-- Executing [1234@locutorios:1] Dial("SIP/1236-00000002", "SIP/1234||tT") in new stack -- Called 1234 -- SIP/1234-00000003 is ringing -- SIP/1234-00000003 answered SIP/1236-00000002 debian*CLI> sip set debug SIP Debugging enabled debian*CLI> <--- SIP read from 10.1.3.16:5060 ---> INFO sip:1236@10.1.2.130 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-8e681b59 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 101 INFO Max-Forwards: 70 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 24 Content-Type: application/dtmf-relay Signal=* Duration=100 <-------------> --- (10 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * <--- Transmitting (no NAT) to 10.1.3.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-8e681b59;received=10.1.3.16 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 101 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Mar 29 12:06:31] DTMF[2312]: channel.c:2303 __ast_read: DTMF end '*' received on SIP/1234-00000003, duration 100 ms [Mar 29 12:06:31] DTMF[2312]: channel.c:2329 __ast_read: DTMF begin emulation of '*' with duration 100 queued on SIP/1234-00000003 [Mar 29 12:06:31] DTMF[2312]: channel.c:2448 __ast_read: DTMF end emulation of '*' queued on SIP/1234-00000003 debian*CLI> <--- SIP read from 10.1.3.16:5060 ---> INFO sip:1236@10.1.2.130 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-a98f53a4 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 102 INFO Max-Forwards: 70 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 24 Content-Type: application/dtmf-relay Signal=2 Duration=100 <-------------> --- (10 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 2 debian*CLI> <--- Transmitting (no NAT) to 10.1.3.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-a98f53a4;received=10.1.3.16 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 102 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Mar 29 12:06:32] DTMF[2312]: channel.c:2303 __ast_read: DTMF end '2' received on SIP/1234-00000003, duration 100 ms [Mar 29 12:06:32] DTMF[2312]: channel.c:2329 __ast_read: DTMF begin emulation of '2' with duration 100 queued on SIP/1234-00000003 [Mar 29 12:06:32] DTMF[2312]: channel.c:2448 __ast_read: DTMF end emulation of '2' queued on SIP/1234-00000003 -- Started music on hold, class 'default', on SIP/1236-00000002 -- Playing 'pbx-transfer' (language 'en') debian*CLI> <--- SIP read from 10.1.3.16:5060 ---> INFO sip:1236@10.1.2.130 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-c686ff5c From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 103 INFO Max-Forwards: 70 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 24 Content-Type: application/dtmf-relay Signal=1 Duration=100 <-------------> --- (10 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 1 <--- Transmitting (no NAT) to 10.1.3.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-c686ff5c;received=10.1.3.16 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 103 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Mar 29 12:06:35] DTMF[2312]: channel.c:2303 __ast_read: DTMF end '1' received on SIP/1234-00000003, duration 100 ms [Mar 29 12:06:35] DTMF[2312]: channel.c:2356 __ast_read: DTMF end passthrough '1' on SIP/1234-00000003 debian*CLI> <--- SIP read from 10.1.3.16:5060 ---> INFO sip:1236@10.1.2.130 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-efc8ef9c From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 104 INFO Max-Forwards: 70 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 24 Content-Type: application/dtmf-relay Signal=2 Duration=100 <-------------> --- (10 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 2 <--- Transmitting (no NAT) to 10.1.3.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-efc8ef9c;received=10.1.3.16 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 104 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Mar 29 12:06:35] DTMF[2312]: channel.c:2303 __ast_read: DTMF end '2' received on SIP/1234-00000003, duration 100 ms [Mar 29 12:06:35] DTMF[2312]: channel.c:2356 __ast_read: DTMF end passthrough '2' on SIP/1234-00000003 debian*CLI> <--- SIP read from 10.1.3.16:5060 ---> INFO sip:1236@10.1.2.130 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-180de069 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 105 INFO Max-Forwards: 70 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 24 Content-Type: application/dtmf-relay Signal=3 Duration=100 <-------------> --- (10 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 3 <--- Transmitting (no NAT) to 10.1.3.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-180de069;received=10.1.3.16 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 105 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Mar 29 12:06:36] DTMF[2312]: channel.c:2303 __ast_read: DTMF end '3' received on SIP/1234-00000003, duration 100 ms [Mar 29 12:06:36] DTMF[2312]: channel.c:2356 __ast_read: DTMF end passthrough '3' on SIP/1234-00000003 debian*CLI> <--- SIP read from 10.1.3.16:5060 ---> INFO sip:1236@10.1.2.130 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-f320ecaa From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 106 INFO Max-Forwards: 70 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 24 Content-Type: application/dtmf-relay Signal=5 Duration=100 <-------------> --- (10 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 5 <--- Transmitting (no NAT) to 10.1.3.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-f320ecaa;received=10.1.3.16 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 106 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Mar 29 12:06:36] DTMF[2312]: channel.c:2303 __ast_read: DTMF end '5' received on SIP/1234-00000003, duration 100 ms [Mar 29 12:06:36] DTMF[2312]: channel.c:2356 __ast_read: DTMF end passthrough '5' on SIP/1234-00000003 -- Executing [1235@locutorios:1] Dial("Local/1235@locutorios-4582,2", "SIP/1235||tT") in new stack Audio is at 10.1.2.130 port 15990 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.1.2.16:5061: INVITE sip:1235@10.1.2.16:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.2.130:5060;branch=z9hG4bK651ddcea;rport From: "1234" ;tag=as3860cda1 To: Contact: Call-ID: 3abaa2bf2d729b111b9444be10a63cb0@10.1.2.130 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 29 Mar 2010 10:06:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 234 v=0 o=root 2121 2121 IN IP4 10.1.2.130 s=session c=IN IP4 10.1.2.130 t=0 0 m=audio 15990 RTP/AVP 8 101 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 1235 debian*CLI> <--- SIP read from 10.1.2.16:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.2.130:5060;received=10.1.2.130;rport=5060;branch=z9hG4bK651ddcea To: From: "1234" ;tag=as3860cda1 Call-ID: 3abaa2bf2d729b111b9444be10a63cb0@10.1.2.130 CSeq: 102 INVITE Server: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- debian*CLI> <--- SIP read from 10.1.2.16:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.2.130:5060;received=10.1.2.130;rport=5060;branch=z9hG4bK651ddcea To: ;tag=kxrvl From: "1234" ;tag=as3860cda1 Call-ID: 3abaa2bf2d729b111b9444be10a63cb0@10.1.2.130 CSeq: 102 INVITE Contact: Server: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/1235-00000004 is ringing -- Local/1235@locutorios-4582,1 is ringing debian*CLI> <--- SIP read from 10.1.3.16:5060 ---> BYE sip:1236@10.1.2.130 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-5aec6325 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 107 BYE Max-Forwards: 70 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 10.1.3.16 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.1.3.16:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.3.16:5060;branch=z9hG4bK-5aec6325;received=10.1.3.16 From: ;tag=49677863dbe01918i1 To: "1236" ;tag=as55a63466 Call-ID: 147b2548397050fa6b4a42a47ec7fc00@10.1.2.130 CSeq: 107 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> debian*CLI> <--- SIP read from 10.1.3.15:5061 ---> BYE sip:1234@10.1.2.130 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-11685bd3 From: "1236" ;tag=a46698ddd93cd999o1 To: "1234" ;tag=as2b7c8616 Call-ID: 650bad2d-965a5189@10.1.3.15 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="1236",realm="asterisk",nonce="3bc4753b",uri="sip:1234@10.1.2.130",algorithm=MD5,response="9a3e458ab27fb1c73a3570c27078593e" User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.1.3.15 : 5061 (no NAT) <--- Transmitting (no NAT) to 10.1.3.15:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-11685bd3;received=10.1.3.15 From: "1236" ;tag=a46698ddd93cd999o1 To: "1234" ;tag=as2b7c8616 Call-ID: 650bad2d-965a5189@10.1.3.15 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> -- Stopped music on hold on SIP/1236-00000002 Really destroying SIP dialog '4d8c0b6a-27fd9018@10.1.3.15' Method: REGISTER debian*CLI> <--- SIP read from 10.1.2.16:5061 ---> SIP/2.0 603 Decline Via: SIP/2.0/UDP 10.1.2.130:5060;received=10.1.2.130;rport=5060;branch=z9hG4bK651ddcea To: ;tag=kxrvl From: "1234" ;tag=as3860cda1 Call-ID: 3abaa2bf2d729b111b9444be10a63cb0@10.1.2.130 CSeq: 102 INVITE Server: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- Got SIP response 603 "Decline" back from 10.1.2.16 Transmitting (no NAT) to 10.1.2.16:5061: ACK sip:1235@10.1.2.16:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.2.130:5060;branch=z9hG4bK651ddcea;rport From: "1234" ;tag=as3860cda1 To: ;tag=kxrvl Contact: Call-ID: 3abaa2bf2d729b111b9444be10a63cb0@10.1.2.130 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/1235-00000004 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing [1235@locutorios:2] Hangup("Local/1235@locutorios-4582,2", "") in new stack == Spawn extension (locutorios, 1235, 2) exited non-zero on 'Local/1235@locutorios-4582,2' == Spawn extension (locutorios, 1234, 1) exited non-zero on 'SIP/1236-00000002'