[Asterisk-Users] Segmentation fault on SIP inbound

[Sir Peter Krassmann]

Hi, I can dial out with SIP, but any inbound call causes a segmentation 
fault. Before recompiling asterisk, the segfault was preceded by a "Ouch.. 
cannot write to audio file" error message. Here are my settings/logs. Any 
help is greatly appreciated...

[sipgate]
type=friend
username=#myUSERID#
host=sipgate.de
fromuser=#myUSERID#
fromdomain=sipgate.de
nat=no
context=from-sip
canreinvite=no

[from-sip]
exten => _.,1,Wait(5)
exten => _.,2,Answer
exten => _.,3,Voicemail,s100
exten => _.,4,Hangup

INVITE sip:#myUSERID#@#myIP#:5060 SIP/2.0
Record-Route: <sip:#myUSERID#@217.10.79.9;ftag=as4eca35df;lr=on>
Max-Forwards: 9
Record-Route: <sip:#myPHONENO#@217.10.79.8;ftag=as4eca35df;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKa96e.98b17d44.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKa96e.8d222c41.0
Via: SIP/2.0/UDP 217.10.66.11:5060;branch=z9hG4bK687d609c
From: "0" <sip:0 at 217.10.66.11>;tag=as4eca35df
To: <sip:#myPHONENO#@sipgate.net>
Contact: <sip:0 at 217.10.66.11>
Call-ID: 41267d7c3292d6685753c0cd5166dd26 at 217.10.66.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 09 Nov 2004 11:47:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 343
Sipgate-Authentication: accepted

v=0
o=root 9974 9974 IN IP4 217.10.66.11
s=session
c=IN IP4 217.10.79.9
t=0 0
m=audio 55426 RTP/AVP 8 0 3 10 97 18 2 5
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=direction:active
a=nortpproxy:yes

18 headers, 16 lines
Using latest request as basis request
Sending to 217.10.79.9 : 5060 (non-NAT)
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 10
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 5
Peer audio RTP is at port 217.10.79.9:55426
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format L16
Found description format iLBC
Found description format G729
Found description format G726-32
Found description format DVI4
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - 
audio=0x57e(GSM|ULAW|ALAW|G726|ADPCM|SLINR|G729A|ILBC)/video=0x0(EMPTY), 
combined - 0xe(GSM|ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 
0x0(EMPTY)
Found peer 'sipgate'
Segmentation fault
debian:~#

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