[SR-Users] Elastix MT (3.0) issue with NAT (ACK is not received for 200 OK)
Mainul Haque
mhaque at routvox.com
Thu May 28 23:09:05 CEST 2015
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 at 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";
$var(rtpproxy_if) = "192.168.1.6";
To New Settings:
[root at new-host-2 ~]# grep 172 /etc/kamailio/kamailio-mhomed-elastix.cfg
if (is_in_subnet($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 at 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 at 192.168.1.6:29470>
To: "*65"<sip:*65_voiptosave.com at 127.0.0.1:5080>
From: "Manny"<sip:110_voiptosave.com at 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 at 127.0.0.1:5080>;tag=f133137e
To: "*65"<sip:*65_voiptosave.com at 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 at 127.0.0.1:5080>
Content-Length: 0
<------------>
[May 28 16:07:58] VERBOSE[9110][C-00000021] pbx.c: -- Executing
[*65 at 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 at 127.0.0.1:5080>;tag=f133137e
To: "*65"<sip:*65_voiptosave.com at 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 at 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 at 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 at 127.0.0.1:5080>;tag=f133137e
To: "*65"<sip:*65_voiptosave.com at 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 at 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 at 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 at 192.168.1.6:29470>
To: "*65"<sip:*65 at voiptosave.com>;tag=as3a404cf6
From: "Manny"<sip:110 at 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 at 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 at 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 at 192.168.1.6:29470>
To: "*65"<sip:*65 at voiptosave.com>;tag=as3a404cf6
From: "Manny"<sip:110 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 127.0.0.1:5080>;tag=as3a404cf6
To: "Manny"<sip:110_voiptosave.com at 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 at 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 at 127.0.0.1:5080>;tag=as3a404cf6
To: "Manny"<sip:110_voiptosave.com at 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 at 192.168.1.6:29470>
To: "Manny"<sip:110_voiptosave.com at 127.0.0.1:5080>;tag=f133137e
From: "*65"<sip:*65_voiptosave.com at 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 at 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 at 127.0.0.1:5080>;tag=as68358641
To: <sip:110 at 192.168.1.6:29470;rinstance=3180b40a89f2e3f4>
Contact: <sip:asterisk at 127.0.0.1:5080>
Call-ID: 03d50e985a2f08445ad315c83a591412 at 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 at 192.168.1.6:29470;rinstance=3180b40a89f2e3f4>;tag=86756729
From: "asterisk"<sip:asterisk at 127.0.0.1:5080>;tag=as68358641
Call-ID: 03d50e985a2f08445ad315c83a591412 at 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 at 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.
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