[asterisk-dev] [regression] Anonymous SIP calls are hung up after 900 seconds (Asterisk v. 1.6.0.18)

[Hans Petter Selasky]

Hi,

I have a problem where anonymous SIP calls, I.E. SIP calls without a 
registered username, start hanging up around 900 seconds (exactly).

This happened when the one end was upgraded from "1.6.0.9" to "1.6.0.18" while 
the other end was already running "1.6.0.18".

In chan_sip.c I find:

#define DEFAULT_EXPIRY 900                          /*!< Expire slowly */

Is this a known issue that has been fixed?

Software used at both ends:

FreeBSD-8 + AMD64 + Asterisk version: 1.6.0.18

--HPS

Messages exchanged around the hangup:

Peer A:

set_destination: Parsing <sip:xxxx> for address/port to send to
set_destination: set destination to xxx, port 5060
Audio is at xxx port 16554
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to XXXX:5060:
INVITE sip:XXXX at XXXX SIP/2.0
Via: SIP/2.0/UDP xxx:5060;branch=XXXX;rport
Max-Forwards: 70
From: <sip:@xxx>;tag=XXX
To: "Hans Petter Selasky" <sip:xxx>;tag=XXX
Contact: <sip:xxx at xxx>
Call-ID: xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.18
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 336

Answer back:

SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 
XXXX:5060;branch=XXXX;received=XXX;rport=5060
From: <sip:XXX at XXX>;tag=XXX
To: "Hans Petter Selasky" <sip:XXX>;tag=XXX
Call-ID: XXX
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16