[Asterisk-Users] Known problem?

[Jeremy McNamara]

what hassles?    cvs update


Jeremy McNamara


Linus Surguy wrote:

>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 #
>
>
>
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