[root@dot90 ~]# asterisk -vr Asterisk SVN-branch-1.2-r48584, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk SVN-branch-1.2-r48584 currently running on dot90 (pid = 12356) Verbosity is at least 1 Core debug is at least 1 dot90*CLI> set verbose 9 Verbosity was 1 and is now 9 dot90*CLI> set debug 9 Core debug was 1 and is now 9 dot90*CLI> sip debug SIP Debugging enabled dot90*CLI> <-- SIP read from 192.168.1.31:2158: --- (0 headers 0 lines) Nat keepalive --- dot90*CLI> <-- SIP read from 192.168.1.34:2069: INVITE sip:101@192.168.1.90;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.34:2069;branch=z9hG4bK-e3tv2igt81yb;rport From: ;tag=j811bts70y To: Call-ID: 3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/6.5.1 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 471 v=0 o=root 44659436 44659436 IN IP4 192.168.1.34 s=call c=IN IP4 192.168.1.34 t=0 0 m=audio 52066 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:aflYXCWGVVMdYixDzZPV+ma672fNyJgpVy019pEX a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv --- (18 headers 19 lines) --- Using INVITE request as basis request - 3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE Sending to 192.168.1.34 : 2069 (NAT) Reliably Transmitting (no NAT) to 192.168.1.34:2069: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.34:2069;branch=z9hG4bK-e3tv2igt81yb;rport;received=192.168.1.34 From: ;tag=j811bts70y To: ;tag=as71a19bdf Call-ID: 3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="242d9689" Content-Length: 0 --- Scheduling destruction of call '3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE' in 15000 ms Found user '102' dot90*CLI> <-- SIP read from 192.168.1.34:2069: ACK sip:101@192.168.1.90;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.34:2069;branch=z9hG4bK-e3tv2igt81yb;rport From: ;tag=j811bts70y To: ;tag=as71a19bdf Call-ID: 3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- dot90*CLI> <-- SIP read from 192.168.1.34:2069: INVITE sip:101@192.168.1.90;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.34:2069;branch=z9hG4bK-g6r15o67re4s;rport From: ;tag=j811bts70y To: Call-ID: 3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/6.5.1 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="102",realm="asterisk",nonce="242d9689",uri="sip:101@192.168.1.90;user=phone",response="3874158f6cc7b209e081d210b03b90ad",algorithm=md5 Content-Type: application/sdp Content-Length: 471 v=0 o=root 44659436 44659436 IN IP4 192.168.1.34 s=call c=IN IP4 192.168.1.34 t=0 0 m=audio 52066 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:aflYXCWGVVMdYixDzZPV+ma672fNyJgpVy019pEX a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv --- (19 headers 19 lines) --- Using INVITE request as basis request - 3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE Sending to 192.168.1.34 : 2069 (NAT) Found user '102' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.34:52066 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 101 in custom-100users (domain 192.168.1.90;user=phone) list_route: hop: Transmitting (no NAT) to 192.168.1.34:2069: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.34:2069;branch=z9hG4bK-g6r15o67re4s;rport;received=192.168.1.34 From: ;tag=j811bts70y To: Call-ID: 3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/102-093d29a8", "exten-vm|101|101") in new stack -- Executing Macro("SIP/102-093d29a8", "user-callerid") in new stack -- Executing GotoIf("SIP/102-093d29a8", "0?report") in new stack -- Executing GotoIf("SIP/102-093d29a8", "0?start") in new stack -- Executing Set("SIP/102-093d29a8", "REALCALLERIDNUM=102") in new stack -- Executing NoOp("SIP/102-093d29a8", "REALCALLERIDNUM is 102") in new stack -- Executing Set("SIP/102-093d29a8", "AMPUSER=102") in new stack -- Executing Set("SIP/102-093d29a8", "AMPUSERCIDNAME=102") in new stack -- Executing GotoIf("SIP/102-093d29a8", "0?report") in new stack -- Executing Set("SIP/102-093d29a8", "CALLERID(all)=102 <102>") in new stack -- Executing NoOp("SIP/102-093d29a8", "Using CallerID "102" <102>") in new stack -- Executing Set("SIP/102-093d29a8", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/102-093d29a8", "VMBOX=101") in new stack -- Executing Set("SIP/102-093d29a8", "EXTTOCALL=101") in new stack -- Executing Set("SIP/102-093d29a8", "CFUEXT=") in new stack -- Executing Set("SIP/102-093d29a8", "RT=15") in new stack -- Executing Macro("SIP/102-093d29a8", "record-enable|101|IN") in new stack -- Executing GotoIf("SIP/102-093d29a8", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/102-093d29a8", "recordingcheck|20061221-230228|1166767348.0") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20061221-230228|1166767348.0: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/102-093d29a8", "No recording needed") in new stack -- Executing GotoIf("SIP/102-093d29a8", "0?dolocaldial|1") in new stack -- Executing Macro("SIP/102-093d29a8", "dial|15|Tt|101") in new stack -- Executing AGI("SIP/102-093d29a8", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi -- dialparties.agi: priority is 1 dialparties.agi: Caller ID name is '102' number is '102' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 101 to extension map -- dialparties.agi: Extension 101 cf is disabled -- dialparties.agi: Extension 101 do not disturb is disabled > dialparties.agi: extnum: 101 > dialparties.agi: exthascw: 1 > dialparties.agi: exthascfb: 0 > dialparties.agi: extcfb: > dialparties.agi: exthascfu: 0 > dialparties.agi: extcfu: -- dialparties.agi: DbSet CALLTRACE/101 to 102 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("SIP/102-093d29a8", "SIP/101|15|Tt") in new stack We're at 192.168.1.90 port 19388 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 192.168.1.31:2158: INVITE sip:101@192.168.1.31:2158;line=gv8x1x75 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK275b21eb From: "102" ;tag=as141708c8 To: Contact: Call-ID: 7402796813ed40d119f62af64921d437@192.168.1.90 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 Dec 2006 06:02:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 12356 12356 IN IP4 192.168.1.90 s=session c=IN IP4 192.168.1.90 t=0 0 m=audio 19388 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 101 dot90*CLI> <-- SIP read from 192.168.1.31:2158: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK275b21eb From: "102" ;tag=as141708c8 To: ;tag=kdgvqby7sr Call-ID: 7402796813ed40d119f62af64921d437@192.168.1.90 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 dot90*CLI> --- (10 headers 0 lines) --- -- SIP/101-093da990 is ringing Transmitting (no NAT) to 192.168.1.34:2069: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.34:2069;branch=z9hG4bK-g6r15o67re4s;rport;received=192.168.1.34 From: ;tag=j811bts70y To: ;tag=as4151ed87 Call-ID: 3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- dot90*CLI> <-- SIP read from 192.168.1.31:2158: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK275b21eb From: "102" ;tag=as141708c8 To: ;tag=kdgvqby7sr Call-ID: 7402796813ed40d119f62af64921d437@192.168.1.90 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 dot90*CLI> --- (10 headers 0 lines) --- -- SIP/101-093da990 is ringing dot90*CLI> <-- SIP read from 192.168.1.31:2158: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK275b21eb From: "102" ;tag=as141708c8 To: ;tag=kdgvqby7sr Call-ID: 7402796813ed40d119f62af64921d437@192.168.1.90 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom360/6.2.3 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 206 v=0 o=root 397487008 397487009 IN IP4 192.168.1.31 s=call c=IN IP4 192.168.1.31 t=0 0 m=audio 50568 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv --- (13 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.31:50568 Found description format pcmu Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.31, port 2158 Transmitting (no NAT) to 192.168.1.31:2158: ACK sip:101@192.168.1.31:2158;line=gv8x1x75 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.90:5060;branch=z9hG4bK1b31b519 From: "102" ;tag=as141708c8 To: ;tag=kdgvqby7sr Contact: Call-ID: 7402796813ed40d119f62af64921d437@192.168.1.90 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/101-093da990 answered SIP/102-093d29a8 We're at 192.168.1.90 port 17636 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.34:2069: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.34:2069;branch=z9hG4bK-g6r15o67re4s;rport;received=192.168.1.34 From: ;tag=j811bts70y To: ;tag=as4151ed87 Call-ID: 3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 12356 12356 IN IP4 192.168.1.90 s=session c=IN IP4 192.168.1.90 t=0 0 m=audio 17636 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- dot90*CLI> <-- SIP read from 192.168.1.34:2069: ACK sip:101@192.168.1.90 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.34:2069;branch=z9hG4bK-zywimigpcbbe;rport From: ;tag=j811bts70y To: ;tag=as4151ed87 Call-ID: 3c27b00e5573-cajwcaj96u3c@snom300-0004132516FE CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- -- Started music on hold, class 'default', on SIP/102-093d29a8 -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/102-093d29a8 -- Started music on hold, class 'default', on SIP/102-093d29a8 == Parked SIP/102-093d29a8 on 71. Will timeout back to extension [custom-100users] 101, 1 in 10 seconds -- Added extension '71' priority 1 to parkedcalls -- Playing 'digits/7' (language 'en') dot90*CLI> <-- SIP read from 192.168.1.31:2158: INVITE sip:102@192.168.1.90 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.31:2158;branch=z9hG4bK-zetgwq365hih;rport From: ;tag=kdgvqby7sr To: "102" ;tag=as141708c8 Call-ID: 7402796813ed40d119f62af64921d437@192.168.1.90 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.2.3 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 361 v=0 o=root 397487008 397487010 IN IP4 192.168.1.31 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 50568 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly --- (18 headers 17 lines) --- Using INVITE request as basis request - 7402796813ed40d119f62af64921d437@192.168.1.90 Sending to 192.168.1.31 : 2158 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:50568 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) We're at 192.168.1.90 port 19388 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.31:2158: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.31:2158;branch=z9hG4bK-zetgwq365hih;rport;received=192.168.1.31 From: ;tag=kdgvqby7sr To: "102" ;tag=as141708c8 Call-ID: 7402796813ed40d119f62af64921d437@192.168.1.90 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 12356 12357 IN IP4 192.168.1.90 s=session c=IN IP4 192.168.1.90 t=0 0 m=audio 19388 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'digits/1' (language 'en') dot90*CLI> <-- SIP read from 192.168.1.31:2158: INVITE sip:71@192.168.1.90;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.31:2158;branch=z9hG4bK-a87raa1pw0n9;rport From: "101" ;tag=wo7xmzm0ll To: Call-ID: 3c2671ca5573-ipuej6gijqfl@snom360-000413235410 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.2.3 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 368 v=0 o=root 1455741242 1455741242 IN IP4 192.168.1.31 s=call c=IN IP4 192.168.1.31 t=0 0 m=audio 55792 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- (18 headers 17 lines) --- Using INVITE request as basis request - 3c2671ca5573-ipuej6gijqfl@snom360-000413235410 Sending to 192.168.1.31 : 2158 (NAT) Reliably Transmitting (no NAT) to 192.168.1.31:2158: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.31:2158;branch=z9hG4bK-a87raa1pw0n9;rport;received=192.168.1.31 From: "101" ;tag=wo7xmzm0ll To: ;tag=as7cf7f542 Call-ID: 3c2671ca5573-ipuej6gijqfl@snom360-000413235410 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="65868284" Content-Length: 0 --- Scheduling destruction of call '3c2671ca5573-ipuej6gijqfl@snom360-000413235410' in 15000 ms Found user '101' dot90*CLI> <-- SIP read from 192.168.1.31:2158: ACK sip:102@192.168.1.90 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.31:2158;branch=z9hG4bK-gjj4qplsq6ky;rport From: ;tag=kdgvqby7sr To: "102" ;tag=as141708c8 Call-ID: 7402796813ed40d119f62af64921d437@192.168.1.90 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- dot90*CLI> <-- SIP read from 192.168.1.31:2158: ACK sip:71@192.168.1.90;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.31:2158;branch=z9hG4bK-a87raa1pw0n9;rport From: "101" ;tag=wo7xmzm0ll To: ;tag=as7cf7f542 Call-ID: 3c2671ca5573-ipuej6gijqfl@snom360-000413235410 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- dot90*CLI> <-- SIP read from 192.168.1.31:2158: INVITE sip:71@192.168.1.90;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.31:2158;branch=z9hG4bK-2yxpf6apeh08;rport From: "101" ;tag=wo7xmzm0ll To: Call-ID: 3c2671ca5573-ipuej6gijqfl@snom360-000413235410 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.2.3 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="101",realm="asterisk",nonce="65868284",uri="sip:71@192.168.1.90;user=phone",response="6b9dc8a1a090ef8e716d0eec9e113e9b",algorithm=md5 Content-Type: application/sdp Content-Length: 368 v=0 o=root 1455741242 1455741242 IN IP4 192.168.1.31 s=call c=IN IP4 192.168.1.31 t=0 0 m=audio 55792 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- (19 headers 17 lines) --- Using INVITE request as basis request - 3c2671ca5573-ipuej6gijqfl@snom360-000413235410 Sending to 192.168.1.31 : 2158 (NAT) Found user '101' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.31:55792 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 71 in custom-100users (domain 192.168.1.90;user=phone) list_route: hop: Transmitting (no NAT) to 192.168.1.31:2158: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.31:2158;branch=z9hG4bK-2yxpf6apeh08;rport;received=192.168.1.31 From: "101" ;tag=wo7xmzm0ll To: Call-ID: 3c2671ca5573-ipuej6gijqfl@snom360-000413235410 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing ParkedCall("SIP/101-093ed128", "71") in new stack We're at 192.168.1.90 port 13142 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.31:2158: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.31:2158;branch=z9hG4bK-2yxpf6apeh08;rport;received=192.168.1.31 From: "101" ;tag=wo7xmzm0ll To: ;tag=as318c06aa Call-ID: 3c2671ca5573-ipuej6gijqfl@snom360-000413235410 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 12356 12356 IN IP4 192.168.1.90 s=session c=IN IP4 192.168.1.90 t=0 0 m=audio 13142 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Stopped music on hold on SIP/102-093d29a8 -- Channel SIP/101-093ed128 connected to parked call 71 -- Attempting native bridge of SIP/101-093ed128 and SIP/102-093d29a8 dot90*CLI> <-- SIP read from 192.168.1.31:2158: ACK sip:71@192.168.1.90 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.31:2158;branch=z9hG4bK-lh7m51mokqyt;rport From: "101" ;tag=wo7xmzm0ll To: ;tag=as318c06aa Call-ID: 3c2671ca5573-ipuej6gijqfl@snom360-000413235410 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- dot90*CLI> Disconnected from Asterisk server Executing last minute cleanups [root@dot90 ~]#