asterisk users - Aug 2005 - [Asterisk-Dev] SIP channels not cleared

Hello all,

When I do 'sip show channels' I have seen a lot of
entries where these calls has already been terminated.
Some of these channels are bolong to calls being made
2 days ago but still showing from the CLI. They look
like

10.223.51.173    0022676583  130b36625fc  00102/00103 
 unknow(d)	Rx: BYE
10.223.51.173    0022676583  5533069e578  00102/00103 
 unknow(d)	Rx: BYE
10.223.51.173    0016513973  234f7bba140  00102/00103 
 unknow(d)	Rx: BYE
10.223.51.173    0027226765  487b770b231  00102/00103 
 unknow(d)	Rx: BYE
10.223.51.173    0016513973  69b59aa2084  00102/00103 
 unknow(d)	Rx: BYE
10.223.51.173    0199820127  60ef984904a  00102/00103 
 unknow(d)	Rx: BYE
10.223.51.173    0081805135  45bf3e8c287  00102/00103 
 unknow(d)	Rx: BYE

I have thousands of them in 'sip show channels' and is
increasing but it only shows 50 calls in 'show
channels'. I believe this eats up memory. Sooner or
later my system will run out of memory or get the 'Too
many file opened' error. 

I have made a sip trace on asterisk and seems like
they all share a same SIP message flow. When asterisk
send an INVITE to other sip server say B. B will reply
with  Trying. When B found out that the actual
destination can not be reached, it sends a BYE to
asterisk. Asterisk then reply with a 200 OK. Call is
hangup succesfully but 'sip show channels' still list
the call record and never go away untill asterisk is
restart. See below:


Aug 15 18:35:32 VERBOSE[12402] logger.c: Reliably
Transmitting (no NAT) to 10.223.51.173:5060:
INVITE sip:0377847785@10.223.51.173 SIP/2.0^M
Via: SIP/2.0/UDP
10.21.99.221:5060;branch=z9hG4bK6caf7db4^M
From: "DADAS"
<sip:301999@10.21.99.221>;tag=as64c4813c^M
To: <sip:0377847785@10.223.51.173>^M
Contact: <sip:301999@10.21.99.221>^M
Call-ID:
3e9c58780a742c244152a5b3433a9db2@10.21.99.221^M
CSeq: 102 INVITE^M
User-Agent: Asterisk PBX^M
Date: Mon, 15 Aug 2005 10:35:32 GMT^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
NOTIFY^M
Content-Type: application/sdp^M
Content-Length: 160^M
^M
v=0^M
o=root 12402 12402 IN IP4 10.21.99.221^M
s=session^M
c=IN IP4 10.21.99.221^M
t=0 0^M
m=audio 10986 RTP/AVP 8^M
a=rtpmap:8 PCMA/8000^M
a=silenceSupp:off - - - -^M



Aug 15 18:35:32 VERBOSE[15229] logger.c:
<-- SIP read from 10.223.51.173:5060:
SIP/2.0 100 Trying
Call-Id: 3e9c58780a742c244152a5b3433a9db2@10.21.99.221
CSeq: 102 INVITE
From: "DADAS" <sip:301999@10.21.99.221>;tag=as64c4813c
To: <sip:0377847785@10.223.51.173>
Via: SIP/2.0/UDP
10.21.99.221:5060;branch=z9hG4bK6caf7db4



Aug 15 18:35:39 VERBOSE[15229] logger.c:
<-- SIP read from 10.223.51.173:5060:
BYE sip:301999@10.21.99.221 SIP/2.0
Call-Id: 3e9c58780a742c244152a5b3433a9db2@10.21.99.221
Content-Length: 0
CSeq: 103 BYE
From:
<sip:0377847785@10.223.51.173>;tag=a10111834662596
To: "DADAS" <sip:301999@10.21.99.221>;tag=as64c4813c
Via: SIP/2.0/UDP 10.223.51.173;branch=z9hG4bK05f6ab33
Via: SIP/2.0/UDP
10.21.99.221:5060;branch=z9hG4bK6caf7db4



Aug 15 18:35:39 VERBOSE[15229] logger.c: Transmitting
(no NAT) to 10.223.51.173:5060:
SIP/2.0 200 OK^M
Via: SIP/2.0/UDP
10.223.51.173;branch=z9hG4bK05f6ab33^M
Via: SIP/2.0/UDP
10.21.99.221:5060;branch=z9hG4bK6caf7db4^M
From:
<sip:0377847785@10.223.51.173>;tag=a10111834662596^M
To: "DADAS" <sip:301999@10.21.99.221>;tag=as64c4813c^M
Call-ID:
3e9c58780a742c244152a5b3433a9db2@10.21.99.221^M
CSeq: 103 BYE^M
User-Agent: Asterisk PBX^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
NOTIFY^M
Contact: <sip:301999@10.21.99.221>^M
Content-Length: 0^M




The SIP message exchange seems to be comply to the
standard. Is this a bug in asterisk?

I have a system where there is always call going on
and I cant schedule asterisk to be restarted at any
time to clear the channels. 

Any idea?

I have CVS HEAD runnung on fedora 3.

Thanks

CCF






		
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