[Asterisk-Users] SIP phone hangs after some hours

Wed Sep 24 05:14:10 MST 2003

Hi,

	I have a problem with sip.conf. After some hours my sip
phone(netergy) hangs. In clonse appears the next logs repeatly:

10 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.168.0.155 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK5ffceb80
From: "asterisk" <sip:asterisk at 192.168.0.207>;tag=as4b104f64
To: <sip:192.168.0.155>
Contact: <sip:asterisk at 192.168.0.207>
Call-ID: 116ecf3f37ce53b1275f2fec48a95375 at 192.168.0.207
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 192.168.0.155:5060
Sip read: 
SIP/2.0 200 OK
Call-ID: 116ecf3f37ce53b1275f2fec48a95375 at 192.168.0.207
From: asterisk<sip:asterisk at 192.168.0.207>;tag=as4b104f64
To: sip:192.168.0.155
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK5ffceb80
Supported: timer,100rel
Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS,PRACK
Accept: application/sdp
Accept-Encoding:  
Accept-Language: en;q=0.8
User-Agent: Netergy MicroElectronics
Content-Length: 0

13 headers, 0 lines
DEBUG[12301]: File chan_sip.c, Line 533 (__sip_ack): Stopping
retransmission on '116ecf3f37ce53b1275f2fec48a95375 at 192.168.0.207' of
Request 102: Found
DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call
'116ecf3f37ce53b1275f2fec48a95375 at 192.168.0.207'

My sip.conf is the next:

;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = 192.168.0.207                ; Address to bind to
context = outgoing              ; Default for incoming calls
disallow=all
allow=alaw
tos=lowdelay
maxexpirey=100000               ; Max length of incoming registration we
allow
defaultexpirey=100000           ; Default length of incoming/outoing
registration

[705]
type=friend
username=705
host=192.168.0.155
dtmfmode=inband
mailbox=705
callerid=705
context=outgoing
reinvite=yes
canreinvite=no
qualify=yes
nat=-1

My sip phone doesn't  register in asterisk due to my decision.

I can send and receive call, but if phones is inactive during some hours
it hangs. It is due to asterisk or my sip phone?

Any idea?

Thanks,

srsergio