asterisk users - Oct 2005 - strange behavior after turning jitter buffer on

This is with asterisk 1.20beta1:

I was experiencing moments of sporadic silence, so I thought to turn on 
the jitter buffer in iax.conf.
I started with the following settings, which are basically ripped from 
the sample config:
jitterbuffer=yes
forcejitterbuffer=no
maxjitterbuffer=1000
resyncthreshold=1000
maxjitterinterps=10

After doing this I encountered something a little different.  When I 
call my cell phone from a SIP phone (via asterisk and an IAX 
connection); the cell rings, I answer it, the cell claims it is 
connected, but I continue to hear ringing on the SIP phone until the 
Dial application times out (45 secs).  I don't see anything bad 
happening in the log except that chan_iax2 seems to think that no one 
has answered.  To make it more interesing, this doesn't happen on every 
call.

Excerpt from the log file detailing the call is below (with my actual 
cell phone number and teliax username obscured).

Does anyone have any thoughts on the matter?





Log stuff:

Oct 18 13:52:12 DEBUG[12842] chan_sip.c: * SIP extension value: 0 for 
call e0dae
938-16d63cb6@168.215.99.200
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: Setting NAT on RTP to 0
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: Stopping retransmission on 
'e0dae938-16
d63cb6@168.215.99.200' of Response 101: Match Found
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: * SIP extension value: 0 for 
call e0dae
938-16d63cb6@168.215.99.200
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: Setting NAT on RTP to 0
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: Checking SIP call limits for 
device PLX
Fax
Oct 18 13:52:12 DEBUG[12842] chan_sip.c: build_route: Contact hop: 
<sip:PLXFax@1
68.215.99.200:5060>
Oct 18 13:52:12 VERBOSE[12842] logger.c:     -- Executing 
SetCallerID("SIP/PLXFa
x-ceb5", "8667594678 |a") in new stack
Oct 18 13:52:12 VERBOSE[12842] logger.c:     -- Executing 
Dial("SIP/PLXFax-ceb5"
, "IAX2/username@teliax/16079999999|45|Tr") in new stack
Oct 18 13:52:12 VERBOSE[12842] logger.c:     -- Called 
username@teliax/160799999
99
Oct 18 13:52:12 VERBOSE[12842] logger.c:     -- Call accepted by 
208.139.204.245
 (format ulaw)
Oct 18 13:52:12 VERBOSE[12842] logger.c:     -- Format for call is ulaw
Oct 18 13:52:19 VERBOSE[12842] logger.c:     -- IAX2/teliax-1 is making 
progress
 passing it to SIP/PLXFax-ceb5
Oct 18 13:52:19 DEBUG[12842] chan_iax2.c: Ooh, voice format changed to 4
Oct 18 13:52:37 DEBUG[12842] chan_iax2.c: Immediately destroying 1, 
having recei
ved hangup
Oct 18 13:52:37 DEBUG[12842] chan_iax2.c: We're hanging up IAX2/teliax-1 
now...
Oct 18 13:52:37 DEBUG[12842] chan_iax2.c: Really destroying 
IAX2/teliax-1 now...
Oct 18 13:52:37 VERBOSE[12842] logger.c:     -- Hungup 'IAX2/teliax-1'
Oct 18 13:52:37 VERBOSE[12842] logger.c:   == No one is available to 
answer at t
his time (1:0/0/0)
Oct 18 13:52:37 DEBUG[12842] app_dial.c: Exiting with DIALSTATUS=NOANSWER.
Oct 18 13:52:37 DEBUG[12842] cdr_addon_mysql.c: cdr_mysql: inserting a 
CDR recor
d.
Oct 18 13:52:37 DEBUG[12842] cdr_addon_mysql.c: cdr_mysql: SQL command 
as follow
s:  INSERT INTO cdr 
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,l
astdata,duration,billsec,disposition,amaflags,accountcode) VALUES 
('2005-10-18 1
3:52:12','8667594678','8667594678','9999999','default',
'SIP/PLXFax-ceb5','IAX2/
teliax-1','Dial','IAX2/username@teliax/16079999999|45|Tr',25,0,'NO
ANSWER',3,'')
Oct 18 13:52:37 DEBUG[12842] chan_sip.c: update_user_counter(PLXFax) - 
decrement
 inUse counter
Oct 18 13:52:37 DEBUG[12842] chan_sip.c: Stopping retransmission on 
'e0dae938-16
d63cb6@168.215.99.200' of Response 102: Match Found

To avoid any confusion, you may note that the Dial Application does not 
time out in this log excerpt as I described.  That's because I hung up 
the cell phone instead of waiting for the timeout.

And before anyone asks, setting jitterbuffer=off made the problem go away.