localhost*CLI> timing test Attempting to test a timer with 50 ticks per second. Using the 'pthread' timing module for this test. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. It has been 1009 milliseconds, and we got 0 timer ticks 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=z9hG4bK178198445 From: ;tag=494956274 To: Call-ID: 1485139328@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 6410 6410 IN IP4 192.168.125.212 s=- c=IN IP4 192.168.125.212 t=0 0 m=audio 50164 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 - 1485139328@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=z9hG4bK178198445;received=192.168.125.212 From: ;tag=494956274 To: ;tag=as0603b732 Call-ID: 1485139328@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="6b6efe67" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1485139328@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=z9hG4bK178198445 From: ;tag=494956274 To: ;tag=as0603b732 Call-ID: 1485139328@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=z9hG4bK2115044777 From: ;tag=494956274 To: Call-ID: 1485139328@192.168.125.212 CSeq: 2 INVITE Contact: Authorization: Digest username="4001", realm="asterisk", nonce="6b6efe67", uri="sip:1111@192.168.125.119", response="23a50f0fa1ec2d9fd9dfda59454e0371", 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 6410 6410 IN IP4 192.168.125.212 s=- c=IN IP4 192.168.125.212 t=0 0 m=audio 50164 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 - 1485139328@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 192.168.125.212:50164 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=z9hG4bK2115044777;received=192.168.125.212 From: ;tag=494956274 To: Call-ID: 1485139328@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/4001-00000004", "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=z9hG4bK2115044777;received=192.168.125.212 From: ;tag=494956274 To: ;tag=as59caa3e7 Call-ID: 1485139328@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 529700572 529700572 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 11206 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=z9hG4bK300293660 From: ;tag=494956274 To: ;tag=as59caa3e7 Call-ID: 1485139328@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> exit