I hope this is not too far off topic.

Environment: Astlinux 0.6.2 (Asterisk 1.4.21.2) using a Linksys PAP2T  
(firmware version 5.1.6(LS) which I think is the latest) to provide  
service to a fax machine. (I know I should upgrade my Astlinux. And I  
plan to once the 0.7 release comes along.)

I am setup to use g711 in both the ATA and in Asterisk. All calls are  
set for "no re-invite" so that all RTP streams go through the Astlinux  
box and my older SIP ignorant dual WAN router only has to deal with  
port mapping for the Asterisk box. Until recently faxes worked with  
sufficient reliability that I considered this to be a solved problem.  
But recently my VoIP provider put compression on the link they use  
between the end point I connect to and the PSTN. Result: Faxes no  
longer work.

My VoIP provider claims that I can setup T.38 with them on that  
number. From the literature it seems Asterisk 1.4.21.2 will perform T. 
38 pass through. But I am getting various results on looking up  
whether or not the PAP2T properly supports T.38. I don't see attempts  
by it to use T.38 in my SIP traces and at least one recent posting on  
a discussion forum indicates that what T.38 is there is broken.

So I am in the market for an ATA that properly supports T.38. The two  
that have my attention at present are the Grandstream HT502 and a  
Cisco ATA186.

Cisco ATA186: It seems to be bulletproof and my office has used one on  
a Cisco CallCenter setup to do faxes successfully for years (but I  
don't know if that is using T.38 or not and it is probably not a SIP  
provisioned device). But it is more expensive (minor nit) and my  
experience setting up a Cisco PIX has led me to be wary of the ease of  
setup issues. And, near as I can tell, you need to subscribe to a  
contract with Cisco to get firmware updates.

Grandstream HT502: Has router capability I don't need. Claims T.38  
support. Price is right. But a number of web forum postings make we  
wonder about hardware build quality and reliability.

What are the experiences of others on this list regarding these (or  
other) ATAs when used in an Asterisk SIP environment?

Thanks for putting up with the slightly off topic post!

--Tod


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