaker.
After a configurable number of rings on the PSTN line, the FXO port goes OFF-HOOK and then notifies Asterisk of the call. Nope. An FXO port should indicate ringing back to Asterisk and let it use ‘Answer’ to actually go off-hook.
Dialing outbound on the FXO port is a pain. Unlike a real FXO port, the phone number used in the SIP request is ignored. The HT488 reports ‘connected’ back to the Asterisk as soon as it dials any number. Then, you have to get Asterisk to send DTMF tones to actually get the HT488 to dial. Fortunately, Asterisk handles this well by using the “D” flag to the dial command. Just don’t forget a few ‘w’ at the beginning of the number to wait before dialing.
The FXS port on the HT-488 does not do ‘early dial’.
When the PSTN line rings, the HT-488 immediately rings the analog phone that you would have thought was associated with the FXS port. This thing is not an FXS and FXO device. It appears to be two FXS ports with a major hack to the PSTN port.
Unlike the HT-486, the “LAN” ethernet port on the device is always NAT’d behind the “WAN” port. The HT-486 lets you configure if you want the LAN port NAT’d to the WAN port of if you want it simply switched(bridged) to the WAN port.
on the plus side, the HT-488 does report “BUSY” if the PSTN line does not appear to be connected and off-hook. Ideally it would return CHANUNAVAIL in the case of no phone line connected at all, but I’ll take this.
on the plus side, The HT-488 allows selecting a few numbers that can be dialed on the analog handset which immediately go out the PSTN port. 911 (or 999 or 112 or whatever your country uses) is an obvious one. (The HT-486 doesn’t have this. It only supports the “*00” (programmable) prefix. Don’t forget to tell your family that when they pick up that particular phone when the house is on fire, they should dial *00, wait for 2nd dial tone and then dial ‘911’ Yes, you can tell I’ve bought HT-486’s too) So, if you ignore the fake-FXO port, the HT-488 actually does what the HT-486 should have done.
Two coworkers use the SPA-3000, but the configuration options are overwhelming. And, nobody seems to have proven that the SPA doesn’t also try to ‘help’ by bridging FXO and FXS ports internally. In my case I want TWO SEPARATE PORTS. I don’t want any “help” from the ATA.
Rather than spending more money on equipment for now, I’m waiting for a coworker to get the SPA-3000 configured in a dumbed-down mode. It’s really geared to people without Asterisk. It tries to be it’s own mini PBX. That’s more than I need or want.
A thousand pardons to Digium for hosting a forum where I trashed their card, but everybody says this thing is the best solution, but it’s unusable for me.
i’ve used both recently, i purchased a SPA3K to try at home as i was having problems with getting a TDM02B to work in the UK.
needs, no, it really needs a PCI2.2 slot. i can’t say this enough. in a mobo i “thought” was PCI2.2 (Dell P4, not on the non-compat list) horrible echo that wasn’t consistent, in a PCI2.2 slot it’s perfectly behaved.
remote disconnect has been an issue. installing HEAD, making sure the correct settings for your location are loaded with wctdm, and getting zap config files right are the keys. before i got this right, then my TDM was reporting CHANUNAVAIL all too often.
like sjmike says, no PSTN pass-thru on powerfail
don’t believe anyone who says it’s 2 minutes to get it going with Asterisk. i spent what felt like days ! i tried all the config examples/tips i found out there, but none seemed to do what i wanted.
so now i have a mix of a few. it works very well, even getting CallerID and ringtones right. but i have copies of my config locked away safe as i don’t want to waste those hours configuring.
the PSTN pass-thru is useful for when you bork your Asterisk setup. pop the power and the POTS phone is back online.
summary … i would go for a TDM every time. sure, the SPA3K does (eventually) what you want it to, but once you have the TDM setup right it’s great. one less PSU/switchport to worry about too. and you’ll never have to face the scary config pages the SPA has !!!
if you plump for an SPA and you’re in the UK, i have one for sale. ready configured for BT lines and Asterisk too
I have a TDM411B rev.I and having trouble getting the zap config files right! Could you please post your config files?
Thanks a 1,000,000.