[Jan 29 13:22:10] WARNING[31991] chan_oss.c: Unable to re-open DSP device /dev/dsp: No such file or directory [Jan 29 13:22:10] NOTICE[31991] console_video.c: voice only, console video support not present [Jan 29 13:22:10] DEBUG[31991] pbx.c: Launching 'Dial' [Jan 29 13:22:10] VERBOSE[31991] logger.c: -- Executing [655111308@oss:1] Dial("OSS/dsp", "SIP/peer-cdr/655111308") in new stack [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Asked to create a SIP channel with formats: 0x40 (slin) [Jan 29 13:22:10] VERBOSE[31991] logger.c: == Using SIP RTP CoS mark 5 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Setting NAT on RTP to Off [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jan 29 13:22:10] DEBUG[31991] acl.c: Found IP address for this socket [Jan 29 13:22:10] DEBUG[31991] frame.c: Could not find preferred codec - Going for the best codec [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: *** Our native formats are 0x40 (slin) [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: *** Joint capabilities are 0x40 (slin) [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Jan 29 13:22:10] DEBUG[31991] frame.c: Could not find preferred codec - Going for the best codec [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x40 (slin) [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: *** Our preferred formats from the incoming channel are 0x40 (slin) [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: This channel will not be able to handle video. [Jan 29 13:22:10] DEBUG[31991] rtp.c: Channel 'OSS/dsp' has no RTP, not doing anything [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Outgoing Call for 655111308 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Updating call counter for outgoing call [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False Text flag: False [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: ** Our prefcodec: 0x40 (slin) [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: This call needs video offers, but there's no video support enabled! [Jan 29 13:22:10] VERBOSE[31991] logger.c: Audio is at 88.92.0.111 port 10006 [Jan 29 13:22:10] VERBOSE[31991] logger.c: Adding codec 0x2 (gsm) to SDP [Jan 29 13:22:10] VERBOSE[31991] logger.c: Adding codec 0x4 (ulaw) to SDP [Jan 29 13:22:10] VERBOSE[31991] logger.c: Adding codec 0x8 (alaw) to SDP [Jan 29 13:22:10] VERBOSE[31991] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: -- Done with adding codecs to SDP [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263) [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Initializing initreq for method INVITE - callid 11d2544110f652792db04c613f7b19f3@mydomain.org [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 0 [ 48]: INVITE sip:655111308@mydomain.org SIP/2.0 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 88.92.0.111:5060;branch=z9hG4bK12777658;rport [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 3 [ 67]: From: "asterisk" ;tag=as6467dc28 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 4 [ 39]: To: [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 5 [ 41]: Contact: [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 6 [ 61]: Call-ID: 11d2544110f652792db04c613f7b19f3@mydomain.org [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 8 [ 30]: User-Agent: Asterisk SVN trunk [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 9 [ 35]: Date: Tue, 29 Jan 2008 12:22:10 GMT [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 10 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 13 [ 19]: Content-Length: 318 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 14 [ 0]: [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 1 [ 47]: o=root 2106297214 2106297214 IN IP4 88.92.0.111 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 2 [ 32]: s=Asterisk PBX SVN-trunk-r100795 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 3 [ 20]: c=IN IP4 88.92.0.111 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 5 [ 31]: m=audio 10006 RTP/AVP 3 0 8 101 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 6 [ 19]: a=rtpmap:3 GSM/8000 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Body 13 [ 10]: a=sendrecv [Jan 29 13:22:10] VERBOSE[31991] logger.c: Reliably Transmitting (no NAT) to 88.92.0.112:5060: INVITE sip:655111308@mydomain.org SIP/2.0 Via: SIP/2.0/UDP 88.92.0.111:5060;branch=z9hG4bK12777658;rport Max-Forwards: 70 From: "asterisk" ;tag=as6467dc28 To: Contact: Call-ID: 11d2544110f652792db04c613f7b19f3@mydomain.org CSeq: 102 INVITE User-Agent: Asterisk SVN trunk Date: Tue, 29 Jan 2008 12:22:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 318 v=0 o=root 2106297214 2106297214 IN IP4 88.92.0.111 s=Asterisk PBX SVN-trunk-r100795 c=IN IP4 88.92.0.111 t=0 0 m=audio 10006 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #8 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Trying to put 'INVITE sip' onto UDP socket... [Jan 29 13:22:10] VERBOSE[31991] logger.c: -- Called peer-cdr/655111308 [Jan 29 13:22:10] VERBOSE[31991] logger.c: <--- SIP read from UDP://88.92.0.112:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 88.92.0.111:5060;branch=z9hG4bK12777658;rport=5060 From: "asterisk" ;tag=as6467dc28 To: ;tag=7307a5a1b66dcf043b7a9005b56a7f91.f8f6 Call-ID: 11d2544110f652792db04c613f7b19f3@mydomain.org CSeq: 102 INVITE Proxy-Authenticate: Digest realm="mydomain.org", nonce="479f1b9e45cb05215439821aeafc420d6946b0a9", qop="auth" Server: OpenSER (1.4.0dev0-tls (i386/linux)) Content-Length: 0 <-------------> [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 0 [ 41]: SIP/2.0 407 Proxy Authentication Required [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.92.0.111:5060;branch=z9hG4bK12777658;rport=5060 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 2 [ 67]: From: "asterisk" ;tag=as6467dc28 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 3 [ 81]: To: ;tag=7307a5a1b66dcf043b7a9005b56a7f91.f8f6 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 4 [ 61]: Call-ID: 11d2544110f652792db04c613f7b19f3@mydomain.org [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 6 [116]: Proxy-Authenticate: Digest realm="mydomain.org", nonce="479f1b9e45cb05215439821aeafc420d6946b0a9", qop="auth" [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 7 [ 44]: Server: OpenSER (1.4.0dev0-tls (i386/linux)) [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Header 9 [ 0]: [Jan 29 13:22:10] VERBOSE[31991] logger.c: --- (9 headers 0 lines) --- [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Acked pending invite 102 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Stopping retransmission on '11d2544110f652792db04c613f7b19f3@mydomain.org' of Request 102: Match Found [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: SIP response 407 to standard invite [Jan 29 13:22:10] VERBOSE[31991] logger.c: Transmitting (no NAT) to 88.92.0.112:5060: ACK sip:655111308@mydomain.org SIP/2.0 Via: SIP/2.0/UDP 88.92.0.111:5060;branch=z9hG4bK12777658;rport Max-Forwards: 70 From: "asterisk" ;tag=as6467dc28 To: ;tag=7307a5a1b66dcf043b7a9005b56a7f91.f8f6 Contact: Call-ID: 11d2544110f652792db04c613f7b19f3@mydomain.org CSeq: 102 ACK User-Agent: Asterisk SVN trunk Content-Length: 0 --- [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Trying to put 'ACK sip:67' onto UDP socket... [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Auth attempt 1 on INVITE [Jan 29 13:22:10] NOTICE[31991] chan_sip.c: Failed to authenticate on INVITE to '"asterisk" ;tag=as6467dc28' [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Setting SIP_ALREADYGONE on dialog 11d2544110f652792db04c613f7b19f3@mydomain.org [Jan 29 13:22:10] VERBOSE[31991] logger.c: -- SIP/peer-cdr-0820cbc0 is circuit-busy [Jan 29 13:22:10] DEBUG[31991] channel.c: Hanging up channel 'SIP/peer-cdr-0820cbc0' [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Hangup call SIP/peer-cdr-0820cbc0, SIP callid 11d2544110f652792db04c613f7b19f3@mydomain.org [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Hanging up channel in state Down (not UP) [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] sched.c: Attempted to delete nonexistent schedule entry 0! [Jan 29 13:22:10] DEBUG[31991] devicestate.c: Notification of state change to be queued on device/channel SIP/peer-cdr-0820cbc0 [Jan 29 13:22:10] DEBUG[31991] devicestate.c: Notification of state change to be queued on device/channel SIP/peer-cdr [Jan 29 13:22:10] DEBUG[31991] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Jan 29 13:22:10] VERBOSE[31991] logger.c: == Everyone is busy/congested at this time (1:0/1/0) [Jan 29 13:22:10] DEBUG[31991] app_dial.c: Exiting with DIALSTATUS=CONGESTION. [Jan 29 13:22:10] VERBOSE[31991] logger.c: -- Auto fallthrough, channel 'OSS/dsp' status is 'CONGESTION' [Jan 29 13:22:10] DEBUG[31991] channel.c: Driver for channel 'OSS/dsp' does not support indication 8, emulating it [Jan 29 13:22:10] DEBUG[31991] channel.c: Prodding channel 'OSS/dsp' [Jan 29 13:22:10] DEBUG[31991] channel.c: Set channel OSS/dsp to write format slin [Jan 29 13:22:10] DEBUG[31991] devicestate.c: Notification of state change to be queued on device/channel OSS/dsp [Jan 29 13:22:10] DEBUG[31991] devicestate.c: No provider found, checking channel drivers for SIP - peer-cdr-0820cbc0 [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Checking device state for peer peer-cdr-0820cbc0 [Jan 29 13:22:10] DEBUG[31991] devicestate.c: Changing state for SIP/peer-cdr-0820cbc0 - state 1 (Not in use) [Jan 29 13:22:10] DEBUG[31991] devicestate.c: No provider found, checking channel drivers for SIP - peer-cdr [Jan 29 13:22:10] DEBUG[31991] chan_sip.c: Checking device state for peer peer-cdr [Jan 29 13:22:10] DEBUG[31991] devicestate.c: Changing state for SIP/peer-cdr - state 1 (Not in use) [Jan 29 13:22:10] DEBUG[31991] devicestate.c: No provider found, checking channel drivers for OSS - dsp [Jan 29 13:22:10] DEBUG[31991] devicestate.c: Changing state for OSS/dsp - state 4 (Invalid) [Jan 29 13:22:10] DEBUG[31991] app_queue.c: Device 'SIP/peer-cdr-0820cbc0' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 13:22:10] DEBUG[31991] app_queue.c: Device 'SIP/peer-cdr' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 13:22:10] DEBUG[31991] app_queue.c: Device 'OSS/dsp' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jan 29 13:22:11] VERBOSE[31991] logger.c: Really destroying SIP dialog '11d2544110f652792db04c613f7b19f3@mydomain.org' Method: INVITE [Jan 29 13:22:11] DEBUG[31991] chan_sip.c: ---------- SIP HISTORY for '11d2544110f652792db04c613f7b19f3@mydomain.org' [Jan 29 13:22:11] DEBUG[31991] chan_sip.c: * SIP Call [Jan 29 13:22:11] DEBUG[31991] chan_sip.c: 001. Cancel Cause Call Rejected [Jan 29 13:22:11] DEBUG[31991] chan_sip.c: ---------- END SIP HISTORY for '11d2544110f652792db04c613f7b19f3@mydomain.org'