[SR-Users] Elastix MT (3.0) issue with NAT (ACK is not received for 200 OK)

Hello. I think I understand the problem, but I do not know how to fix it. I have done the following when the problem accord: -Download Elastix MT 64bit (downloaded from - http://www.elastix.com/en/downloads/) -Install it in VMware ESXi 5.1. Elastix works fine within the LAN -Then I pushed the VM to EC2 by exporting VMware OVF -I noticed that I could not registered my softphone (eyebeam 1.5) -After doing research online, I had to modify /etc/kamailio/kamailio-mhomed-elastix.cfg to know about the new IP of EC2 instance. See below of what I had to change: Old settings: [root@new-host-2 endpoint_configurator]# grep 192 /etc/kamailio/kamailio-mhomed-elastix.cfg if (is_in_subnet($var(target_remote_ip), "192.168.1.0/24")) { $var(rtpproxy_if) = "192.168.1.62"; $var(rtpproxy_if) = "192.168.1.6"; To New Settings: [root@new-host-2 ~]# grep 172 /etc/kamailio/kamailio-mhomed-elastix.cfg if (is_in_subnet($var(target_remote_ip), "172.31.48.0/18")) { $var(rtpproxy_if) = "172.31.55.101"; $var(rtpproxy_if) = "172.31.55.101"; -Now I can register my eyebeam softphone. -When ever I dial anything, the connection would *drop after 10 to 30 seconds.* -When I enable SIP debug, I get the following: ############################################################# [May 28 16:07:58] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> INVITE sip:*65@voiptosave.com SIP/2.0 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Max-Forwards: 69 Contact: <sip:110@192.168.1.6:29470> To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 457 v=0 o=- 2 2 IN IP4 192.168.1.62 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.62 t=0 0 m=audio 18598 RTP/AVP 100 106 6 0 105 8 18 3 5 101 a=alt:1 1 : KjAnzMl+ lOdYVIyR 192.168.1.6 15880 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:60C3A67CD0E643D8A6945053F94B1D3E a=nortpproxy:yes <-------------> [May 28 16:07:58] VERBOSE[2688] chan_sip.c: --- (15 headers 17 lines) --- [May 28 16:07:58] VERBOSE[2688] chan_sip.c: Sending to 127.0.0.1:5060 (NAT) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Sending to 127.0.0.1:5060 (NAT) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Using INVITE request as basis request - Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found peer ' 110_voiptosave.com' for '110_voiptosave.com' from 127.0.0.1:5060 [May 28 16:07:58] VERBOSE[2688][C-00000021] netsock2.c: == Using SIP RTP TOS bits 184 [May 28 16:07:58] VERBOSE[2688][C-00000021] netsock2.c: == Using SIP RTP CoS mark 5 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 100 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 106 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 6 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 0 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 105 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 8 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 18 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 3 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 5 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 101 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found audio description format SPEEX for ID 100 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found unknown media description format SPEEX-FEC for ID 106 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found unknown media description format SPEEX-FEC for ID 105 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found audio description format G729 for ID 18 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found audio description format telephone-event for ID 101 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw|alaw|adpcm|g729|speex16)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Peer audio RTP is at port 192.168.1.62:18598 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Looking for *65 in voiptosave.com-from-internal (domain voiptosave.com) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: list_route: hop: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: list_route: hop: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080> Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Server: Asterisk PBX 11.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:*65@127.0.0.1:5080> Content-Length: 0 <------------> [May 28 16:07:58] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:1] Answer("SIP/110_voiptosave.com-0000001a", "") in new stack [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Audio is at 17370 [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding codec 100003 (ulaw) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding codec 100004 (alaw) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding codec 100002 (gsm) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: <--- Reliably Transmitting (NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Server: Asterisk PBX 11.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:*65@127.0.0.1:5080> Content-Type: application/sdp Content-Length: 275 v=0 o=root 153043023 153043023 IN IP4 127.0.0.1 s=Asterisk PBX 11.13.0 c=IN IP4 127.0.0.1 t=0 0 m=audio 17370 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [May 28 16:07:58] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:2] Wait("SIP/110_voiptosave.com-0000001a", "1") in new stack [May 28 16:07:58] VERBOSE[2688] chan_sip.c: Retransmitting #1 (NAT) to 127.0.0.1:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Server: Asterisk PBX 11.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:*65@127.0.0.1:5080> Content-Type: application/sdp Content-Length: 275 v=0 o=root 153043023 153043023 IN IP4 127.0.0.1 s=Asterisk PBX 11.13.0 c=IN IP4 127.0.0.1 t=0 0 m=audio 17370 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [May 28 16:07:58] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> ACK sip:*65@127.0.0.1:5080 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.1131e459ea9f93c78896b0fdaf99b3be.0 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-9f74bc13426dd510-1--d87543-;rport=29470 Max-Forwards: 69 Contact: <sip:110@192.168.1.6:29470> To: "*65"<sip:*65@voiptosave.com>;tag=as3a404cf6 From: "Manny"<sip:110@voiptosave.com>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 ACK Proxy-Authorization: Digest username="110",realm="voiptosave.com ",nonce="VWd2ylVndZ5Yo3kaAdjLh/W6zc+CeoE1",uri="sip:*65@voiptosave.com ",response="ed5ad3dd093d74414340aa68dca6a246",algorithm=MD5 User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:07:58] VERBOSE[2688] chan_sip.c: --- (12 headers 0 lines) --- [May 28 16:07:58] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> ACK sip:*65@127.0.0.1:5080 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.1131e459ea9f93c78896b0fdaf99b3be.0 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-9f74bc13426dd510-1--d87543-;rport=29470 Max-Forwards: 69 Contact: <sip:110@192.168.1.6:29470> To: "*65"<sip:*65@voiptosave.com>;tag=as3a404cf6 From: "Manny"<sip:110@voiptosave.com>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 ACK Proxy-Authorization: Digest username="110",realm="voiptosave.com ",nonce="VWd2ylVndZ5Yo3kaAdjLh/W6zc+CeoE1",uri="sip:*65@voiptosave.com ",response="ed5ad3dd093d74414340aa68dca6a246",algorithm=MD5 User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:07:58] VERBOSE[2688] chan_sip.c: --- (12 headers 0 lines) --- [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:3] Macro("SIP/110_voiptosave.com-0000001a", "voiptosave.com-user-callerid,") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:1] Set("SIP/110_voiptosave.com-0000001a", "EXTUSER=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:2] GotoIf("SIP/110_voiptosave.com-0000001a", "0?report") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:3] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(REALCALLERIDNUM=110)") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:4] Set("SIP/110_voiptosave.com-0000001a", "EXTUSER=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:5] Set("SIP/110_voiptosave.com-0000001a", "EXTUSERCIDNAME=User_110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:6] GotoIf("SIP/110_voiptosave.com-0000001a", "0?report") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:7] Set("SIP/110_voiptosave.com-0000001a", "EXTUSERCID=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:8] Set("SIP/110_voiptosave.com-0000001a", "CALLERID(all)="User_110" <110>") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:9] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(CHANNEL(language)=en)") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:10] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(CHANNEL(language)=en)") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:11] GotoIf("SIP/110_voiptosave.com-0000001a", "0?continue") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:12] Set("SIP/110_voiptosave.com-0000001a", "__TTL=64") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:13] GotoIf("SIP/110_voiptosave.com-0000001a", "1?continue") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-user-callerid,s,20) [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:20] Set("SIP/110_voiptosave.com-0000001a", "CALLERID(number)=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:21] Set("SIP/110_voiptosave.com-0000001a", "CALLERID(name)=User_110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:22] NoOp("SIP/110_voiptosave.com-0000001a", "Using CallerID "User_110" <110>") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:4] Playback("SIP/110_voiptosave.com-0000001a", "your") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'your.gsm' (language 'en') [May 28 16:08:00] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:5] Playback("SIP/110_voiptosave.com-0000001a", "extension") in new stack [May 28 16:08:00] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'extension.gsm' (language 'en') [May 28 16:08:01] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:6] Playback("SIP/110_voiptosave.com-0000001a", "number") in new stack [May 28 16:08:01] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'number.gsm' (language 'en') [May 28 16:08:02] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:7] Playback("SIP/110_voiptosave.com-0000001a", "is") in new stack [May 28 16:08:02] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'is.gsm' (language 'en') [May 28 16:08:03] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:8] SayDigits("SIP/110_voiptosave.com-0000001a", "110") in new stack [May 28 16:08:03] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'digits/1.gsm' (language 'en') [May 28 16:08:03] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'digits/1.gsm' (language 'en') [May 28 16:08:04] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'digits/0.gsm' (language 'en') [May 28 16:08:05] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:9] Wait("SIP/110_voiptosave.com-0000001a", "2") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:10] Hangup("SIP/110_voiptosave.com-0000001a", "") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: == Spawn extension (voiptosave.com-from-internal, *65, 10) exited non-zero on 'SIP/110_voiptosave.com-0000001a' [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [h@voiptosave.com-from-internal:1] Macro("SIP/110_voiptosave.com-0000001a", "voiptosave.com-hangupcall") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:1] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(CDR(organization_domain)= voiptosave.com)") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:2] GotoIf("SIP/110_voiptosave.com-0000001a", "1?endmixmoncheck") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,7) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:7] NoOp("SIP/110_voiptosave.com-0000001a", "End of MIXMON check") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:8] GotoIf("SIP/110_voiptosave.com-0000001a", "1?nomeetmemon") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,14) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:14] NoOp("SIP/110_voiptosave.com-0000001a", "MEETME_RECORDINGFILE=") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:15] GotoIf("SIP/110_voiptosave.com-0000001a", "1?noautomon") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,23) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:23] NoOp("SIP/110_voiptosave.com-0000001a", "TOUCH_MONITOR_OUTPUT=") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:24] GotoIf("SIP/110_voiptosave.com-0000001a", "1?noautomon2") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,31) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:31] NoOp("SIP/110_voiptosave.com-0000001a", "MONITOR_FILENAME=") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:32] GotoIf("SIP/110_voiptosave.com-0000001a", "1?skiprg") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,35) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:35] GotoIf("SIP/110_voiptosave.com-0000001a", "1?skipblkvm") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,38) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:38] GotoIf("SIP/110_voiptosave.com-0000001a", "1?theend") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,40) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:40] Hangup("SIP/110_voiptosave.com-0000001a", "") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] app_macro.c: == Spawn extension (macro-voiptosave.com-hangupcall, s, 40) exited non-zero on 'SIP/110_voiptosave.com-0000001a' in macro 'voiptosave.com-hangupcall' [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: == Spawn extension (voiptosave.com-from-internal, h, 1) exited non-zero on 'SIP/110_voiptosave.com-0000001a' [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: Scheduling destruction of SIP dialog 'Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA.' in 6400 ms (Method: ACK) [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: set_destination: Parsing <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> for address/port to send to [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: set_destination: set destination to 127.0.0.1:5060 [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: Reliably Transmitting (NAT) to 127.0.0.1:5060: BYE sip:110@192.168.1.6:29470 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2355deaa;rport Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes>,<sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Max-Forwards: 70 From: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 To: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 102 BYE User-Agent: Asterisk PBX 11.13.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [May 28 16:08:07] VERBOSE[2688] chan_sip.c: Retransmitting #1 (NAT) to 127.0.0.1:5060: BYE sip:110@192.168.1.6:29470 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2355deaa;rport Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes>,<sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Max-Forwards: 70 From: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 To: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 102 BYE User-Agent: Asterisk PBX 11.13.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [May 28 16:08:07] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2355deaa;rport=5080 Contact: <sip:110@192.168.1.6:29470> To: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e From: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 102 BYE User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:08:07] VERBOSE[2688] chan_sip.c: --- (9 headers 0 lines) --- [May 28 16:08:07] VERBOSE[2688][C-00000021] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [May 28 16:08:07] VERBOSE[2688] chan_sip.c: Really destroying SIP dialog 'Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA.' Method: ACK [May 28 16:08:08] VERBOSE[2688] chan_sip.c: Reliably Transmitting (NAT) to 127.0.0.1:5060: OPTIONS sip:110@192.168.1.6:29470;rinstance=3180b40a89f2e3f4 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK0bcd981c;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as68358641 To: <sip:110@192.168.1.6:29470;rinstance=3180b40a89f2e3f4> Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 03d50e985a2f08445ad315c83a591412@127.0.0.1:5080 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.13.0 Date: Thu, 28 May 2015 20:08:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [May 28 16:08:08] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK0bcd981c;rport=5080 Contact: <sip:192.168.1.6:29470> To: <sip:110@192.168.1.6:29470;rinstance=3180b40a89f2e3f4>;tag=86756729 From: "asterisk"<sip:asterisk@127.0.0.1:5080>;tag=as68358641 Call-ID: 03d50e985a2f08445ad315c83a591412@127.0.0.1:5080 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:08:08] VERBOSE[2688] chan_sip.c: --- (12 headers 0 lines) --- [May 28 16:08:08] VERBOSE[2688] chan_sip.c: Really destroying SIP dialog ' 03d50e985a2f08445ad315c83a591412@127.0.0.1:5080' Method: OPTIONS ################################################################## Please let me know if you need any settings. I am more than happy to provide it. Thank you.