[Asterisk-Users] Known problem?

[Linus Surguy]

Hi all,

We're using an older version of *, built a couple of months ago and before
we go through all the hassle of updating source files and checking latest
dependancies on other kernels etc, I'd like to know if the following is a
known fault:

We're running a PSTN to FWD gateway in the UK and just whilst I was looking
at something else I noticed a call come in which caused Asterisk to simply
halt, terminating all processes.

I've got a SIP trace of the call, which is quoted below. Any ideas?


voip-gw1:/etc/asterisk # asterisk
voip-gw1:/etc/asterisk # asterisk -rvvv
  == Parsing '/etc/asterisk/asterisk.conf': Found
Asterisk 0.4.0, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer <markster at linux-support.net>
=========================================================================
Connected to Asterisk 0.4.0
 currently running on voip-gw1 (pid = 31349)
    -- Remote UNIX connection
voip-gw1*CLI> sip debug
SIP Debugging Enabled
voip-gw1*CLI> iax2 no debug
IAX2 Debugging Disabled
    -- B-channel 1 successfully restarted on span 1
    -- B-channel 2 successfully restarted on span 1
    -- B-channel 3 successfully restarted on span 1
    -- B-channel 4 successfully restarted on span 1
    -- B-channel 5 successfully restarted on span 1
    -- B-channel 6 successfully restarted on span 1
    -- B-channel 7 successfully restarted on span 1
    -- B-channel 8 successfully restarted on span 1
    -- B-channel 9 successfully restarted on span 1
    -- B-channel 10 successfully restarted on span 1
    -- B-channel 11 successfully restarted on span 1
    -- B-channel 12 successfully restarted on span 1
    -- B-channel 13 successfully restarted on span 1
    -- B-channel 14 successfully restarted on span 1
    -- B-channel 15 successfully restarted on span 1
    -- B-channel 17 successfully restarted on span 1
    -- B-channel 18 successfully restarted on span 1
    -- B-channel 19 successfully restarted on span 1
    -- B-channel 20 successfully restarted on span 1
    -- B-channel 21 successfully restarted on span 1
    -- B-channel 22 successfully restarted on span 1
    -- B-channel 23 successfully restarted on span 1
    -- B-channel 24 successfully restarted on span 1
    -- B-channel 25 successfully restarted on span 1
    -- B-channel 26 successfully restarted on span 1
    -- B-channel 27 successfully restarted on span 1
    -- B-channel 28 successfully restarted on span 1
    -- B-channel 29 successfully restarted on span 1
    -- B-channel 30 successfully restarted on span 1
    -- B-channel 31 successfully restarted on span 1
    -- Executing Dial("Zap/3-1", "Sip/38269 at fwd.pulver.com") in new stack
    -- Accepting call from '1189000000' to '099138269' on channel 3, span 1
Interface is eth0
IP Address is 213.166.5.129
We're at 213.166.5.129 port 2738
Answering with preferred capability 8
Answering with preferred capability 4
Answering with preferred capability 2
Answering with non-codec capability 1
10 headers, 11 lines
Reliably Transmitting:
INVITE sip:38269 at 192.246.69.223 SIP/2.0
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK3f7299f7
From: "1189000000" <sip:1189000000 at 213.166.5.129>;tag=as5770a04f
To: <sip:38269 at 192.246.69.223>
Contact: <sip:1189000000 at 213.166.5.129>
Call-ID: 65b9101a1e70a8253e59d8e31235728f at 213.166.5.129
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 31365 31365 IN IP4 213.166.5.129
s=session
c=IN IP4 213.166.5.129
t=0 0
m=audio 2738 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 (no NAT) to 192.246.69.223:5060
    -- Called 38269 at fwd.pulver.com
Sip read: LI>
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK3f7299f7
From: "1189000000" <sip:1189000000 at 213.166.5.129>;tag=as5770a04f
To: <sip:38269 at 192.246.69.223>
Call-ID: 65b9101a1e70a8253e59d8e31235728f at 213.166.5.129
CSeq: 102 INVITE
Server: Free World Dialup (0.8.11pre31 (i386/linux))
Content-Length: 0


8 headers, 0 lines
Sip read: LI>
SIP/2.0 302 MovedTemporarily
Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bK3f7299f7
Call-ID: 65b9101a1e70a8253e59d8e31235728f at 213.166.5.129
From: <sip:1189000000 at 213.166.5.129>;tag=as5770a04f
To: edc-soft <sip:38269 at 192.246.69.223>;tag=16f2d190
CSeq: 102 INVITE
Contact: <sip:38269 at 81.103.138.84:5062>;q=1.000
User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13
Content-Length: 0


9 headers, 0 lines
    -- Got SIP response 302 "MovedTemporarily" back from 192.246.69.223
Transmitting:
ACK sip:38269 at 192.246.69.223 SIP/2.0
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK3f7299f7
From: "1189000000" <sip:1189000000 at 213.166.5.129>;tag=as5770a04f
To: <sip:38269 at 192.246.69.223>;tag=16f2d190
Contact: <sip:1189000000 at 213.166.5.129>
Call-ID: 65b9101a1e70a8253e59d8e31235728f at 213.166.5.129
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.246.69.223:5060
    -- Now forwarding Zap/3-1 to '38269 at default' (thanks to
SIP/fwd.pulver.com-c473)
voip-gw1*CLI>
Disconnected from Asterisk server
voip-gw1:/etc/asterisk #