[Asterisk-Users] BudgeTone NAT issues

[John Todd]

>Just for the record and to possibly help with others who get BudgeTone phones.
>
>My asterisk box is behind NAT, and I use Vonage, NuFone, and 
>iconnecthere for my "POTS backhaul."
>
>On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102.
>
>The BudgeTone definitely has issues wrt the RTP stream and NATting, 
>although unfortunately I haven't yet been able to dig deeply enough 
>to be greatly more specific.
>
>However, I do know that:
>
>Using NuFone as backhaul, the BudgeTone works fine except in the 
>cases of the called party's line being busy.  In this case, I don't 
>get any busy indication, and eventually the call "times out" at the 
>NuFone end, all the while giving the user at the instrument a ring 
>indication.

I am uncertain that this particular problem you describe is a 
NuFone-specific problem.  I have seen similar problems with my own 
IAX2 connections which don't involve NuFone, and I've had 
circumstances where calls fail at the PSTN side but IAX2 and/or SIP 
don't get the message, and continues to "ring" in my ear despite the 
Zap line on the other end having hung up.  I am currently swamped 
with bug tasks, so I suspect it will be some time before I narrow 
this problem down and submit an official report with full diagnosis. 
Anyone else is welcome to the task.  :)

JT


>Using the BudgeTone with iconnecthere, outgoing calls set up just 
>fine, and I get call progress messages from asterisk.  The 
>connection dies immediately the called party picks up, with a SIP 
>"486 Temporarily Not Available" error.
>
>The phone appears to work mostly normally with Vonage, EXCEPT it 
>appears that the RTP stream has to have something "fed into it" at 
>the calling end; i.e. the called party cannot hear me, nor can I 
>hear the other end, until *I* have said something once the call 
>picks up.  Then conversation proceeds normally.
>
>I don't have these issues with either the ATA186 nor the TDM20.
>
>FYI.
>
>B.