# Roles asterisk IP: 10.34.1.11 SIP/je5qtq IP: 10.34.1.101 User "Alice" SIP/as2mkq IP: 10.34.1.168 User "Bob" # Scenario 11:28:31 - "Alice" calls "Bob" via the 145 extension 11:28:32 - "Bob" answers; reINVITE happens and "Alice" and "Bob" are in direct media 11:28:35 - "Alice" sends a first SIP INFO DTMF with duration 160 11:28:38 - "Alice" sends a second SIP INFO DTMF with duration 160; upon receiving that DTMF, asterisk sends a first SIP INFO DTMF to "Bob" with duration 2796 (we expected that DTMF to be sent earlier and not have such a long duration) 11:28:41 - "Alice" sends a third SIP INFO DTMF with duration 160; upon receiving that DTMF, asterisk sends a second SIP INFO DTMF to "Bob" with duration 3162 11:28:45 - The call is hung up by "Alice"; the third DTMF was never sent by asterisk to "Bob" # CLI output [Jun 30 11:28:31] <--- SIP read from UDP:10.34.1.101:5060 ---> [Jun 30 11:28:31] INVITE sip:145@10.34.1.11;user=phone SIP/2.0 [Jun 30 11:28:31] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK9d4c8632e33850d23 [Jun 30 11:28:31] Max-Forwards: 70 [Jun 30 11:28:31] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:31] To: [Jun 30 11:28:31] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:31] CSeq: 18688 INVITE [Jun 30 11:28:31] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:31] Allow-Events: aastra-xml, talk, hold, conference, LocalModeStatus [Jun 30 11:28:31] Contact: "Alice" ;+sip.instance="" [Jun 30 11:28:31] Supported: path, gruu, 100rel, replaces [Jun 30 11:28:31] User-Agent: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:31] Content-Type: application/sdp [Jun 30 11:28:31] Content-Length: 302 [Jun 30 11:28:31] [Jun 30 11:28:31] v=0 [Jun 30 11:28:31] o=MxSIP 0 1 IN IP4 10.34.1.101 [Jun 30 11:28:31] s=SIP Call [Jun 30 11:28:31] c=IN IP4 10.34.1.101 [Jun 30 11:28:31] t=0 0 [Jun 30 11:28:31] m=audio 3000 RTP/AVP 8 0 18 101 [Jun 30 11:28:31] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:31] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:31] a=rtpmap:18 G729/8000 [Jun 30 11:28:31] a=rtpmap:101 telephone-event/8000 [Jun 30 11:28:31] a=silenceSupp:on - - - - [Jun 30 11:28:31] a=fmtp:18 annexb=yes [Jun 30 11:28:31] a=fmtp:101 0-15 [Jun 30 11:28:31] a=ptime:20 [Jun 30 11:28:31] a=sendrecv [Jun 30 11:28:31] <-------------> [Jun 30 11:28:31] --- (14 headers 15 lines) --- [Jun 30 11:28:31] Sending to 10.34.1.101:5060 (no NAT) [Jun 30 11:28:31] Sending to 10.34.1.101:5060 (no NAT) [Jun 30 11:28:31] Using INVITE request as basis request - 42b2ea4fa1b0ed52 [Jun 30 11:28:31] Found peer 'je5qtq' for 'je5qtq' from 10.34.1.101:5060 [Jun 30 11:28:31] [Jun 30 11:28:31] <--- Reliably Transmitting (no NAT) to 10.34.1.101:5060 ---> [Jun 30 11:28:31] SIP/2.0 401 Unauthorized [Jun 30 11:28:31] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK9d4c8632e33850d23;received=10.34.1.101 [Jun 30 11:28:31] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:31] To: ;tag=as01890daf [Jun 30 11:28:31] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:31] CSeq: 18688 INVITE [Jun 30 11:28:31] Server: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:31] Supported: replaces, timer [Jun 30 11:28:31] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dc0228a" [Jun 30 11:28:31] Content-Length: 0 [Jun 30 11:28:31] [Jun 30 11:28:31] [Jun 30 11:28:31] <------------> [Jun 30 11:28:31] Scheduling destruction of SIP dialog '42b2ea4fa1b0ed52' in 32000 ms (Method: INVITE) [Jun 30 11:28:31] [Jun 30 11:28:31] <--- SIP read from UDP:10.34.1.101:5060 ---> [Jun 30 11:28:31] ACK sip:145@10.34.1.11;user=phone SIP/2.0 [Jun 30 11:28:31] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK9d4c8632e33850d23 [Jun 30 11:28:31] Max-Forwards: 70 [Jun 30 11:28:31] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:31] To: ;tag=as01890daf [Jun 30 11:28:31] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:31] CSeq: 18688 ACK [Jun 30 11:28:31] User-Agent: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:31] Content-Length: 0 [Jun 30 11:28:31] [Jun 30 11:28:31] <-------------> [Jun 30 11:28:31] --- (9 headers 0 lines) --- [Jun 30 11:28:31] [Jun 30 11:28:31] <--- SIP read from UDP:10.34.1.101:5060 ---> [Jun 30 11:28:31] INVITE sip:145@10.34.1.11;user=phone SIP/2.0 [Jun 30 11:28:31] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK0def3b78919c4eb4a [Jun 30 11:28:31] Max-Forwards: 70 [Jun 30 11:28:31] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:31] To: [Jun 30 11:28:31] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:31] CSeq: 18689 INVITE [Jun 30 11:28:31] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:31] Allow-Events: aastra-xml, talk, hold, conference, LocalModeStatus [Jun 30 11:28:31] Authorization: Digest username="je5qtq",realm="asterisk",nonce="2dc0228a",uri="sip:145@10.34.1.11;user=phone",response="e1b7423d29ae7841161fc45b595afd4f",algorithm=MD5 [Jun 30 11:28:31] Contact: "Alice" ;+sip.instance="" [Jun 30 11:28:31] Supported: path, gruu, 100rel, replaces [Jun 30 11:28:31] User-Agent: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:31] Content-Type: application/sdp [Jun 30 11:28:31] Content-Length: 302 [Jun 30 11:28:31] [Jun 30 11:28:31] v=0 [Jun 30 11:28:31] o=MxSIP 0 1 IN IP4 10.34.1.101 [Jun 30 11:28:31] s=SIP Call [Jun 30 11:28:31] c=IN IP4 10.34.1.101 [Jun 30 11:28:31] t=0 0 [Jun 30 11:28:31] m=audio 3000 RTP/AVP 8 0 18 101 [Jun 30 11:28:31] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:31] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:31] a=rtpmap:18 G729/8000 [Jun 30 11:28:31] a=rtpmap:101 telephone-event/8000 [Jun 30 11:28:31] a=silenceSupp:on - - - - [Jun 30 11:28:31] a=fmtp:18 annexb=yes [Jun 30 11:28:31] a=fmtp:101 0-15 [Jun 30 11:28:31] a=ptime:20 [Jun 30 11:28:31] a=sendrecv [Jun 30 11:28:31] <-------------> [Jun 30 11:28:31] --- (15 headers 15 lines) --- [Jun 30 11:28:31] Sending to 10.34.1.101:5060 (no NAT) [Jun 30 11:28:31] Using INVITE request as basis request - 42b2ea4fa1b0ed52 [Jun 30 11:28:31] Found peer 'je5qtq' for 'je5qtq' from 10.34.1.101:5060 [Jun 30 11:28:31] == Using SIP RTP CoS mark 5 [Jun 30 11:28:31] Found RTP audio format 8 [Jun 30 11:28:31] Found RTP audio format 0 [Jun 30 11:28:31] Found RTP audio format 18 [Jun 30 11:28:31] Found RTP audio format 101 [Jun 30 11:28:31] Found audio description format PCMA for ID 8 [Jun 30 11:28:31] Found audio description format PCMU for ID 0 [Jun 30 11:28:31] Found audio description format G729 for ID 18 [Jun 30 11:28:31] Found audio description format telephone-event for ID 101 [Jun 30 11:28:31] Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Jun 30 11:28:31] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [Jun 30 11:28:31] Peer audio RTP is at port 10.34.1.101:3000 [Jun 30 11:28:31] Looking for 145 in default (domain 10.34.1.11) [Jun 30 11:28:31] sip_route_dump: route/path hop: [Jun 30 11:28:31] [Jun 30 11:28:31] <--- Transmitting (no NAT) to 10.34.1.101:5060 ---> [Jun 30 11:28:31] SIP/2.0 100 Trying [Jun 30 11:28:31] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK0def3b78919c4eb4a;received=10.34.1.101 [Jun 30 11:28:31] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:31] To: [Jun 30 11:28:31] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:31] CSeq: 18689 INVITE [Jun 30 11:28:31] Server: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:31] Supported: replaces, timer [Jun 30 11:28:31] Contact: [Jun 30 11:28:31] Content-Length: 0 [Jun 30 11:28:31] [Jun 30 11:28:31] [Jun 30 11:28:31] <------------> [Jun 30 11:28:31] -- Executing [145@default:1] NoOp("SIP/je5qtq-00000004", "") in new stack [Jun 30 11:28:31] -- Executing [145@default:2] Dial("SIP/je5qtq-00000004", "SIP/as2mkq") in new stack [Jun 30 11:28:31] == Using SIP RTP CoS mark 5 [Jun 30 11:28:31] Audio is at 18484 [Jun 30 11:28:31] Adding codec ulaw to SDP [Jun 30 11:28:31] Adding codec alaw to SDP [Jun 30 11:28:31] Adding codec gsm to SDP [Jun 30 11:28:31] Reliably Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:31] INVITE sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:31] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK48f5e60f [Jun 30 11:28:31] Max-Forwards: 70 [Jun 30 11:28:31] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:31] To: [Jun 30 11:28:31] Contact: [Jun 30 11:28:31] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:31] CSeq: 102 INVITE [Jun 30 11:28:31] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:31] Date: Tue, 30 Jun 2015 15:28:31 GMT [Jun 30 11:28:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:31] Supported: replaces, timer [Jun 30 11:28:31] Content-Type: application/sdp [Jun 30 11:28:31] Content-Length: 238 [Jun 30 11:28:31] [Jun 30 11:28:31] v=0 [Jun 30 11:28:31] o=root 602169165 602169165 IN IP4 10.34.1.11 [Jun 30 11:28:31] s=Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:31] c=IN IP4 10.34.1.11 [Jun 30 11:28:31] t=0 0 [Jun 30 11:28:31] m=audio 18484 RTP/AVP 0 8 3 [Jun 30 11:28:31] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:31] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:31] a=rtpmap:3 GSM/8000 [Jun 30 11:28:31] a=maxptime:150 [Jun 30 11:28:31] a=sendrecv [Jun 30 11:28:31] [Jun 30 11:28:31] --- [Jun 30 11:28:31] -- Called SIP/as2mkq [Jun 30 11:28:31] [Jun 30 11:28:31] <--- SIP read from UDP:10.34.1.168:5060 ---> [Jun 30 11:28:31] SIP/2.0 180 Ringing [Jun 30 11:28:31] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK48f5e60f [Jun 30 11:28:31] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:31] To: ;tag=2639294411 [Jun 30 11:28:31] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:31] CSeq: 102 INVITE [Jun 30 11:28:31] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:31] Allow-Events: talk, hold, conference, LocalModeStatus [Jun 30 11:28:31] Contact: "Bob" ;+sip.instance="" [Jun 30 11:28:31] Server: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:31] Supported: path [Jun 30 11:28:31] Content-Length: 0 [Jun 30 11:28:31] [Jun 30 11:28:31] <-------------> [Jun 30 11:28:31] --- (12 headers 0 lines) --- [Jun 30 11:28:31] sip_route_dump: route/path hop: [Jun 30 11:28:31] -- SIP/as2mkq-00000005 is ringing [Jun 30 11:28:31] [Jun 30 11:28:31] <--- Transmitting (no NAT) to 10.34.1.101:5060 ---> [Jun 30 11:28:31] SIP/2.0 180 Ringing [Jun 30 11:28:31] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK0def3b78919c4eb4a;received=10.34.1.101 [Jun 30 11:28:31] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:31] To: ;tag=as66b808db [Jun 30 11:28:31] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:31] CSeq: 18689 INVITE [Jun 30 11:28:31] Server: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:31] Supported: replaces, timer [Jun 30 11:28:31] Contact: [Jun 30 11:28:31] Content-Length: 0 [Jun 30 11:28:31] [Jun 30 11:28:31] [Jun 30 11:28:31] <------------> [Jun 30 11:28:32] [Jun 30 11:28:32] <--- SIP read from UDP:10.34.1.168:5060 ---> [Jun 30 11:28:32] SIP/2.0 200 OK [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK48f5e60f [Jun 30 11:28:32] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:32] To: ;tag=2639294411 [Jun 30 11:28:32] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:32] CSeq: 102 INVITE [Jun 30 11:28:32] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:32] Allow-Events: talk, hold, conference, LocalModeStatus [Jun 30 11:28:32] Contact: "Bob" ;+sip.instance="" [Jun 30 11:28:32] Server: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:32] Supported: path, replaces [Jun 30 11:28:32] Content-Type: application/sdp [Jun 30 11:28:32] Content-Length: 187 [Jun 30 11:28:32] [Jun 30 11:28:32] v=0 [Jun 30 11:28:32] o=MxSIP 0 1 IN IP4 10.34.1.168 [Jun 30 11:28:32] s=SIP Call [Jun 30 11:28:32] c=IN IP4 10.34.1.168 [Jun 30 11:28:32] t=0 0 [Jun 30 11:28:32] m=audio 3000 RTP/AVP 0 8 [Jun 30 11:28:32] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:32] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:32] a=silenceSupp:off - - - - [Jun 30 11:28:32] a=sendrecv [Jun 30 11:28:32] <-------------> [Jun 30 11:28:32] --- (13 headers 10 lines) --- [Jun 30 11:28:32] Found RTP audio format 0 [Jun 30 11:28:32] Found RTP audio format 8 [Jun 30 11:28:32] Found audio description format PCMU for ID 0 [Jun 30 11:28:32] Found audio description format PCMA for ID 8 [Jun 30 11:28:32] Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Jun 30 11:28:32] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jun 30 11:28:32] Peer audio RTP is at port 10.34.1.168:3000 [Jun 30 11:28:32] sip_route_dump: route/path hop: [Jun 30 11:28:32] set_destination: Parsing for address/port to send to [Jun 30 11:28:32] set_destination: set destination to 10.34.1.168:5060 [Jun 30 11:28:32] Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:32] ACK sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK06ee2aae [Jun 30 11:28:32] Max-Forwards: 70 [Jun 30 11:28:32] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:32] To: ;tag=2639294411 [Jun 30 11:28:32] Contact: [Jun 30 11:28:32] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:32] CSeq: 102 ACK [Jun 30 11:28:32] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] Content-Length: 0 [Jun 30 11:28:32] [Jun 30 11:28:32] [Jun 30 11:28:32] --- [Jun 30 11:28:32] -- SIP/as2mkq-00000005 answered SIP/je5qtq-00000004 [Jun 30 11:28:32] Audio is at 11772 [Jun 30 11:28:32] Adding codec ulaw to SDP [Jun 30 11:28:32] Adding codec alaw to SDP [Jun 30 11:28:32] Adding codec gsm to SDP [Jun 30 11:28:32] [Jun 30 11:28:32] <--- Reliably Transmitting (no NAT) to 10.34.1.101:5060 ---> [Jun 30 11:28:32] SIP/2.0 200 OK [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK0def3b78919c4eb4a;received=10.34.1.101 [Jun 30 11:28:32] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:32] To: ;tag=as66b808db [Jun 30 11:28:32] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:32] CSeq: 18689 INVITE [Jun 30 11:28:32] Server: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:32] Supported: replaces, timer [Jun 30 11:28:32] Contact: [Jun 30 11:28:32] Content-Type: application/sdp [Jun 30 11:28:32] Content-Length: 240 [Jun 30 11:28:32] [Jun 30 11:28:32] v=0 [Jun 30 11:28:32] o=root 1248309523 1248309523 IN IP4 10.34.1.11 [Jun 30 11:28:32] s=Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] c=IN IP4 10.34.1.11 [Jun 30 11:28:32] t=0 0 [Jun 30 11:28:32] m=audio 11772 RTP/AVP 0 8 3 [Jun 30 11:28:32] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:32] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:32] a=rtpmap:3 GSM/8000 [Jun 30 11:28:32] a=maxptime:150 [Jun 30 11:28:32] a=sendrecv [Jun 30 11:28:32] [Jun 30 11:28:32] <------------> [Jun 30 11:28:32] -- Channel SIP/je5qtq-00000004 joined 'simple_bridge' basic-bridge <90c51b22-c746-44ee-811e-53346db1f05e> [Jun 30 11:28:32] -- Channel SIP/as2mkq-00000005 joined 'simple_bridge' basic-bridge <90c51b22-c746-44ee-811e-53346db1f05e> [Jun 30 11:28:32] > Bridge 90c51b22-c746-44ee-811e-53346db1f05e: switching from simple_bridge technology to native_rtp [Jun 30 11:28:32] set_destination: Parsing for address/port to send to [Jun 30 11:28:32] set_destination: set destination to 10.34.1.168:5060 [Jun 30 11:28:32] Audio is at 18484 [Jun 30 11:28:32] Adding codec ulaw to SDP [Jun 30 11:28:32] Adding codec alaw to SDP [Jun 30 11:28:32] Adding codec gsm to SDP [Jun 30 11:28:32] Reliably Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:32] INVITE sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK17e9655b [Jun 30 11:28:32] Max-Forwards: 70 [Jun 30 11:28:32] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:32] To: ;tag=2639294411 [Jun 30 11:28:32] Contact: [Jun 30 11:28:32] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:32] CSeq: 103 INVITE [Jun 30 11:28:32] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:32] Supported: replaces, timer [Jun 30 11:28:32] X-asterisk-Info: SIP re-invite (External RTP bridge) [Jun 30 11:28:32] Content-Type: application/sdp [Jun 30 11:28:32] Content-Length: 239 [Jun 30 11:28:32] [Jun 30 11:28:32] v=0 [Jun 30 11:28:32] o=root 602169165 602169166 IN IP4 10.34.1.101 [Jun 30 11:28:32] s=Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] c=IN IP4 10.34.1.101 [Jun 30 11:28:32] t=0 0 [Jun 30 11:28:32] m=audio 3000 RTP/AVP 0 8 3 [Jun 30 11:28:32] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:32] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:32] a=rtpmap:3 GSM/8000 [Jun 30 11:28:32] a=maxptime:150 [Jun 30 11:28:32] a=sendrecv [Jun 30 11:28:32] [Jun 30 11:28:32] --- [Jun 30 11:28:32] > Remotely bridged 'SIP/as2mkq-00000005' and 'SIP/je5qtq-00000004' - media will flow directly between them [Jun 30 11:28:32] > Remotely bridged 'SIP/as2mkq-00000005' and 'SIP/je5qtq-00000004' - media will flow directly between them [Jun 30 11:28:32] [Jun 30 11:28:32] <--- SIP read from UDP:10.34.1.101:5060 ---> [Jun 30 11:28:32] ACK sip:145@10.34.1.11:5060 SIP/2.0 [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK301a4530a96712d96 [Jun 30 11:28:32] Max-Forwards: 70 [Jun 30 11:28:32] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:32] To: ;tag=as66b808db [Jun 30 11:28:32] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:32] CSeq: 18689 ACK [Jun 30 11:28:32] Authorization: Digest username="je5qtq",realm="asterisk",nonce="2dc0228a",uri="sip:145@10.34.1.11;user=phone",response="e1b7423d29ae7841161fc45b595afd4f",algorithm=MD5 [Jun 30 11:28:32] User-Agent: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:32] Content-Length: 0 [Jun 30 11:28:32] [Jun 30 11:28:32] <-------------> [Jun 30 11:28:32] --- (10 headers 0 lines) --- [Jun 30 11:28:32] set_destination: Parsing for address/port to send to [Jun 30 11:28:32] set_destination: set destination to 10.34.1.101:5060 [Jun 30 11:28:32] Audio is at 11772 [Jun 30 11:28:32] Adding codec ulaw to SDP [Jun 30 11:28:32] Adding codec alaw to SDP [Jun 30 11:28:32] Adding codec gsm to SDP [Jun 30 11:28:32] Reliably Transmitting (no NAT) to 10.34.1.101:5060: [Jun 30 11:28:32] INVITE sip:je5qtq@10.34.1.101:5060;transport=udp SIP/2.0 [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK0e94368f [Jun 30 11:28:32] Max-Forwards: 70 [Jun 30 11:28:32] From: ;tag=as66b808db [Jun 30 11:28:32] To: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:32] Contact: [Jun 30 11:28:32] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:32] CSeq: 102 INVITE [Jun 30 11:28:32] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:32] Supported: replaces, timer [Jun 30 11:28:32] X-asterisk-Info: SIP re-invite (External RTP bridge) [Jun 30 11:28:32] Content-Type: application/sdp [Jun 30 11:28:32] Content-Length: 241 [Jun 30 11:28:32] [Jun 30 11:28:32] v=0 [Jun 30 11:28:32] o=root 1248309523 1248309524 IN IP4 10.34.1.168 [Jun 30 11:28:32] s=Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] c=IN IP4 10.34.1.168 [Jun 30 11:28:32] t=0 0 [Jun 30 11:28:32] m=audio 3000 RTP/AVP 0 8 3 [Jun 30 11:28:32] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:32] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:32] a=rtpmap:3 GSM/8000 [Jun 30 11:28:32] a=maxptime:150 [Jun 30 11:28:32] a=sendrecv [Jun 30 11:28:32] [Jun 30 11:28:32] --- [Jun 30 11:28:32] [Jun 30 11:28:32] <--- SIP read from UDP:10.34.1.168:5060 ---> [Jun 30 11:28:32] SIP/2.0 200 OK [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK17e9655b [Jun 30 11:28:32] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:32] To: ;tag=2639294411 [Jun 30 11:28:32] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:32] CSeq: 103 INVITE [Jun 30 11:28:32] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:32] Allow-Events: talk, hold, conference, LocalModeStatus [Jun 30 11:28:32] Contact: "Bob" ;+sip.instance="" [Jun 30 11:28:32] Server: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:32] Supported: path, replaces [Jun 30 11:28:32] Content-Type: application/sdp [Jun 30 11:28:32] Content-Length: 187 [Jun 30 11:28:32] [Jun 30 11:28:32] v=0 [Jun 30 11:28:32] o=MxSIP 0 2 IN IP4 10.34.1.168 [Jun 30 11:28:32] s=SIP Call [Jun 30 11:28:32] c=IN IP4 10.34.1.168 [Jun 30 11:28:32] t=0 0 [Jun 30 11:28:32] m=audio 3000 RTP/AVP 0 8 [Jun 30 11:28:32] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:32] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:32] a=silenceSupp:off - - - - [Jun 30 11:28:32] a=sendrecv [Jun 30 11:28:32] <-------------> [Jun 30 11:28:32] --- (13 headers 10 lines) --- [Jun 30 11:28:32] Found RTP audio format 0 [Jun 30 11:28:32] Found RTP audio format 8 [Jun 30 11:28:32] Found audio description format PCMU for ID 0 [Jun 30 11:28:32] Found audio description format PCMA for ID 8 [Jun 30 11:28:32] Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Jun 30 11:28:32] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jun 30 11:28:32] Peer audio RTP is at port 10.34.1.168:3000 [Jun 30 11:28:32] set_destination: Parsing for address/port to send to [Jun 30 11:28:32] set_destination: set destination to 10.34.1.168:5060 [Jun 30 11:28:32] Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:32] ACK sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK5900ad4f [Jun 30 11:28:32] Max-Forwards: 70 [Jun 30 11:28:32] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:32] To: ;tag=2639294411 [Jun 30 11:28:32] Contact: [Jun 30 11:28:32] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:32] CSeq: 103 ACK [Jun 30 11:28:32] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] Content-Length: 0 [Jun 30 11:28:32] [Jun 30 11:28:32] [Jun 30 11:28:32] --- [Jun 30 11:28:32] [Jun 30 11:28:32] <--- SIP read from UDP:10.34.1.101:5060 ---> [Jun 30 11:28:32] SIP/2.0 200 OK [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK0e94368f [Jun 30 11:28:32] From: ;tag=as66b808db [Jun 30 11:28:32] To: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:32] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:32] CSeq: 102 INVITE [Jun 30 11:28:32] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:32] Allow-Events: aastra-xml, talk, hold, conference, LocalModeStatus [Jun 30 11:28:32] Contact: "Alice" ;+sip.instance="" [Jun 30 11:28:32] Server: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:32] Supported: path, replaces [Jun 30 11:28:32] Content-Type: application/sdp [Jun 30 11:28:32] Content-Length: 187 [Jun 30 11:28:32] [Jun 30 11:28:32] v=0 [Jun 30 11:28:32] o=MxSIP 0 2 IN IP4 10.34.1.101 [Jun 30 11:28:32] s=SIP Call [Jun 30 11:28:32] c=IN IP4 10.34.1.101 [Jun 30 11:28:32] t=0 0 [Jun 30 11:28:32] m=audio 3000 RTP/AVP 0 8 [Jun 30 11:28:32] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:32] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:32] a=silenceSupp:off - - - - [Jun 30 11:28:32] a=sendrecv [Jun 30 11:28:32] <-------------> [Jun 30 11:28:32] --- (13 headers 10 lines) --- [Jun 30 11:28:32] Found RTP audio format 0 [Jun 30 11:28:32] Found RTP audio format 8 [Jun 30 11:28:32] Found audio description format PCMU for ID 0 [Jun 30 11:28:32] Found audio description format PCMA for ID 8 [Jun 30 11:28:32] Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Jun 30 11:28:32] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jun 30 11:28:32] Peer audio RTP is at port 10.34.1.101:3000 [Jun 30 11:28:32] set_destination: Parsing for address/port to send to [Jun 30 11:28:32] set_destination: set destination to 10.34.1.101:5060 [Jun 30 11:28:32] Transmitting (no NAT) to 10.34.1.101:5060: [Jun 30 11:28:32] ACK sip:je5qtq@10.34.1.101:5060;transport=udp SIP/2.0 [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK1bc0c7de [Jun 30 11:28:32] Max-Forwards: 70 [Jun 30 11:28:32] From: ;tag=as66b808db [Jun 30 11:28:32] To: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:32] Contact: [Jun 30 11:28:32] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:32] CSeq: 102 ACK [Jun 30 11:28:32] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] Content-Length: 0 [Jun 30 11:28:32] [Jun 30 11:28:32] [Jun 30 11:28:32] --- [Jun 30 11:28:32] set_destination: Parsing for address/port to send to [Jun 30 11:28:32] set_destination: set destination to 10.34.1.168:5060 [Jun 30 11:28:32] Audio is at 18484 [Jun 30 11:28:32] Adding codec ulaw to SDP [Jun 30 11:28:32] Adding codec alaw to SDP [Jun 30 11:28:32] Adding codec gsm to SDP [Jun 30 11:28:32] Reliably Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:32] INVITE sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK01686dbf [Jun 30 11:28:32] Max-Forwards: 70 [Jun 30 11:28:32] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:32] To: ;tag=2639294411 [Jun 30 11:28:32] Contact: [Jun 30 11:28:32] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:32] CSeq: 104 INVITE [Jun 30 11:28:32] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:32] Supported: replaces, timer [Jun 30 11:28:32] X-asterisk-Info: SIP re-invite (External RTP bridge) [Jun 30 11:28:32] Content-Type: application/sdp [Jun 30 11:28:32] Content-Length: 239 [Jun 30 11:28:32] [Jun 30 11:28:32] v=0 [Jun 30 11:28:32] o=root 602169165 602169167 IN IP4 10.34.1.101 [Jun 30 11:28:32] s=Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] c=IN IP4 10.34.1.101 [Jun 30 11:28:32] t=0 0 [Jun 30 11:28:32] m=audio 3000 RTP/AVP 0 8 3 [Jun 30 11:28:32] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:32] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:32] a=rtpmap:3 GSM/8000 [Jun 30 11:28:32] a=maxptime:150 [Jun 30 11:28:32] a=sendrecv [Jun 30 11:28:32] [Jun 30 11:28:32] --- [Jun 30 11:28:32] [Jun 30 11:28:32] <--- SIP read from UDP:10.34.1.168:5060 ---> [Jun 30 11:28:32] SIP/2.0 200 OK [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK01686dbf [Jun 30 11:28:32] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:32] To: ;tag=2639294411 [Jun 30 11:28:32] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:32] CSeq: 104 INVITE [Jun 30 11:28:32] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:32] Allow-Events: talk, hold, conference, LocalModeStatus [Jun 30 11:28:32] Contact: "Bob" ;+sip.instance="" [Jun 30 11:28:32] Server: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:32] Supported: path, replaces [Jun 30 11:28:32] Content-Type: application/sdp [Jun 30 11:28:32] Content-Length: 187 [Jun 30 11:28:32] [Jun 30 11:28:32] v=0 [Jun 30 11:28:32] o=MxSIP 0 3 IN IP4 10.34.1.168 [Jun 30 11:28:32] s=SIP Call [Jun 30 11:28:32] c=IN IP4 10.34.1.168 [Jun 30 11:28:32] t=0 0 [Jun 30 11:28:32] m=audio 3000 RTP/AVP 0 8 [Jun 30 11:28:32] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:32] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:32] a=silenceSupp:off - - - - [Jun 30 11:28:32] a=sendrecv [Jun 30 11:28:32] <-------------> [Jun 30 11:28:32] --- (13 headers 10 lines) --- [Jun 30 11:28:32] Found RTP audio format 0 [Jun 30 11:28:32] Found RTP audio format 8 [Jun 30 11:28:32] Found audio description format PCMU for ID 0 [Jun 30 11:28:32] Found audio description format PCMA for ID 8 [Jun 30 11:28:32] Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Jun 30 11:28:32] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jun 30 11:28:32] Peer audio RTP is at port 10.34.1.168:3000 [Jun 30 11:28:32] set_destination: Parsing for address/port to send to [Jun 30 11:28:32] set_destination: set destination to 10.34.1.168:5060 [Jun 30 11:28:32] Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:32] ACK sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:32] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK4934023b [Jun 30 11:28:32] Max-Forwards: 70 [Jun 30 11:28:32] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:32] To: ;tag=2639294411 [Jun 30 11:28:32] Contact: [Jun 30 11:28:32] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:32] CSeq: 104 ACK [Jun 30 11:28:32] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:32] Content-Length: 0 [Jun 30 11:28:32] [Jun 30 11:28:32] [Jun 30 11:28:32] --- [Jun 30 11:28:35] [Jun 30 11:28:35] <--- SIP read from UDP:10.34.1.101:5060 ---> [Jun 30 11:28:35] INFO sip:145@10.34.1.11:5060 SIP/2.0 [Jun 30 11:28:35] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK6141d6901458f8394 [Jun 30 11:28:35] Max-Forwards: 70 [Jun 30 11:28:35] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:35] To: ;tag=as66b808db [Jun 30 11:28:35] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:35] CSeq: 18690 INFO [Jun 30 11:28:35] Authorization: Digest username="je5qtq",realm="asterisk",nonce="2dc0228a",uri="sip:145@10.34.1.11:5060",response="e24e6b3aa873605a9220591809c75839",algorithm=MD5 [Jun 30 11:28:35] User-Agent: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:35] Content-Type: application/dtmf-relay [Jun 30 11:28:35] Content-Length: 24 [Jun 30 11:28:35] [Jun 30 11:28:35] Signal=1 [Jun 30 11:28:35] Duration=160 [Jun 30 11:28:35] <-------------> [Jun 30 11:28:35] --- (11 headers 2 lines) --- [Jun 30 11:28:35] Receiving INFO! [Jun 30 11:28:35] * DTMF-relay event received: 1 [Jun 30 11:28:35] [Jun 30 11:28:35] <--- Transmitting (no NAT) to 10.34.1.101:5060 ---> [Jun 30 11:28:35] SIP/2.0 200 OK [Jun 30 11:28:35] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK6141d6901458f8394;received=10.34.1.101 [Jun 30 11:28:35] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:35] To: ;tag=as66b808db [Jun 30 11:28:35] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:35] CSeq: 18690 INFO [Jun 30 11:28:35] Server: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:35] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:35] Supported: replaces, timer [Jun 30 11:28:35] Content-Length: 0 [Jun 30 11:28:35] [Jun 30 11:28:35] [Jun 30 11:28:35] <------------> [Jun 30 11:28:35] DTMF[12718][C-00000002]: channel.c:3953 __ast_read: DTMF end '1' received on SIP/je5qtq-00000004, duration 160 ms [Jun 30 11:28:35] DTMF[12718][C-00000002]: channel.c:3980 __ast_read: DTMF begin emulation of '1' with duration 160 queued on SIP/je5qtq-00000004 [Jun 30 11:28:38] [Jun 30 11:28:38] <--- SIP read from UDP:10.34.1.101:5060 ---> [Jun 30 11:28:38] INFO sip:145@10.34.1.11:5060 SIP/2.0 [Jun 30 11:28:38] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK049b06366e9dd9596 [Jun 30 11:28:38] Max-Forwards: 70 [Jun 30 11:28:38] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:38] To: ;tag=as66b808db [Jun 30 11:28:38] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:38] CSeq: 18691 INFO [Jun 30 11:28:38] Authorization: Digest username="je5qtq",realm="asterisk",nonce="2dc0228a",uri="sip:145@10.34.1.11:5060",response="e24e6b3aa873605a9220591809c75839",algorithm=MD5 [Jun 30 11:28:38] User-Agent: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:38] Content-Type: application/dtmf-relay [Jun 30 11:28:38] Content-Length: 24 [Jun 30 11:28:38] [Jun 30 11:28:38] Signal=2 [Jun 30 11:28:38] Duration=160 [Jun 30 11:28:38] <-------------> [Jun 30 11:28:38] --- (11 headers 2 lines) --- [Jun 30 11:28:38] Receiving INFO! [Jun 30 11:28:38] * DTMF-relay event received: 2 [Jun 30 11:28:38] [Jun 30 11:28:38] <--- Transmitting (no NAT) to 10.34.1.101:5060 ---> [Jun 30 11:28:38] SIP/2.0 200 OK [Jun 30 11:28:38] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK049b06366e9dd9596;received=10.34.1.101 [Jun 30 11:28:38] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:38] To: ;tag=as66b808db [Jun 30 11:28:38] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:38] CSeq: 18691 INFO [Jun 30 11:28:38] Server: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:38] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:38] Supported: replaces, timer [Jun 30 11:28:38] Content-Length: 0 [Jun 30 11:28:38] [Jun 30 11:28:38] [Jun 30 11:28:38] <------------> [Jun 30 11:28:38] DTMF[12718][C-00000002]: channel.c:4073 __ast_read: DTMF end emulation of '1' queued on SIP/je5qtq-00000004 [Jun 30 11:28:38] DTMF[12718][C-00000002]: channel.c:3953 __ast_read: DTMF end '2' received on SIP/je5qtq-00000004, duration 160 ms [Jun 30 11:28:38] DTMF[12718][C-00000002]: channel.c:3980 __ast_read: DTMF begin emulation of '2' with duration 160 queued on SIP/je5qtq-00000004 [Jun 30 11:28:38] set_destination: Parsing for address/port to send to [Jun 30 11:28:38] set_destination: set destination to 10.34.1.168:5060 [Jun 30 11:28:38] Reliably Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:38] INFO sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:38] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK6a603c43 [Jun 30 11:28:38] Max-Forwards: 70 [Jun 30 11:28:38] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:38] To: ;tag=2639294411 [Jun 30 11:28:38] Contact: [Jun 30 11:28:38] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:38] CSeq: 105 INFO [Jun 30 11:28:38] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:38] Content-Type: application/dtmf-relay [Jun 30 11:28:38] Content-Length: 25 [Jun 30 11:28:38] [Jun 30 11:28:38] Signal=1 [Jun 30 11:28:38] Duration=2796 [Jun 30 11:28:38] [Jun 30 11:28:38] --- [Jun 30 11:28:38] [Jun 30 11:28:38] <--- SIP read from UDP:10.34.1.168:5060 ---> [Jun 30 11:28:38] SIP/2.0 200 OK [Jun 30 11:28:38] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK6a603c43 [Jun 30 11:28:38] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:38] To: ;tag=2639294411 [Jun 30 11:28:38] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:38] CSeq: 105 INFO [Jun 30 11:28:38] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:38] Allow-Events: talk, hold, conference, LocalModeStatus [Jun 30 11:28:38] Server: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:38] Supported: path [Jun 30 11:28:38] Content-Length: 0 [Jun 30 11:28:38] [Jun 30 11:28:38] <-------------> [Jun 30 11:28:38] --- (11 headers 0 lines) --- [Jun 30 11:28:41] [Jun 30 11:28:41] <--- SIP read from UDP:10.34.1.101:5060 ---> [Jun 30 11:28:41] INFO sip:145@10.34.1.11:5060 SIP/2.0 [Jun 30 11:28:41] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK9a50f5f2c7d3bfb4b [Jun 30 11:28:41] Max-Forwards: 70 [Jun 30 11:28:41] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:41] To: ;tag=as66b808db [Jun 30 11:28:41] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:41] CSeq: 18692 INFO [Jun 30 11:28:41] Authorization: Digest username="je5qtq",realm="asterisk",nonce="2dc0228a",uri="sip:145@10.34.1.11:5060",response="e24e6b3aa873605a9220591809c75839",algorithm=MD5 [Jun 30 11:28:41] User-Agent: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:41] Content-Type: application/dtmf-relay [Jun 30 11:28:41] Content-Length: 24 [Jun 30 11:28:41] [Jun 30 11:28:41] Signal=3 [Jun 30 11:28:41] Duration=160 [Jun 30 11:28:41] <-------------> [Jun 30 11:28:41] --- (11 headers 2 lines) --- [Jun 30 11:28:41] Receiving INFO! [Jun 30 11:28:41] * DTMF-relay event received: 3 [Jun 30 11:28:41] [Jun 30 11:28:41] <--- Transmitting (no NAT) to 10.34.1.101:5060 ---> [Jun 30 11:28:41] SIP/2.0 200 OK [Jun 30 11:28:41] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK9a50f5f2c7d3bfb4b;received=10.34.1.101 [Jun 30 11:28:41] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:41] To: ;tag=as66b808db [Jun 30 11:28:41] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:41] CSeq: 18692 INFO [Jun 30 11:28:41] Server: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:41] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:41] Supported: replaces, timer [Jun 30 11:28:41] Content-Length: 0 [Jun 30 11:28:41] [Jun 30 11:28:41] [Jun 30 11:28:41] <------------> [Jun 30 11:28:41] DTMF[12718][C-00000002]: channel.c:4073 __ast_read: DTMF end emulation of '2' queued on SIP/je5qtq-00000004 [Jun 30 11:28:41] DTMF[12718][C-00000002]: channel.c:3953 __ast_read: DTMF end '3' received on SIP/je5qtq-00000004, duration 160 ms [Jun 30 11:28:41] DTMF[12718][C-00000002]: channel.c:3980 __ast_read: DTMF begin emulation of '3' with duration 160 queued on SIP/je5qtq-00000004 [Jun 30 11:28:41] set_destination: Parsing for address/port to send to [Jun 30 11:28:41] set_destination: set destination to 10.34.1.168:5060 [Jun 30 11:28:41] Reliably Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:41] INFO sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:41] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK1d2d0e96 [Jun 30 11:28:41] Max-Forwards: 70 [Jun 30 11:28:41] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:41] To: ;tag=2639294411 [Jun 30 11:28:41] Contact: [Jun 30 11:28:41] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:41] CSeq: 106 INFO [Jun 30 11:28:41] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:41] Content-Type: application/dtmf-relay [Jun 30 11:28:41] Content-Length: 25 [Jun 30 11:28:41] [Jun 30 11:28:41] Signal=2 [Jun 30 11:28:41] Duration=3162 [Jun 30 11:28:41] [Jun 30 11:28:41] --- [Jun 30 11:28:41] [Jun 30 11:28:41] <--- SIP read from UDP:10.34.1.168:5060 ---> [Jun 30 11:28:41] SIP/2.0 200 OK [Jun 30 11:28:41] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK1d2d0e96 [Jun 30 11:28:41] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:41] To: ;tag=2639294411 [Jun 30 11:28:41] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:41] CSeq: 106 INFO [Jun 30 11:28:41] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:41] Allow-Events: talk, hold, conference, LocalModeStatus [Jun 30 11:28:41] Server: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:41] Supported: path [Jun 30 11:28:41] Content-Length: 0 [Jun 30 11:28:41] [Jun 30 11:28:41] <-------------> [Jun 30 11:28:41] --- (11 headers 0 lines) --- [Jun 30 11:28:45] [Jun 30 11:28:45] <--- SIP read from UDP:10.34.1.101:5060 ---> [Jun 30 11:28:45] BYE sip:145@10.34.1.11:5060 SIP/2.0 [Jun 30 11:28:45] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK974cf291f5a9ad39a [Jun 30 11:28:45] Max-Forwards: 70 [Jun 30 11:28:45] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:45] To: ;tag=as66b808db [Jun 30 11:28:45] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:45] CSeq: 18693 BYE [Jun 30 11:28:45] Authorization: Digest username="je5qtq",realm="asterisk",nonce="2dc0228a",uri="sip:145@10.34.1.11:5060",response="2c6cdc93d83d85e750e533f580ad4dfc",algorithm=MD5 [Jun 30 11:28:45] User-Agent: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:45] Content-Length: 0 [Jun 30 11:28:45] [Jun 30 11:28:45] <-------------> [Jun 30 11:28:45] --- (10 headers 0 lines) --- [Jun 30 11:28:45] Sending to 10.34.1.101:5060 (no NAT) [Jun 30 11:28:45] Scheduling destruction of SIP dialog '42b2ea4fa1b0ed52' in 32000 ms (Method: BYE) [Jun 30 11:28:45] [Jun 30 11:28:45] <--- Transmitting (no NAT) to 10.34.1.101:5060 ---> [Jun 30 11:28:45] SIP/2.0 200 OK [Jun 30 11:28:45] Via: SIP/2.0/UDP 10.34.1.101:5060;branch=z9hG4bK974cf291f5a9ad39a;received=10.34.1.101 [Jun 30 11:28:45] From: "Alice" ;tag=68a0ec1b70 [Jun 30 11:28:45] To: ;tag=as66b808db [Jun 30 11:28:45] Call-ID: 42b2ea4fa1b0ed52 [Jun 30 11:28:45] CSeq: 18693 BYE [Jun 30 11:28:45] Server: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:45] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:45] Supported: replaces, timer [Jun 30 11:28:45] Content-Length: 0 [Jun 30 11:28:45] [Jun 30 11:28:45] [Jun 30 11:28:45] <------------> [Jun 30 11:28:45] -- Channel SIP/je5qtq-00000004 left 'native_rtp' basic-bridge <90c51b22-c746-44ee-811e-53346db1f05e> [Jun 30 11:28:45] set_destination: Parsing for address/port to send to [Jun 30 11:28:45] set_destination: set destination to 10.34.1.168:5060 [Jun 30 11:28:45] Audio is at 18484 [Jun 30 11:28:45] Adding codec ulaw to SDP [Jun 30 11:28:45] Adding codec alaw to SDP [Jun 30 11:28:45] Adding codec gsm to SDP [Jun 30 11:28:45] Reliably Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:45] INVITE sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:45] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK7b34346c [Jun 30 11:28:45] Max-Forwards: 70 [Jun 30 11:28:45] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:45] To: ;tag=2639294411 [Jun 30 11:28:45] Contact: [Jun 30 11:28:45] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:45] CSeq: 107 INVITE [Jun 30 11:28:45] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:45] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 30 11:28:45] Supported: replaces, timer [Jun 30 11:28:45] X-asterisk-Info: SIP re-invite (External RTP bridge) [Jun 30 11:28:45] Content-Type: application/sdp [Jun 30 11:28:45] Content-Length: 238 [Jun 30 11:28:45] [Jun 30 11:28:45] v=0 [Jun 30 11:28:45] o=root 602169165 602169168 IN IP4 10.34.1.11 [Jun 30 11:28:45] s=Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:45] c=IN IP4 10.34.1.11 [Jun 30 11:28:45] t=0 0 [Jun 30 11:28:45] m=audio 18484 RTP/AVP 0 8 3 [Jun 30 11:28:45] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:45] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:45] a=rtpmap:3 GSM/8000 [Jun 30 11:28:45] a=maxptime:150 [Jun 30 11:28:45] a=sendrecv [Jun 30 11:28:45] [Jun 30 11:28:45] --- [Jun 30 11:28:45] DTMF[12718][C-00000002]: bridge_channel.c:687 bridge_channel_settle_owed_events: DTMF end '3' simulated to bridge 90c51b22-c746-44ee-811e-53346db1f05e because SIP/je5qtq-00000004 left. Duration 3577 ms. [Jun 30 11:28:45] == Spawn extension (default, 145, 2) exited non-zero on 'SIP/je5qtq-00000004' [Jun 30 11:28:45] -- Channel SIP/as2mkq-00000005 left 'native_rtp' basic-bridge <90c51b22-c746-44ee-811e-53346db1f05e> [Jun 30 11:28:45] Scheduling destruction of SIP dialog '7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060' in 32000 ms (Method: INVITE) [Jun 30 11:28:45] [Jun 30 11:28:45] <--- SIP read from UDP:10.34.1.168:5060 ---> [Jun 30 11:28:45] SIP/2.0 200 OK [Jun 30 11:28:45] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK7b34346c [Jun 30 11:28:45] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:45] To: ;tag=2639294411 [Jun 30 11:28:45] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:45] CSeq: 107 INVITE [Jun 30 11:28:45] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:45] Allow-Events: talk, hold, conference, LocalModeStatus [Jun 30 11:28:45] Contact: "Bob" ;+sip.instance="" [Jun 30 11:28:45] Server: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:45] Supported: path, replaces [Jun 30 11:28:45] Content-Type: application/sdp [Jun 30 11:28:45] Content-Length: 187 [Jun 30 11:28:45] [Jun 30 11:28:45] v=0 [Jun 30 11:28:45] o=MxSIP 0 4 IN IP4 10.34.1.168 [Jun 30 11:28:45] s=SIP Call [Jun 30 11:28:45] c=IN IP4 10.34.1.168 [Jun 30 11:28:45] t=0 0 [Jun 30 11:28:45] m=audio 3000 RTP/AVP 0 8 [Jun 30 11:28:45] a=rtpmap:0 PCMU/8000 [Jun 30 11:28:45] a=rtpmap:8 PCMA/8000 [Jun 30 11:28:45] a=silenceSupp:off - - - - [Jun 30 11:28:45] a=sendrecv [Jun 30 11:28:45] <-------------> [Jun 30 11:28:45] --- (13 headers 10 lines) --- [Jun 30 11:28:45] Found RTP audio format 0 [Jun 30 11:28:45] Found RTP audio format 8 [Jun 30 11:28:45] Found audio description format PCMU for ID 0 [Jun 30 11:28:45] Found audio description format PCMA for ID 8 [Jun 30 11:28:45] Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Jun 30 11:28:45] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jun 30 11:28:45] Peer audio RTP is at port 10.34.1.168:3000 [Jun 30 11:28:45] set_destination: Parsing for address/port to send to [Jun 30 11:28:45] set_destination: set destination to 10.34.1.168:5060 [Jun 30 11:28:45] Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:45] ACK sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:45] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK31c492e0 [Jun 30 11:28:45] Max-Forwards: 70 [Jun 30 11:28:45] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:45] To: ;tag=2639294411 [Jun 30 11:28:45] Contact: [Jun 30 11:28:45] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:45] CSeq: 107 ACK [Jun 30 11:28:45] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:45] Content-Length: 0 [Jun 30 11:28:45] [Jun 30 11:28:45] [Jun 30 11:28:45] --- [Jun 30 11:28:45] set_destination: Parsing for address/port to send to [Jun 30 11:28:45] set_destination: set destination to 10.34.1.168:5060 [Jun 30 11:28:45] Reliably Transmitting (no NAT) to 10.34.1.168:5060: [Jun 30 11:28:45] BYE sip:as2mkq@10.34.1.168:5060;transport=udp SIP/2.0 [Jun 30 11:28:45] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK62849981 [Jun 30 11:28:45] Max-Forwards: 70 [Jun 30 11:28:45] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:45] To: ;tag=2639294411 [Jun 30 11:28:45] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:45] CSeq: 108 BYE [Jun 30 11:28:45] User-Agent: Asterisk PBX 13.4.0~20150629.1425 [Jun 30 11:28:45] X-Asterisk-HangupCause: Normal Clearing [Jun 30 11:28:45] X-Asterisk-HangupCauseCode: 16 [Jun 30 11:28:45] Content-Length: 0 [Jun 30 11:28:45] [Jun 30 11:28:45] [Jun 30 11:28:45] --- [Jun 30 11:28:45] Scheduling destruction of SIP dialog '7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060' in 32000 ms (Method: INVITE) [Jun 30 11:28:45] [Jun 30 11:28:45] <--- SIP read from UDP:10.34.1.168:5060 ---> [Jun 30 11:28:45] SIP/2.0 200 OK [Jun 30 11:28:45] Via: SIP/2.0/UDP 10.34.1.11:5060;branch=z9hG4bK62849981 [Jun 30 11:28:45] From: "Alice" ;tag=as2cbfa82b [Jun 30 11:28:45] To: ;tag=2639294411 [Jun 30 11:28:45] Call-ID: 7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060 [Jun 30 11:28:45] CSeq: 108 BYE [Jun 30 11:28:45] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO [Jun 30 11:28:45] Allow-Events: talk, hold, conference, LocalModeStatus [Jun 30 11:28:45] Server: Aastra 6731i/3.3.1.4295 [Jun 30 11:28:45] Supported: path [Jun 30 11:28:45] Content-Length: 0 [Jun 30 11:28:45] [Jun 30 11:28:45] <-------------> [Jun 30 11:28:45] --- (11 headers 0 lines) --- [Jun 30 11:28:45] Really destroying SIP dialog '7dedfc1a414fa6bc3602fedd7c91d82c@10.34.1.11:5060' Method: INVITE