<-- SIP read from 192.168.70.196:65471: INVITE sip:2501@172.25.49.213 SIP/2.0 Supported: 100rel User-Agent: OxO GW To: sip:2501@172.25.49.213 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118651476 1118651476 IN IP4 192.168.70.196 s=- c=IN IP4 192.168.70.196 t=0 0 m=audio 32002 RTP/AVP 18 4 8 0 a=sendrecv --- (12 headers 7 lines)--- Using INVITE request as basis request - 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 Jun 13 08:29:28 WARNING[12216]: chan_sip.c:6013 check_via: Don't know how to respond via 'SIP/2.0/udp' Found no matching peer or user for '192.168.70.196:65471' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 192.168.70.196:32002 Capabilities: us - 0x8020e (gsm|ulaw|alaw|speex|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 2501 in internal list_route: hop: Transmitting (no NAT) to 192.168.70.196:65471: SIP/2.0 100 Trying Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 ---  -- Executing Macro("SIP/192.168.70.196-0817ad78", "stdextension|2501") in new stack  -- Executing Answer("SIP/192.168.70.196-0817ad78", "") in new stack We're at 172.25.49.213 port 18560 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x200 (speex) Reliably Transmitting (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12216 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - ---  -- Executing DBget("SIP/192.168.70.196-0817ad78", "divto=CFIM/2501") in new stack Jun 13 08:29:28 WARNING[12216]: app_db.c:189 get_exec: This application has been deprecated, please use the ${DB(family/key)} function instead.  -- DBget: varname=divto, family=CFIM, key=2501  -- DBget: Value not found in database.  -- Executing Goto("SIP/192.168.70.196-0817ad78", "s|5") in new stack  -- Goto (macro-stdextension,s,5)  -- Executing Dial("SIP/192.168.70.196-0817ad78", "SIP/2501|15") in new stack We're at 172.25.49.213 port 14816 Answering/Requesting with root capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x200 (speex) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting (no NAT) to 192.168.70.199:5060: INVITE sip:2501@192.168.70.199:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK4f371a33 From: "2241" ;tag=as4720e7cf To: Contact: Call-ID: 581e1a104223f5a60b0ea2e2189971ea@172.25.49.213 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 13 Jun 2005 06:29:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 294 v=0 o=root 12216 12216 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14816 RTP/AVP 8 0 3 110 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ---  -- Called 2501 <-- SIP read from 192.168.70.199:5060: SIP/2.0 100 Trying Content-Length: 0 Server: IP SIP Phone/2.0.1 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK4f371a33 To: ;tag=5de1fd From: "2241" ;tag=as4720e7cf Call-ID: 581e1a104223f5a60b0ea2e2189971ea@172.25.49.213 CSeq: 102 INVITE Contact: --- (9 headers 0 lines)--- <-- SIP read from 192.168.70.196:65471: INVITE sip:2501@172.25.49.213 SIP/2.0 Supported: 100rel User-Agent: OxO GW To: sip:2501@172.25.49.213 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118651476 1118651476 IN IP4 192.168.70.196 s=- c=IN IP4 192.168.70.196 t=0 0 m=audio 32002 RTP/AVP 18 4 8 0 a=sendrecv --- (12 headers 7 lines)--- Ignoring this request We're at 172.25.49.213 port 18560 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x200 (speex) Reliably Transmitting (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12217 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- <-- SIP read from 192.168.70.199:5060: SIP/2.0 180 Ringing Content-Length: 0 Server: IP SIP Phone/2.0.1 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK4f371a33 To: ;tag=5de1fd From: "2241" ;tag=as4720e7cf Call-ID: 581e1a104223f5a60b0ea2e2189971ea@172.25.49.213 CSeq: 102 INVITE Contact: --- (9 headers 0 lines)---  -- SIP/2501-1da9 is ringing <-- SIP read from 192.168.70.196:65471: INVITE sip:2501@172.25.49.213 SIP/2.0 Supported: 100rel User-Agent: OxO GW To: sip:2501@172.25.49.213 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118651476 1118651476 IN IP4 192.168.70.196 s=- c=IN IP4 192.168.70.196 t=0 0 m=audio 32002 RTP/AVP 18 4 8 0 a=sendrecv --- (12 headers 7 lines)--- Ignoring this request We're at 172.25.49.213 port 18560 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x200 (speex) Reliably Transmitting (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12218 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12216 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- <-- SIP read from 192.168.70.199:5060: SIP/2.0 200 OK Server: IP SIP Phone/2.0.1 Content-Length: 222 Content-Type: application/sdp Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK4f371a33 To: ;tag=5de1fd From: "2241" ;tag=as4720e7cf Call-ID: 581e1a104223f5a60b0ea2e2189971ea@172.25.49.213 CSeq: 102 INVITE Contact: v=0 o=iprDC3000689 3327643940 3327643940 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 --- (10 headers 10 lines)--- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.70.199:8000 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8020e (gsm|ulaw|alaw|speex|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) 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 192.168.70.199, port 5060 Transmitting (no NAT) to 192.168.70.199:5060: ACK sip:2501@192.168.70.199:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK2996ef00 From: "2241" ;tag=as4720e7cf To: ;tag=5de1fd Contact: Call-ID: 581e1a104223f5a60b0ea2e2189971ea@172.25.49.213 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 ---  -- SIP/2501-1da9 answered SIP/192.168.70.196-0817ad78  -- Attempting native bridge of SIP/192.168.70.196-0817ad78 and SIP/2501-1da9 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.199, port 5060 We're at 172.25.49.213 port 14816 Answering/Requesting with root capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with capability 0x1 (g723) Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 13 lines Reliably Transmitting (no NAT) to 192.168.70.199:5060: INVITE sip:2501@192.168.70.199:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK4e4e8589 From: "2241" ;tag=as4720e7cf To: ;tag=5de1fd Contact: Call-ID: 581e1a104223f5a60b0ea2e2189971ea@172.25.49.213 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 294 v=0 o=root 12216 12217 IN IP4 192.168.70.196 s=session c=IN IP4 192.168.70.196 t=0 0 m=audio 32002 RTP/AVP 8 0 4 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12217 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Destroying call '3bd323ed384e43a652188e73449a22a0@172.25.49.213' Destroying call '3af9e2d0407b57da6e7428036e4b160b@172.25.49.213' <-- SIP read from 192.168.70.199:5060: SIP/2.0 200 OK Server: IP SIP Phone/2.0.1 Content-Length: 222 Content-Type: application/sdp Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK4e4e8589 To: ;tag=5de1fd From: "2241" ;tag=as4720e7cf Call-ID: 581e1a104223f5a60b0ea2e2189971ea@172.25.49.213 CSeq: 103 INVITE Contact: v=0 o=iprDC3000689 3327643941 3327643941 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 --- (10 headers 10 lines)--- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.70.199:8000 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8020e (gsm|ulaw|alaw|speex|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.199, port 5060 Transmitting (no NAT) to 192.168.70.199:5060: ACK sip:2501@192.168.70.199:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK76abb6a9 From: "2241" ;tag=as4720e7cf To: ;tag=5de1fd Contact: Call-ID: 581e1a104223f5a60b0ea2e2189971ea@172.25.49.213 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 --- Retransmitting #1 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12218 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- <-- SIP read from 192.168.70.196:65471: INVITE sip:2501@172.25.49.213 SIP/2.0 Supported: 100rel User-Agent: OxO GW To: sip:2501@172.25.49.213 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118651476 1118651476 IN IP4 192.168.70.196 s=- c=IN IP4 192.168.70.196 t=0 0 m=audio 32002 RTP/AVP 18 4 8 0 a=sendrecv --- (12 headers 7 lines)--- Ignoring this request We're at 172.25.49.213 port 18560 Answering with preferred capability 0x8 (alaw) Reliably Transmitting (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12219 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12216 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12217 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12218 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12219 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #3 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12216 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- <-- SIP read from 172.25.53.254:5060: --- (0 headers 0 lines) Nat keepalive --- Retransmitting #3 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12217 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #3 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12218 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12219 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- <-- SIP read from 192.168.70.196:65471: INVITE sip:2501@172.25.49.213 SIP/2.0 Supported: 100rel User-Agent: OxO GW To: sip:2501@172.25.49.213 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118651476 1118651476 IN IP4 192.168.70.196 s=- c=IN IP4 192.168.70.196 t=0 0 m=audio 32002 RTP/AVP 18 4 8 0 a=sendrecv --- (12 headers 7 lines)--- Ignoring this request We're at 172.25.49.213 port 18560 Answering with preferred capability 0x8 (alaw) Reliably Transmitting (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12220 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #4 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12216 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #4 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12217 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #4 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12218 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #3 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12219 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12220 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #5 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12216 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- <-- SIP read from 192.168.70.199:5060: SUBSCRIBE sip:111@172.25.49.213 SIP/2.0 Content-Length: 0 Date: Tue, 13 Jun 2005 09:32:29 GMT Via: SIP/2.0/UDP 192.168.70.199:5060;branch=z9hG4bK3b014c To: "Christian Cayeux" ;tag=3f093 From: "Christian Cayeux" ;tag=3f093 Call-ID: 6e25d6b7-872fb2d6-6c46911d-da167108@192.168.70.199 Event: message-summary CSeq: 659 SUBSCRIBE Expires: 3600 User-Agent: IP SIP Phone/2.0.1 Max-Forwards: 70 Accept: application/simple-message-summary Contact: --- (14 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 192.168.70.199 : 5060 (non-NAT) Found peer '2501' Looking for 111 in internal Transmitting (no NAT) to 192.168.70.199:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.199:5060;branch=z9hG4bK3b014c From: "Christian Cayeux" ;tag=3f093 To: "Christian Cayeux" ;tag=3f093 Call-ID: 6e25d6b7-872fb2d6-6c46911d-da167108@192.168.70.199 CSeq: 659 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Destroying call '6e25d6b7-872fb2d6-6c46911d-da167108@192.168.70.199' Retransmitting #5 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12217 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #5 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 12216 12218 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 18560 RTP/AVP 8 0 3 110 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=silenceSupp:off - - - - --- Retransmitting #4 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12219 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12220 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Jun 13 08:29:40 WARNING[12216]: chan_sip.c:908 retrans_pkt: Maximum retries exceeded on call 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 for seqno 1203868680 (Non-critical Response) Jun 13 08:29:40 WARNING[12216]: chan_sip.c:908 retrans_pkt: Maximum retries exceeded on call 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 for seqno 1203868680 (Non-critical Response) Jun 13 08:29:41 WARNING[12216]: chan_sip.c:908 retrans_pkt: Maximum retries exceeded on call 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 for seqno 1203868680 (Non-critical Response) Retransmitting #5 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12219 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Retransmitting #3 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12220 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- *CLI> stop nJun 13 08:29:43 WARNING[12216]: chan_sip.c:908 retrans_pkt: Maximum retries exceeded on call 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 for seqno 1203868680 (Non-critical Response) Retransmitting #4 (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12220 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- <-- SIP read from 192.168.70.196:65471: INVITE sip:2501@172.25.49.213 SIP/2.0 Supported: 100rel User-Agent: OxO GW To: sip:2501@172.25.49.213 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118651476 1118651476 IN IP4 192.168.70.196 s=- c=IN IP4 192.168.70.196 t=0 0 m=audio 32002 RTP/AVP 18 4 8 0 a=sendrecv --- (12 headers 7 lines)--- Ignoring this request We're at 172.25.49.213 port 18560 Answering with preferred capability 0x8 (alaw) Reliably Transmitting (no NAT) to 192.168.70.196:65471: SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKc98d26d6c14ffc092a0ce6a3ba102991 From: sip:2241@192.168.70.196;tag=9496da203eff274f0a2619d45ca9c9f4 To: sip:2501@172.25.49.213;tag=as592cc1d4 Call-ID: 9cf77884a5e80c6fe7b8d5b494355355@192.168.70.196 CSeq: 1203868680 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 12216 12221 IN IP4 192.168.70.199 s=session c=IN IP4 192.168.70.199 t=0 0 m=audio 8000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- ow  Beginning asterisk shutdown.... set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.199, port 5060 Reliably Transmitting (no NAT) to 192.168.70.199:5060: BYE sip:2501@192.168.70.199:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK4858d0e8 From: "2241" ;tag=as4720e7cf To: ;tag=5de1fd Contact: Call-ID: 581e1a104223f5a60b0ea2e2189971ea@172.25.49.213 CSeq: 104 BYE User-Agent: Asterisk PBX Content-Length: 0 ---  == Spawn extension (macro-stdextension, s, 5) exited non-zero on 'SIP/192.168.70.196-0817ad78' in macro 'stdextension'  == Spawn extension (internal, 2501, 1) exited non-zero on 'SIP/192.168.70.196-0817ad78' Executing last minute cleanups  == Destroying musiconhold processes Asterisk cleanly ending (0).