oxygen*CLI> agent show 21 (Agent 21) not logged in (musiconhold is 'default') 22 (Agent 22) available at '2002@test_queue' (musiconhold is 'default') 23 (Agent 23) available at '2003@test_queue' (musiconhold is 'default') 3 agents configured [2 online , 1 offline] oxygen*CLI> queue show queue21 queue21 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/22 (Not in use) has taken no calls yet Agent/21 (Unavailable) has taken no calls yet Agent/23 (Not in use) has taken no calls yet No Callers oxygen*CLI> <--- SIP read from 192.168.123.202:5063 ---> INVITE sip:80@192.168.123.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.202:5063;rport;branch=z9hG4bKnibbbyvi Max-Forwards: 70 To: From: "2mille1" ;tag=ghlvl Call-ID: izvyadsfmxshrcf@tux CSeq: 917 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 315 v=0 o=twinkle 2127288723 2084850699 IN IP4 192.168.123.202 s=- c=IN IP4 192.168.123.202 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 14 lines) --- Sending to 192.168.123.202 : 5063 (NAT) Using INVITE request as basis request - izvyadsfmxshrcf@tux <--- Reliably Transmitting (no NAT) to 192.168.123.202:5063 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.123.202:5063;branch=z9hG4bKnibbbyvi;received=192.168.123.202;rport=5063 From: "2mille1" ;tag=ghlvl To: ;tag=as1dba0ef2 Call-ID: izvyadsfmxshrcf@tux CSeq: 917 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7d31ec11" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'izvyadsfmxshrcf@tux' in 32000 ms (Method: INVITE) Found user '2001' <--- SIP read from 192.168.123.202:5063 ---> ACK sip:80@192.168.123.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.202:5063;rport;branch=z9hG4bKnibbbyvi Max-Forwards: 70 To: ;tag=as1dba0ef2 From: "2mille1" ;tag=ghlvl Call-ID: izvyadsfmxshrcf@tux CSeq: 917 ACK User-Agent: Twinkle/1.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 192.168.123.202:5063 ---> INVITE sip:80@192.168.123.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.202:5063;rport;branch=z9hG4bKnjmuaobg Max-Forwards: 70 Proxy-Authorization: Digest username="2001",realm="asterisk",nonce="7d31ec11",uri="sip:80@192.168.123.2",response="57beebe5aed9d061c5a32834b9177342",algorithm=MD5 To: From: "2mille1" ;tag=ghlvl Call-ID: izvyadsfmxshrcf@tux CSeq: 918 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 315 v=0 o=twinkle 2127288723 2084850699 IN IP4 192.168.123.202 s=- c=IN IP4 192.168.123.202 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 14 lines) --- Sending to 192.168.123.202 : 5063 (NAT) Using INVITE request as basis request - izvyadsfmxshrcf@tux Found user '2001' Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.123.202:8000 Found audio description format speex for ID 98 Found audio description format speex for ID 97 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.123.202:8000 Looking for 80 in test_queue (domain 192.168.123.2) list_route: hop: <--- Transmitting (no NAT) to 192.168.123.202:5063 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.123.202:5063;branch=z9hG4bKnjmuaobg;received=192.168.123.202;rport=5063 From: "2mille1" ;tag=ghlvl To: Call-ID: izvyadsfmxshrcf@tux CSeq: 918 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [80@test_queue:1] Answer("SIP/2001-081e9988", "") in new stack Audio is at 192.168.123.2 port 19534 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP oxygen*CLI> <--- Reliably Transmitting (no NAT) to 192.168.123.202:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.123.202:5063;branch=z9hG4bKnjmuaobg;received=192.168.123.202;rport=5063 From: "2mille1" ;tag=ghlvl To: ;tag=as5d772233 Call-ID: izvyadsfmxshrcf@tux CSeq: 918 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 27566 27566 IN IP4 192.168.123.2 s=session c=IN IP4 192.168.123.2 t=0 0 m=audio 19534 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 <------------> -- Executing [80@test_queue:2] Queue("SIP/2001-081e9988", "queue21|t|||120") in new stack -- Started music on hold, class 'default', on SIP/2001-081e9988 -- outgoing agentcall, to agent '23', on 'Local/2003@test_queue-4ffa,1' -- outgoing agentcall, to agent '22', on 'Local/2002@test_queue-c680,1' -- Executing [2003@test_queue:1] Answer("Local/2003@test_queue-4ffa,2", "") in new stack -- Executing [2003@test_queue:2] Dial("Local/2003@test_queue-4ffa,2", "SIP/2003|60") in new stack Audio is at 192.168.123.2 port 16184 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.123.101:5060: INVITE sip:2003@192.168.123.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK02ef2354;rport From: "2mille1" ;tag=as0903244b To: Contact: Call-ID: 28a0bc0c1de6662912c1dda161efd5e7@192.168.123.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 Aug 2008 16:25:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 289 v=0 o=root 27566 27566 IN IP4 192.168.123.2 s=session c=IN IP4 192.168.123.2 t=0 0 m=audio 16184 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 --- oxygen*CLI> <--- SIP read from 192.168.123.202:5063 ---> ACK sip:80@192.168.123.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.202:5063;rport;branch=z9hG4bKyndlnsvp Max-Forwards: 70 Proxy-Authorization: Digest username="2001",realm="asterisk",nonce="7d31ec11",uri="sip:80@192.168.123.2",response="57beebe5aed9d061c5a32834b9177342",algorithm=MD5 To: ;tag=as5d772233 From: "2mille1" ;tag=ghlvl Call-ID: izvyadsfmxshrcf@tux CSeq: 918 ACK User-Agent: Twinkle/1.2 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Called 2003 -- Agent/23 answered SIP/2001-081e9988 -- Stopped music on hold on SIP/2001-081e9988 [Aug 22 19:25:49] WARNING[27602]: app_queue.c:3014 try_calling: The device state of this queue member, Agent/23, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. -- Local/2003@test_queue-4ffa,2 requested special control 20, passing it to SIP/2003-081f3460 -- Executing [2002@test_queue:1] Answer("Local/2002@test_queue-c680,2", "") in new stack oxygen*CLI> <--- SIP read from 192.168.123.101:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK02ef2354;rport From: "2mille1";tag=as0903244b To: Call-ID: 28a0bc0c1de6662912c1dda161efd5e7@192.168.123.2 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 192.168.123.101:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK02ef2354;rport From: "2mille1";tag=as0903244b To: ;tag=c0a80101-2668b2 Call-ID: 28a0bc0c1de6662912c1dda161efd5e7@192.168.123.2 CSeq: 102 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/2003-081f3460 is ringing -- Local/2003@test_queue-4ffa,2 requested special control 20, passing it to SIP/2003-081f3460 -- Local/2003@test_queue-4ffa,2 requested special control 20, passing it to SIP/2003-081f3460 -- Executing [2002@test_queue:2] Dial("Local/2002@test_queue-c680,2", "SIP/2002|60") in new stack Audio is at 192.168.123.2 port 11056 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.123.100:5060: INVITE sip:2002@192.168.123.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK016da573;rport From: "2mille1" ;tag=as4777fdfe To: Contact: Call-ID: 730788d467c654776a93efa376f11252@192.168.123.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 Aug 2008 16:25:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 289 v=0 o=root 27566 27566 IN IP4 192.168.123.2 s=session c=IN IP4 192.168.123.2 t=0 0 m=audio 11056 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 --- -- Called 2002 Reliably Transmitting (no NAT) to 192.168.123.100:5060: CANCEL sip:2002@192.168.123.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK016da573;rport From: "2mille1" ;tag=as4777fdfe To: Call-ID: 730788d467c654776a93efa376f11252@192.168.123.2 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '730788d467c654776a93efa376f11252@192.168.123.2' in 32000 ms (Method: INVITE) == Spawn extension (test_queue, 2002, 2) exited non-zero on 'Local/2002@test_queue-c680,2' oxygen*CLI> <--- SIP read from 192.168.123.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK016da573;rport From: "2mille1";tag=as4777fdfe To: Call-ID: 730788d467c654776a93efa376f11252@192.168.123.2 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 192.168.123.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK016da573;rport From: "2mille1";tag=as4777fdfe To: ;tag=c0a80101-2a8f64 Call-ID: 730788d467c654776a93efa376f11252@192.168.123.2 CSeq: 102 CANCEL Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 192.168.123.100:5060 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK016da573;rport From: "2mille1";tag=as4777fdfe To: ;tag=c0a80101-2a8f64 Call-ID: 730788d467c654776a93efa376f11252@192.168.123.2 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Transmitting (no NAT) to 192.168.123.100:5060: ACK sip:2002@192.168.123.100:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK016da573;rport From: "2mille1" ;tag=as4777fdfe To: ;tag=c0a80101-2a8f64 Contact: Call-ID: 730788d467c654776a93efa376f11252@192.168.123.2 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '730788d467c654776a93efa376f11252@192.168.123.2' Method: INVITE -- Nobody picked up in 60000 ms Scheduling destruction of SIP dialog '28a0bc0c1de6662912c1dda161efd5e7@192.168.123.2' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.123.101:5060: CANCEL sip:2003@192.168.123.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK02ef2354;rport From: "2mille1" ;tag=as0903244b To: Call-ID: 28a0bc0c1de6662912c1dda161efd5e7@192.168.123.2 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '28a0bc0c1de6662912c1dda161efd5e7@192.168.123.2' in 32000 ms (Method: INVITE) -- Executing [2003@test_queue:3] Hangup("Local/2003@test_queue-4ffa,2", "") in new stack == Spawn extension (test_queue, 2003, 3) exited non-zero on 'Local/2003@test_queue-4ffa,2' == Spawn extension (test_queue, 80, 2) exited non-zero on 'SIP/2001-081e9988' Scheduling destruction of SIP dialog 'izvyadsfmxshrcf@tux' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.123.202, port 5063 Reliably Transmitting (no NAT) to 192.168.123.202:5063: BYE sip:2001@192.168.123.202:5063 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK540c9954;rport From: ;tag=as5d772233 To: "2mille1" ;tag=ghlvl Call-ID: izvyadsfmxshrcf@tux CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- oxygen*CLI> <--- SIP read from 192.168.123.202:5063 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.123.2:5060;rport=5060;branch=z9hG4bK540c9954 To: "2mille1" ;tag=ghlvl From: ;tag=as5d772233 Call-ID: izvyadsfmxshrcf@tux CSeq: 102 BYE Server: Twinkle/1.2 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'izvyadsfmxshrcf@tux' Method: ACK <--- SIP read from 192.168.123.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK02ef2354;rport From: "2mille1";tag=as0903244b To: ;tag=c0a80101-2668b2 Call-ID: 28a0bc0c1de6662912c1dda161efd5e7@192.168.123.2 CSeq: 102 CANCEL Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 192.168.123.101:5060 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK02ef2354;rport From: "2mille1";tag=as0903244b To: ;tag=c0a80101-2668b2 Call-ID: 28a0bc0c1de6662912c1dda161efd5e7@192.168.123.2 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Transmitting (no NAT) to 192.168.123.101:5060: ACK sip:2003@192.168.123.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.123.2:5060;branch=z9hG4bK02ef2354;rport From: "2mille1" ;tag=as0903244b To: ;tag=c0a80101-2668b2 Contact: Call-ID: 28a0bc0c1de6662912c1dda161efd5e7@192.168.123.2 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '28a0bc0c1de6662912c1dda161efd5e7@192.168.123.2' Method: INVITE oxygen*CLI>