Sir Peter Krassmann
2004-Nov-09 13:04 UTC
[Asterisk-Users] Segmentation fault on SIP inbound
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@217.10.66.11>;tag=as4eca35df To: <sip:#myPHONENO#@sipgate.net> Contact: <sip:0@217.10.66.11> Call-ID: 41267d7c3292d6685753c0cd5166dd26@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:~# _________________________________________________________________ Messenger 6. 1 - lassen Sie alle B?rokollegen an Ihren genialen Ideen teilhaben. http://messenger.msn.at?DI=1031&XAPID=2532