[Asterisk-Users] Known problem?

[Linus Surguy]

Haha. Last time I discovered my libc or something wasnt quite there so I
couldnt compile chan_enum or something, so I didnt bother and carried on
with the older version (only from a couple of months back)

Linus

----- Original Message -----
From: "Jeremy McNamara" <jj at nufone.net>
To: <asterisk-users at lists.digium.com>
Sent: Monday, August 11, 2003 2:53 PM
Subject: Re: [Asterisk-Users] Known problem?


>
> 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 #
> >
> >
> >
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>