asterisk users - Jun 2005 - Problems with IAX Trunks

I have two asterisk servers connected using IAX. Server A has a  
TE410P running on a Xeon 2.4Ghz with 2GB RAM and 36G IDE HD on Debian  
2.6.11-1-686 and Asterisk CVS-Nv1-0-7-06/01/05-01:27:25.

Server B does not have any Digium board, but has ztdummy and zaptel  
loaded. It's runnin on a P4 1.6Ghz with 1GB RAM and 36G SCSI RAID 10  
on Gentoo 2.6.11-gentoo-r9 and Asterisk 1.0.7.

The relevant section of iax.conf looks like:

[gateway0]
type=friend
user=gateway0
secret=guess
context=default
host=10.0.10.199
trunk=yes
notransfer=yes
canreinvite=no
disallow=all
allow=ulaw

When I dial from Server B thru Server A, I simply issue: Dial($ 
{GATEWA}/${EXTEN},,r), where ${GATEWAY} points to the IAX2 trunk  
information.

The problem I have is that every once in a while, people complained  
that voice quality gets really bad, even to the point that one party  
doesn't hear the other. This probably happens once or twice a day.  
What I did to resolve it, was simply to run 'restart now' on Server  
B, and that fixed the problem.

I am looking at the server today and I see that there is only two  
people on the phone. However, when I do show channels on Server B, it  
seems like there were 52 active channels, all of them showing  
outbound calls thru Server A and a similarly high count on Server A.

I guess what is happening is that the calls don't seem to be getting  
disconnected. I don't know if the actual leg to the PSTN is still  
open (and I'm being billed) or if it's simply the channel in the  
trunk between the two machines. How can I find out what is exactly  
happening? When I do the show channels in Server A, it does show the  
channels going out on Zap/g?, which leads me to think that I'm being  
billed for these calls which were disconnected a while ago.

Also, I think the fact that calls are not getting disconnected and  
keep the trunk open are the cause of the audio quality being reported  
and when doing a restart now, it simply terminates all those calls.

Is there something in the config I can change to fix this or should I  
upgrade to a newer CVS version? Help please.

Could it have something to do with ztdummy? I used to run Server B  
without ztdummy for about a week and I don't really recall getting  
the audio quality complaints. Of course, I haven't tested again  
Server B without ztdummy running.

Thanks,
Waldo

I posted this http://lists.digium.com/pipermail/asterisk-users/2005- 
June/111815.html and never received a response. I just wanted to  
share with you that I think I fixed the problem. The only thing I  
changed was my Dial command by removing the 'r' option. Since then,  
asterisk seems to properly discard all terminated calls. I don't know  
if it's a bug or expected with the 'r' option.

Thanks,
Waldo