Sip read: 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=40e715f986d866c2ecd1bd4f57245eb0 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118302586 1118302586 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 latest request as basis request Jun 9 07:34:36 WARNING[31579]: chan_sip.c:5248 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 - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 2501 in internal list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.70.196:65471  -- Executing Macro("SIP/192.168.70.196-0814d738", "stdextension|2501") in new stack  -- Executing Answer("SIP/192.168.70.196-0814d738", "") in new stack We're at 172.25.49.213 port 14856 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31579 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471  -- Executing DBget("SIP/192.168.70.196-0814d738", "divto=CFIM/2501") in new stack  -- DBget: varname=divto, family=CFIM, key=2501  -- DBget: Value not found in database.  -- Executing Goto("SIP/192.168.70.196-0814d738", "s|5") in new stack  -- Goto (macro-stdextension,s,5)  -- Executing Dial("SIP/192.168.70.196-0814d738", "SIP/2501|15") in new stack We're at 172.25.49.213 port 15956 Answering/Requesting with root capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting: INVITE sip:2501@192.168.70.199:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK4e51fa74 From: "2241" ;tag=as2b2e57cc To: Contact: Call-ID: 2576fcca76a920c132f718e77c2b6cf9@172.25.49.213 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 09 Jun 2005 05:34:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 265 v=0 o=root 31579 31579 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 15956 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 192.168.70.199:5060  -- Called 2501 Sip read: 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=z9hG4bK4e51fa74 To: ;tag=659965cf From: "2241" ;tag=as2b2e57cc Call-ID: 2576fcca76a920c132f718e77c2b6cf9@172.25.49.213 CSeq: 102 INVITE Contact: 9 headers, 0 lines Sip read: 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=40e715f986d866c2ecd1bd4f57245eb0 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118302586 1118302586 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 14856 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31580 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Jun 9 07:34:37 WARNING[31579]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 15444cc12797d7e812538d534ab18be1@172.25.49.213 for seqno 102 (Non-critical Request) Jun 9 07:34:37 WARNING[31579]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 58fee9a02bbc96670fd26a1a76c57831@172.25.49.213 for seqno 102 (Non-critical Request) Retransmitting #1 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31579 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Sip read: 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=z9hG4bK4e51fa74 To: ;tag=659965cf From: "2241" ;tag=as2b2e57cc Call-ID: 2576fcca76a920c132f718e77c2b6cf9@172.25.49.213 CSeq: 102 INVITE Contact: 9 headers, 0 lines  -- SIP/2501-d79c is ringing Sip read: 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=40e715f986d866c2ecd1bd4f57245eb0 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118302586 1118302586 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 14856 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31581 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #1 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31580 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Sip read: SUBSCRIBE sip:111@172.25.49.213 SIP/2.0 Content-Length: 0 Date: Fri, 08 Jan 1970 17:56:53 GMT Via: SIP/2.0/UDP 192.168.70.199:5060;branch=z9hG4bK6702630d To: "Christian Cayeux" ;tag=4dc04f04 From: "Christian Cayeux" ;tag=4dc04f04 Call-ID: 9a36b217-2baf61ff-1471cd17-19fb4f85@192.168.70.199 Event: message-summary CSeq: 24505 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): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.70.199:5060;branch=z9hG4bK6702630d From: "Christian Cayeux" ;tag=4dc04f04 To: "Christian Cayeux" ;tag=4dc04f04 Call-ID: 9a36b217-2baf61ff-1471cd17-19fb4f85@192.168.70.199 CSeq: 24505 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.70.199:5060 Destroying call '9a36b217-2baf61ff-1471cd17-19fb4f85@192.168.70.199' Retransmitting #2 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31579 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #2 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31580 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #1 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31581 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Sip read: 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=z9hG4bK4e51fa74 To: ;tag=659965cf From: "2241" ;tag=as2b2e57cc Call-ID: 2576fcca76a920c132f718e77c2b6cf9@172.25.49.213 CSeq: 102 INVITE Contact: v=0 o=iprDC3000689 2209658212 2209658212 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 - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.199, port 5060 Transmitting: ACK sip:2501@192.168.70.199:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK41096a96 From: "2241" ;tag=as2b2e57cc To: ;tag=659965cf Contact: Call-ID: 2576fcca76a920c132f718e77c2b6cf9@172.25.49.213 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.70.199:5060  -- SIP/2501-d79c answered SIP/192.168.70.196-0814d738  -- Attempting native bridge of SIP/192.168.70.196-0814d738 and SIP/2501-d79c 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 15956 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: INVITE sip:2501@192.168.70.199:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK2c4c8e9e From: "2241" ;tag=as2b2e57cc To: ;tag=659965cf Contact: Call-ID: 2576fcca76a920c132f718e77c2b6cf9@172.25.49.213 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 294 v=0 o=root 31579 31580 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 - - - - (no NAT) to 192.168.70.199:5060 Sip read: 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=z9hG4bK2c4c8e9e To: ;tag=659965cf From: "2241" ;tag=as2b2e57cc Call-ID: 2576fcca76a920c132f718e77c2b6cf9@172.25.49.213 CSeq: 103 INVITE Contact: v=0 o=iprDC3000689 2209658214 2209658214 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 - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.70.199, port 5060 Transmitting: ACK sip:2501@192.168.70.199:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK1a426ec9 From: "2241" ;tag=as2b2e57cc To: ;tag=659965cf Contact: Call-ID: 2576fcca76a920c132f718e77c2b6cf9@172.25.49.213 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.70.199:5060 Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31579 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31580 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #2 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31581 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Sip read: 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=40e715f986d866c2ecd1bd4f57245eb0 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118302586 1118302586 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 14856 Answering with preferred capability 0x8 (alaw) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31582 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 - - - - to 192.168.70.196:65471 Retransmitting #4 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31579 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #4 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31580 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31581 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #1 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31582 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 - - - - to 192.168.70.196:65471 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31579 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31580 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #4 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31581 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #2 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31582 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 - - - - to 192.168.70.196:65471 Jun 9 07:34:42 WARNING[31579]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 for seqno 2046050713 (Non-critical Response) Jun 9 07:34:43 WARNING[31579]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 for seqno 2046050713 (Non-critical Response) Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=root 31579 31581 IN IP4 172.25.49.213 s=session c=IN IP4 172.25.49.213 t=0 0 m=audio 14856 RTP/AVP 8 0 3 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 192.168.70.196:65471 Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31582 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 - - - - to 192.168.70.196:65471 Jun 9 07:34:44 WARNING[31579]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 for seqno 2046050713 (Non-critical Response) Retransmitting #4 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31582 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 - - - - to 192.168.70.196:65471 Sip read: 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=40e715f986d866c2ecd1bd4f57245eb0 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118302586 1118302586 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 14856 Answering with preferred capability 0x8 (alaw) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31583 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 - - - - to 192.168.70.196:65471 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31582 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 - - - - to 192.168.70.196:65471 Retransmitting #1 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31583 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 - - - - to 192.168.70.196:65471 Jun 9 07:34:46 WARNING[31579]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 for seqno 2046050713 (Non-critical Response) Retransmitting #2 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31583 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 - - - - to 192.168.70.196:65471 Destroying call '15444cc12797d7e812538d534ab18be1@172.25.49.213' Destroying call '58fee9a02bbc96670fd26a1a76c57831@172.25.49.213' Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31583 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 - - - - to 192.168.70.196:65471 Retransmitting #4 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31583 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 - - - - to 192.168.70.196:65471 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31583 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 - - - - to 192.168.70.196:65471 Jun 9 07:34:50 WARNING[31579]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 for seqno 2046050713 (Non-critical Response) *CLI> Sip read: 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=40e715f986d866c2ecd1bd4f57245eb0 Contact: sip:2241@192.168.70.196 Content-Type: application/sdp Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 Max-Forwards: 70 Content-Length: 141 v=0 o=default 1118302586 1118302586 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 14856 Answering with preferred capability 0x8 (alaw) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31584 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 - - - - to 192.168.70.196:65471 Retransmitting #1 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31584 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 - - - - to 192.168.70.196:65471 Retransmitting #2 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31584 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 - - - - to 192.168.70.196:65471 Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31584 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 - - - - to 192.168.70.196:65471 stop n Retransmitting #4 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/udp 192.168.70.196;branch=z9hG4bKe636bfc839e5808d54df557d28091ab0 From: sip:2241@192.168.70.196;tag=40e715f986d866c2ecd1bd4f57245eb0 To: sip:2501@172.25.49.213;tag=as105bba5c Call-ID: 6dc8c42c05ed69378ccf76e958e12daf@192.168.70.196 CSeq: 2046050713 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 163 v=0 o=root 31579 31584 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 - - - - to 192.168.70.196:65471 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: BYE sip:2501@192.168.70.199:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.49.213:5060;branch=z9hG4bK1950d426 From: "2241" ;tag=as2b2e57cc To: ;tag=659965cf Contact: Call-ID: 2576fcca76a920c132f718e77c2b6cf9@172.25.49.213 CSeq: 104 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.70.199:5060  == Spawn extension (macro-stdextension, s, 5) exited non-zero on 'SIP/192.168.70.196-0814d738' in macro 'stdextension'  == Spawn extension (internal, 2501, 1) exited non-zero on 'SIP/192.168.70.196-0814d738' Executing last minute cleanups  == Destroying any remaining musiconhold processes Asterisk cleanly ending (0).