*CLI> *CLI> <--- SIP read from 202.83.183.33:5060 ---> INVITE sip:0290371449@202.83.176.38:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.183.33:5060;branch=z9hG4bK1ca53b72146c19f20-111-1 To: From: "0409227633" ;tag=t1187094304-co273 Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 73830 INVITE Max-Forwards: 70 P-Asserted-Identity: Contact: User-Agent: ENS2.2.95 Content-Type: application/sdp Content-Length: 263 v=0 o=- 4278193382 4278193382 IN IP4 202.83.183.33 s=ENS Session c=IN IP4 202.83.183.40 t=0 0 m=audio 8790 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (12 headers 12 lines) --- Sending to 202.83.183.33 : 5060 (no NAT) Using INVITE request as basis request - 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 Found peer 'sip-ivox' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 202.83.183.40:8790 Found description format PCMA for ID 8 Found description format PCMU for ID 0 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 202.83.183.40:8790 Looking for 0290371449 in barnet-from-internet (domain 202.83.176.38) list_route: hop: <--- Transmitting (no NAT) to 202.83.183.33:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 202.83.183.33:5060;branch=z9hG4bK1ca53b72146c19f20-111-1;received=202.83.183.33 From: "0409227633" ;tag=t1187094304-co273 To: Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 73830 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [0290371449@barnet-from-internet:1] Macro("SIP/5060-08c1bef0", "call-int-sip|9096@lvl04.barnet.com.au") in new stack -- Executing [s@macro-call-int-sip:1] NoOp("SIP/5060-08c1bef0", "9096@lvl04.barnet.com.au") in new stack -- Executing [s@macro-call-int-sip:2] Dial("SIP/5060-08c1bef0", "SIP/9096@lvl04.barnet.com.au||r") in new stack Audio is at 202.83.176.38 port 14670 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 10.252.11.130:5060: INVITE sip:9096@lvl04.barnet.com.au SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK30d7b9e9;rport From: "0409227633" ;tag=as685bacc3 To: Contact: Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 14 Aug 2007 12:25:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 289 v=0 o=root 13152 13152 IN IP4 202.83.176.38 s=session c=IN IP4 202.83.176.38 t=0 0 m=audio 14670 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 9096@lvl04.barnet.com.au <--- Transmitting (no NAT) to 202.83.183.33:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 202.83.183.33:5060;branch=z9hG4bK1ca53b72146c19f20-111-1;received=202.83.183.33 From: "0409227633" ;tag=t1187094304-co273 To: ;tag=as0e80d48c Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 73830 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 10.252.11.130:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK30d7b9e9;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Date: Tue, 14 Aug 2007 12:25:04 GMT Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.252.11.130:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK30d7b9e9;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Date: Tue, 14 Aug 2007 12:25:04 GMT Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: "Edwin Groothuis" ;party=called;screen=no;privacy=off Contact: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- -- SIP/lvl04.barnet.com.au-08c213d8 is ringing <--- SIP read from 10.252.11.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK30d7b9e9;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Date: Tue, 14 Aug 2007 12:25:04 GMT Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: "Edwin Groothuis" ;party=called;screen=yes;privacy=off Contact: Content-Type: application/sdp Content-Length: 229 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 10.252.11.130 s=SIP Call c=IN IP4 202.83.178.13 t=0 0 m=audio 17236 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 202.83.178.13:17236 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 202.83.178.13:17236 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.252.11.130, port 5060 Transmitting (no NAT) to 10.252.11.130:5060: ACK sip:9096@10.252.11.130:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK32152be8;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Contact: Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/lvl04.barnet.com.au-08c213d8 answered SIP/5060-08c1bef0 Audio is at 202.83.176.38 port 16464 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 202.83.183.33:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 202.83.183.33:5060;branch=z9hG4bK1ca53b72146c19f20-111-1;received=202.83.183.33 From: "0409227633" ;tag=t1187094304-co273 To: ;tag=as0e80d48c Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 73830 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 266 v=0 o=root 13152 13152 IN IP4 202.83.176.38 s=session c=IN IP4 202.83.176.38 t=0 0 m=audio 16464 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 - - - - a=ptime:20 a=sendrecv <------------> -- Native bridging SIP/5060-08c1bef0 and SIP/lvl04.barnet.com.au-08c213d8 set_destination: Parsing for address/port to send to set_destination: set destination to 10.252.11.130, port 5060 Audio is at 202.83.176.38 port 14670 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.252.11.130:5060: INVITE sip:9096@10.252.11.130:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK7d2ccc9c;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Contact: Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 241 v=0 o=root 13152 13153 IN IP4 202.83.183.40 s=session c=IN IP4 202.83.183.40 t=0 0 m=audio 8790 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 10.252.11.130:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK7d2ccc9c;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Date: Tue, 14 Aug 2007 12:25:07 GMT Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 103 INVITE Allow-Events: telephone-event Remote-Party-ID: "Edwin Groothuis" ;party=called;screen=yes;privacy=off Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.252.11.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK7d2ccc9c;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Date: Tue, 14 Aug 2007 12:25:07 GMT Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: "Edwin Groothuis" ;party=called;screen=no;privacy=off Contact: Content-Type: application/sdp Content-Length: 229 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 10.252.11.130 s=SIP Call c=IN IP4 202.83.178.13 t=0 0 m=audio 17236 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 202.83.178.13:17236 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 202.83.178.13:17236 set_destination: Parsing for address/port to send to set_destination: set destination to 10.252.11.130, port 5060 Transmitting (no NAT) to 10.252.11.130:5060: ACK sip:9096@10.252.11.130:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK591cd8ec;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Contact: Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 202.83.183.33:5060 ---> ACK sip:0290371449@202.83.176.38:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.183.33:5060;branch=z9hG4bK1ca53b72146c19f20-111-2 To: ;tag=as0e80d48c From: "0409227633" ;tag=t1187094304-co273 Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 73830 ACK Max-Forwards: 70 User-Agent: ENS2.2.95 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 202.83.183.33, port 5060 Audio is at 202.83.176.38 port 16464 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 202.83.183.33:5060: INVITE sip:0409227633@202.83.183.33:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK37a1e9f5;rport From: ;tag=as0e80d48c To: "0409227633" ;tag=t1187094304-co273 Contact: Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 13152 13153 IN IP4 202.83.178.13 s=session c=IN IP4 202.83.178.13 t=0 0 m=audio 17236 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #1 (no NAT) to 202.83.183.33:5060: INVITE sip:0409227633@202.83.183.33:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK37a1e9f5;rport From: ;tag=as0e80d48c To: "0409227633" ;tag=t1187094304-co273 Contact: Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 13152 13153 IN IP4 202.83.178.13 s=session c=IN IP4 202.83.178.13 t=0 0 m=audio 17236 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (no NAT) to 202.83.183.33:5060: INVITE sip:0409227633@202.83.183.33:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK37a1e9f5;rport From: ;tag=as0e80d48c To: "0409227633" ;tag=t1187094304-co273 Contact: Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 13152 13153 IN IP4 202.83.178.13 s=session c=IN IP4 202.83.178.13 t=0 0 m=audio 17236 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (no NAT) to 202.83.183.33:5060: INVITE sip:0409227633@202.83.183.33:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK37a1e9f5;rport From: ;tag=as0e80d48c To: "0409227633" ;tag=t1187094304-co273 Contact: Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 13152 13153 IN IP4 202.83.178.13 s=session c=IN IP4 202.83.178.13 t=0 0 m=audio 17236 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #4 (no NAT) to 202.83.183.33:5060: INVITE sip:0409227633@202.83.183.33:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK37a1e9f5;rport From: ;tag=as0e80d48c To: "0409227633" ;tag=t1187094304-co273 Contact: Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 13152 13153 IN IP4 202.83.178.13 s=session c=IN IP4 202.83.178.13 t=0 0 m=audio 17236 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #5 (no NAT) to 202.83.183.33:5060: INVITE sip:0409227633@202.83.183.33:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK37a1e9f5;rport From: ;tag=as0e80d48c To: "0409227633" ;tag=t1187094304-co273 Contact: Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 13152 13153 IN IP4 202.83.178.13 s=session c=IN IP4 202.83.178.13 t=0 0 m=audio 17236 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #6 (no NAT) to 202.83.183.33:5060: INVITE sip:0409227633@202.83.183.33:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK37a1e9f5;rport From: ;tag=as0e80d48c To: "0409227633" ;tag=t1187094304-co273 Contact: Call-ID: 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 13152 13153 IN IP4 202.83.178.13 s=session c=IN IP4 202.83.178.13 t=0 0 m=audio 17236 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- == Parsing '/etc/asterisk/manager.conf': Found == Manager 'nagios' logged on from 127.0.0.1 == Manager 'nagios' logged off from 127.0.0.1 [Aug 14 22:25:07] WARNING[13163]: chan_sip.c:1921 retrans_pkt: Maximum retries exceeded on transmission 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 for seqno 102 (Critical Request) [Aug 14 22:25:07] WARNING[13163]: chan_sip.c:1945 retrans_pkt: Hanging up call 00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4 - no reply to our critical packet. set_destination: Parsing for address/port to send to set_destination: set destination to 10.252.11.130, port 5060 Audio is at 202.83.176.38 port 14670 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.252.11.130:5060: INVITE sip:9096@10.252.11.130:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK30549162;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Contact: Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 13152 13154 IN IP4 202.83.176.38 s=session c=IN IP4 202.83.176.38 t=0 0 m=audio 14670 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Scheduling destruction of SIP dialog '118a172c7532562c4e39897f15ded53d@202.83.176.38' in 32000 ms (Method: INVITE) == Spawn extension (macro-call-int-sip, s, 2) exited non-zero on 'SIP/5060-08c1bef0' in macro 'call-int-sip' == Spawn extension (macro-call-int-sip, s, 2) exited non-zero on 'SIP/5060-08c1bef0' <--- SIP read from 10.252.11.130:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK30549162;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Date: Tue, 14 Aug 2007 12:25:08 GMT Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 104 INVITE Allow-Events: telephone-event Remote-Party-ID: "Edwin Groothuis" ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.252.11.130, port 5060 Reliably Transmitting (no NAT) to 10.252.11.130:5060: BYE sip:9096@10.252.11.130:5060 SIP/2.0 Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK2fef03fb;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '118a172c7532562c4e39897f15ded53d@202.83.176.38' in 32000 ms (Method: INVITE) Really destroying SIP dialog '00B2-00E2-9EC233C2-0@6CAF0C3765B8A38F4' Method: ACK <--- SIP read from 10.252.11.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK30549162;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Date: Tue, 14 Aug 2007 12:25:08 GMT Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 CSeq: 104 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: "Edwin Groothuis" ;party=called;screen=no;privacy=off Contact: Content-Type: application/sdp Content-Length: 229 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 10.252.11.130 s=SIP Call c=IN IP4 202.83.178.13 t=0 0 m=audio 17236 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 11 lines) --- <--- SIP read from 10.252.11.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 202.83.176.38:5060;branch=z9hG4bK2fef03fb;rport From: "0409227633" ;tag=as685bacc3 To: ;tag=17417059 Date: Tue, 14 Aug 2007 12:25:08 GMT Call-ID: 118a172c7532562c4e39897f15ded53d@202.83.176.38 Content-Length: 0 CSeq: 105 BYE <-------------> --- == Parsing '/etc/asterisk/manager.conf': Found == Manager 'nagios' logged on from 127.0.0.1 == Manager 'nagios' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'nagios' logged on from 127.0.0.1 == Manager 'nagios' logged off from 127.0.0.1 sip no debugRetransmitting #2 (no NAT) to 10.252.11.3:5060: ---