localhost*CLI> timing test Attempting to test a timer with 50 ticks per second. Using the 'pthread' timing module for this test. localhost*CLI> localhost*CLI> localhost*CLI> It has been 1000 milliseconds, and we got 50 timer ticks localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> localhost*CLI> <--- SIP read from UDP:192.168.125.212:5060 ---> INVITE sip:1111@192.168.125.119 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.212:5060;branch=z9hG4bK903258324 From: ;tag=823250688 To: Call-ID: 899718498@192.168.125.212 CSeq: 1 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, BYE, PRACK, REFER, NOTIFY, INFO Content-Encoding: identity Max-Forwards: 70 User-Agent: ZyXEL V500-Series Content-Length: 226 v=0 o=4001 9082 9082 IN IP4 192.168.125.212 s=- c=IN IP4 192.168.125.212 t=0 0 m=audio 50162 RTP/AVP 0 8 18 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=sendrecv a=ptime:20 <-------------> --- (13 headers 12 lines) --- Sending to 192.168.125.212:5060 (no NAT) Using INVITE request as basis request - 899718498@192.168.125.212 Found peer '4001' for '4001' from 192.168.125.212:5060 <--- Reliably Transmitting (no NAT) to 192.168.125.212:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.125.212:5060;branch=z9hG4bK903258324;received=192.168.125.212 From: ;tag=823250688 To: ;tag=as70001ef1 Call-ID: 899718498@192.168.125.212 CSeq: 1 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d07122e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '899718498@192.168.125.212' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.125.212:5060 ---> ACK sip:1111@192.168.125.119 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.212:5060;branch=z9hG4bK903258324 From: ;tag=823250688 To: ;tag=as70001ef1 Call-ID: 899718498@192.168.125.212 CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:192.168.125.212:5060 ---> INVITE sip:1111@192.168.125.119 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.212:5060;branch=z9hG4bK448357958 From: ;tag=823250688 To: Call-ID: 899718498@192.168.125.212 CSeq: 2 INVITE Contact: Authorization: Digest username="4001", realm="asterisk", nonce="1d07122e", uri="sip:1111@192.168.125.119", response="e6915fcad04569d4db6b2a499fe20fe2", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, BYE, PRACK, REFER, NOTIFY, INFO Content-Encoding: identity Max-Forwards: 70 User-Agent: ZyXEL V500-Series Content-Length: 226 v=0 o=4001 9082 9082 IN IP4 192.168.125.212 s=- c=IN IP4 192.168.125.212 t=0 0 m=audio 50162 RTP/AVP 0 8 18 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=sendrecv a=ptime:20 <-------------> --- (14 headers 12 lines) --- Sending to 192.168.125.212:5060 (no NAT) Using INVITE request as basis request - 899718498@192.168.125.212 Found peer '4001' for '4001' from 192.168.125.212:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Capabilities: us - 0x18130e (gsm|ulaw|alaw|g729|speex|g722|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.125.212:50162 Looking for 1111 in default (domain 192.168.125.119) list_route: hop: <--- Transmitting (no NAT) to 192.168.125.212:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.125.212:5060;branch=z9hG4bK448357958;received=192.168.125.212 From: ;tag=823250688 To: Call-ID: 899718498@192.168.125.212 CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [1111@default:1] Playback("SIP/4001-00000002", "agent-pass") in new stack Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP <--- Reliably Transmitting (no NAT) to 192.168.125.212:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.125.212:5060;branch=z9hG4bK448357958;received=192.168.125.212 From: ;tag=823250688 To: ;tag=as35c9d852 Call-ID: 899718498@192.168.125.212 CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 282 v=0 o=root 383476244 383476244 IN IP4 192.168.125.119 s=Asterisk PBX 1.8.2.2 c=IN IP4 192.168.125.119 t=0 0 m=audio 12334 RTP/AVP 0 8 18 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.125.212:5060 ---> ACK sip:1111@192.168.125.119:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.212:5060;branch=z9hG4bK1468232609 From: ;tag=823250688 To: ;tag=as35c9d852 Call-ID: 899718498@192.168.125.212 CSeq: 2 ACK Max-Forwards: 70 User-Agent: ZyXEL V500-Series Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- Playing 'agent-pass.gsm' (language 'en') localhost*CLI> localhost*CLI> localhost*CLI> -- Executing [1111@default:2] Dial("SIP/4001-00000002", "SIP/1111") in new stack == Using SIP RTP CoS mark 5 [Mar 4 14:44:31] WARNING[23191]: acl.c:698 ast_ouraddrfor: Cannot connect Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x800000000000 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 0.0.4.87:5060: INVITE sip:1111 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK20d22441 Max-Forwards: 70 From: "4001" ;tag=as76c95d6d To: Contact: Call-ID: 4ec084870238ae43462efd0d31c0e9b6@127.0.0.1:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.2.2 Date: Fri, 04 Mar 2011 06:44:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 304 v=0 o=root 2131649163 2131649163 IN IP4 127.0.0.1 s=Asterisk PBX 1.8.2.2 c=IN IP4 127.0.0.1 t=0 0 m=audio 13364 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/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 --- [Mar 4 14:44:31] WARNING[23191]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x9f6bf88 (len 817) to 0.0.4.87:5060 returned -1: Invalid argument -- Called 1111