[Asterisk-Users] strange behavior after turning jitter buffer on

Adam Moffett
2005-10-18T11:16:43Z

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