[Freeswitch-users] Call is interrupted in 30 seconds, configuring NAT on Amazon EC2
Стас Тельнов
stasan89 at gmail.com
Mon Apr 11 13:59:05 MSD 2016
>>Artur Mega: Do you use sip-proxy? If you use sip proxy, maybe you forget
to add "Record-Route" header to sip-query
"Record-Route" header already exists.
>>Regis M: 99% of the time 30 seconds hangup (or 32seconds) means a NAT
problem... or a response not come back to source...
I understand, but I don`t know how fix it.
Jurijs Ivolga,
I analize packets in ngrep.
First BYE packet include reason: NORMAL_CLEARING. BYE packet for caller
include reason: ACK Timeout.
About NORMAL_CLEARING in freeswitch documentation (
https://wiki.freeswitch.org/wiki/Hangup_Causes):
This cause indicates that the call is being cleared because one of the
users involved in the call has requested that the call be cleared. Under
normal situations, the source of this cause is not the network.
But phone clients do not send this command.
I suppose that this is because the server has not received confirmation from
one of the customers that the call took place.
All packages on call:
#
U 172.31.0.169:5060 -> 172.31.22.124:5060
INVITE sip:7906*******@sip0.MY_SIP_DOMAIN.com SIP/2.0.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Max-Forwards: 69.
Contact: <sip:8 at 85.
*.*.4:53712;ob;transport=tls>;+sip.instance="<urn:uuid:E458598F-FBED-B020-670D-167CE2ADB38A>".
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>.
From: "8"<sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER,
SUBSCRIBE, INFO.
Content-Type: application/sdp.
Supported: replaces, outbound, path.
User-Agent: PortSIP SDK for IOS.
Content-Length: 219.
.
v=0.
o=- 1460361915 1 IN IP4 85.*.*.4.
s=portsip.com.
c=IN IP4 52.*.*.177.
t=0 0.
m=audio 40772 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=sendrecv.
a=nortpproxy:yes.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Content-Length: 0.
.
#
U 172.31.22.124:5060 -> 178.*.*.12:5060
INVITE sip:7906*******@freelycall.com SIP/2.0.
Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK5vNr86BpDFNSN.
Max-Forwards: 67.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844894 INVITE.
Contact: <sip:gw+freelycall.com at 52.*.*.198:5060;transport=udp;gw=
freelycall.com>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 244.
X-FS-Support: update_display,send_info.
Remote-Party-ID: "8" <sip:8 at freelycall.com
>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1460338046 1460338047 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 23870 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.
#
U 178.*.*.12:5060 -> 172.31.22.124:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844894 INVITE.
Server: Asterisk PBX 11.11.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces.
Contact: <sip:7906*******@178.*.*.12:5060>.
Content-Length: 0.
.
#
U 178.*.*.12:5060 -> 172.31.22.124:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP
52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844894 INVITE.
Server: Asterisk PBX 11.11.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces.
Contact: <sip:7906*******@178.*.*.12:5060>.
Content-Type: application/sdp.
Content-Length: 240.
.
v=0.
o=root 1395806664 1395806664 IN IP4 178.*.*.12.
s=Asterisk PBX 11.11.0.
c=IN IP4 178.*.*.12.
t=0 0.
m=audio 11574 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 178.*.*.12:5060 -> 172.31.22.124:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844894 INVITE.
Server: Asterisk PBX 11.11.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces.
Contact: <sip:7906*******@178.*.*.12:5060>.
Content-Type: application/sdp.
Content-Length: 240.
.
v=0.
o=root 1395806664 1395806664 IN IP4 178.*.*.12.
s=Asterisk PBX 11.11.0.
c=IN IP4 178.*.*.12.
t=0 0.
m=audio 11574 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
#
U 172.31.22.124:5060 -> 178.*.*.12:5060
ACK sip:7906*******@178.*.*.12:5060 SIP/2.0.
Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK65eHa2vSarBcH.
Max-Forwards: 70.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844894 ACK.
Contact: <sip:gw+freelycall.com at 52.*.*.198:5060;transport=udp;gw=
freelycall.com>.
Content-Length: 0.
.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
#
U 172.31.22.124:5060 -> 178.*.*.12:5060
BYE sip:7906*******@178.*.*.12:5060 SIP/2.0.
Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK8Q12Dry049QHr.
Max-Forwards: 70.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844895 BYE.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.
#
U 178.*.*.12:5060 -> 172.31.22.124:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
52.*.*.198;branch=z9hG4bK8Q12Dry049QHr;received=52.*.*.198;rport=5060.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844895 BYE.
Server: Asterisk PBX 11.11.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces.
Content-Length: 0.
.
#
U 172.31.22.124:5060 -> 52.*.*.177:5060
BYE sip:8 at 85.*.*.4:53712;ob;transport=tls SIP/2.0.
Via: SIP/2.0/UDP 172.31.22.124;rport;branch=z9hG4bK7e89BXDX701yc.
Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
Max-Forwards: 70.
From: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
To: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 89844917 BYE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Reason: SIP;cause=408;text="ACK Timeout".
Content-Length: 0.
.
#
U 52.*.*.177:5060 -> 172.31.22.124:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
172.31.22.124;received=52.*.*.198;rport=5060;branch=z9hG4bK7e89BXDX701yc.
Contact: <sip:8 at 192.168.0.110:10977
;ob;transport=tls>;+sip.instance="<urn:uuid:E458598F-FBED-B020-670D-167CE2ADB38A>".
To: "8"<sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
From: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 89844917 BYE.
User-Agent: PortSIP SDK for IOS.
Content-Length: 0.
.
2016-04-08 20:01 GMT+03:00 Artur Mega <findmeinwland at gmail.com>:
> Do you use sip-proxy? If you use sip proxy, maybe you forget to add
> "Record-Route" header to sip-query
>
> 2016-04-08 21:13 GMT+05:00 Regis M <regis.freeswitch.org at tornad.net>:
>
>> 99% of the time 30 seconds hangup (or 32seconds) means a NAT problem...
>> or a response not come back to source...
>>
>> 2016-04-08 17:06 GMT+02:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:
>>
>>> Hi,
>>>
>>> This is not what I need, please use ngrep:
>>>
>>>
>>> http://jonathanmanning.com/2009/11/17/how-to-sip-capture-using-ngrep-debug-sip-packets/
>>>
>>> With kind regards,
>>>
>>> Jurijs
>>>
>>> On Fri, Apr 8, 2016 at 6:02 PM, Стас Тельнов <stasan89 at gmail.com> wrote:
>>>
>>>> Yes, of cause. I hide some ip and real phone numbers.
>>>> 178.*.*.12 - ip of provider.
>>>>
>>>> *On start call:*
>>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
>>>> sofia/external/8 at sip0.MY_DOMAIN.com
>>>> [c618eafe-fd98-11e5-a353-831849fc41a3]
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_NEW
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:9248 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com receiving invite from 172.31.0.169:5060 version:
>>>> 1.6.6 64bit
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com entering state [received][100]
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6770 Remote SDP:
>>>> v=0
>>>> o=- 1460126829 1 IN IP4 85.*.*.4
>>>> s=portsip.com
>>>> c=IN IP4 52.*.*.177
>>>> t=0 0
>>>> m=audio 40082 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=nortpproxy:yes
>>>>
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:7125 (sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com) State Change CS_NEW -> CS_INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:492
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State NEW
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_INIT ->
>>>> CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_channel.c:2247
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change DOWN -> RINGING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
>>>> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
>>>> <8>->7906******* in context public
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>> [public->from_opensips] continue=false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
>>>> [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/
>>>> break=on-false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>>> transfer(${destination_number} XML default)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
>>>> CS_EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com transfer(7906******* XML
>>>> default)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr.c:2085 (sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com) State Change CS_EXECUTE -> CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_ivr.c:2092 Transfer
>>>> sofia/external/8 at sip0.MY_DOMAIN.com to XML[7906*******@default]
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
>>>> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
>>>> <8>->7906******* in context default
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>> [default->unloop] continue=false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS) [unloop]
>>>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (FAIL) [unloop]
>>>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>> [default->tod_example] continue=true
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Date/Time Match (PASS)
>>>> [tod_example] break=on-false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action set(open=true)
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>> [default->outbound_calls_to_freelycall] continue=false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
>>>> [outbound_calls_to_freelycall] destination_number(7906*******) =~ /^(.+)/
>>>> break=on-true
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>>> set(hangup_after_bridge=true)
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>>> bridge(sofia/gateway/freelycall.com/7906*******)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
>>>> CS_EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com set(open=true)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET
>>>> sofia/external/8 at sip0.MY_DOMAIN.com [open]=[true]
>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com
>>>> set(hangup_after_bridge=true)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET
>>>> sofia/external/8 at sip0.MY_DOMAIN.com [hangup_after_bridge]=[true]
>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com bridge(sofia/gateway/
>>>> freelycall.com/7906*******)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:2128 Parsing
>>>> global variables
>>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
>>>> sofia/external/7906******* [c619366c-fd98-11e5-a35c-831849fc41a3]
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:4776
>>>> (sofia/external/7906*******) State Change CS_NEW -> CS_INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>> (sofia/external/7906*******) State INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88
>>>> sofia/external/7906******* SOFIA INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia_glue.c:1257
>>>> sofia/external/7906******* sending invite version: 1.6.6 64bit
>>>> Local SDP:
>>>> v=0
>>>> o=FreeSWITCH 1460100428 1460100429 IN IP4 52.*.*.198
>>>> s=FreeSWITCH
>>>> c=IN IP4 52.*.*.198
>>>> t=0 0
>>>> m=audio 26402 RTP/AVP 8 101 13
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=rtpmap:13 CN/8000
>>>> a=ptime:20
>>>> a=sendrecv
>>>>
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
>>>> sofia/external/7906******* Standard INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
>>>> (sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>> (sofia/external/7906*******) State INIT going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel
>>>> sofia/external/7906******* entering state [calling][0]
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/7906*******) State ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141
>>>> sofia/external/7906******* SOFIA ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:67
>>>> (sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/7906*******) State ROUTING going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
>>>> (sofia/external/7906*******) State CONSUME_MEDIA
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
>>>> (sofia/external/7906*******) State CONSUME_MEDIA going to sleep
>>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel
>>>> sofia/external/7906******* entering state [proceeding][183]
>>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6770 Remote SDP:
>>>> v=0
>>>> o=root 153112258 153112258 IN IP4 178.*.*.12
>>>> s=Asterisk PBX 11.11.0
>>>> c=IN IP4 178.*.*.12
>>>> t=0 0
>>>> m=audio 17362 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>>
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
>>>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
>>>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
>>>> telephone-event payload to 101 at 8000
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
>>>> sofia/external/7906******* PCMA/8000 20 ms 160 samples 64000 bits 1 channels
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
>>>> sofia/external/7906******* Original read codec set to PCMA:8
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
>>>> telephone-event payload to 101 at 8000
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
>>>> sofia/external/7906******* Set 2833 dtmf send payload to 101 recv payload
>>>> to 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
>>>> [sofia/external/7906*******] 172.31.22.124 port 26402 -> 178.*.*.12 port
>>>> 17362 codec: 8 ms: 20
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer
>>>> [soft] 160 bytes per 20ms
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
>>>> sofia/external/7906******* Set 2833 dtmf send payload to 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
>>>> sofia/external/7906******* Set 2833 dtmf receive payload to 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
>>>> sofia/external/7906******* Set rtp dtmf delay to 40
>>>> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
>>>> sofia/external/7906*******!
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468
>>>> (sofia/external/7906*******) Callstate Change DOWN -> EARLY
>>>> 2016-04-08 10:47:17.443282 [INFO] switch_ivr_originate.c:3557 Sending
>>>> early media
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
>>>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
>>>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
>>>> telephone-event payload to 101 at 8000
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
>>>> sofia/external/8 at sip0.MY_DOMAIN.com PCMA/8000 20 ms 160 samples 64000
>>>> bits 1 channels
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Original read codec set to PCMA:8
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
>>>> telephone-event payload to 101 at 8000
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
>>>> recv payload to 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
>>>> [sofia/external/8 at sip0.MY_DOMAIN.com] 172.31.22.124 port 30630 ->
>>>> 52.*.*.177 port 40082 codec: 8 ms: 20
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer
>>>> [soft] 160 bytes per 20ms
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf receive payload to
>>>> 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set rtp dtmf delay to 40
>>>> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
>>>> sofia/external/8 at sip0.MY_DOMAIN.com!
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change RINGING -> EARLY
>>>> 2016-04-08 10:47:17.443282 [DEBUG] mod_sofia.c:2330 Ring SDP:
>>>> v=0
>>>> o=FreeSWITCH 1460096207 1460096208 IN IP4 52.*.*.198
>>>> s=FreeSWITCH
>>>> c=IN IP4 52.*.*.198
>>>> t=0 0
>>>> m=audio 30630 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>> a=sendrecv
>>>>
>>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com entering state [early][183]
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_ivr_originate.c:3608
>>>> Originate Resulted in Success: [sofia/external/7906*******]
>>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_ivr_bridge.c:1591
>>>> (sofia/external/7906*******) State Change CS_CONSUME_MEDIA ->
>>>> CS_EXCHANGE_MEDIA
>>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_EXCHANGE_MEDIA
>>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:542
>>>> (sofia/external/7906*******) State EXCHANGE_MEDIA
>>>> 2016-04-08 10:47:17.463254 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
>>>> 2016-04-08 10:47:17.503264 [DEBUG] switch_rtp.c:6654 Correct audio
>>>> ip/port confirmed.
>>>> 2016-04-08 10:47:17.663261 [DEBUG] switch_rtp.c:6654 Correct audio
>>>> ip/port confirmed.
>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
>>>> sofia/external/7906******* entering state [completing][200]
>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6767 Duplicate SDP
>>>> v=0
>>>> o=root 153112258 153112258 IN IP4 178.*.*.12
>>>> s=Asterisk PBX 11.11.0
>>>> c=IN IP4 178.*.*.12
>>>> t=0 0
>>>> m=audio 17362 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>>
>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
>>>> sofia/external/7906******* entering state [ready][200]
>>>> 2016-04-08 10:47:21.323279 [NOTICE] sofia.c:7665 Channel
>>>> [sofia/external/7906*******] has been answered
>>>> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767
>>>> (sofia/external/7906*******) Callstate Change EARLY -> ACTIVE
>>>> 2016-04-08 10:47:21.323279 [DEBUG] mod_sofia.c:799 Local SDP
>>>> sofia/external/8 at sip0.MY_DOMAIN.com:
>>>> v=0
>>>> o=FreeSWITCH 1460096207 1460096209 IN IP4 52.*.*.198
>>>> s=FreeSWITCH
>>>> c=IN IP4 52.*.*.198
>>>> t=0 0
>>>> m=audio 30630 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>> a=sendrecv
>>>>
>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com entering state [completed][200]
>>>> 2016-04-08 10:47:21.323279 [NOTICE] switch_ivr_bridge.c:616 Channel
>>>> [sofia/external/8 at sip0.MY_DOMAIN.com] has been answered
>>>> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change EARLY -> ACTIVE
>>>> 2016-04-08 10:47:21.383279 [DEBUG] switch_rtp.c:6654 Correct audio
>>>> ip/port confirmed.
>>>> 2016-04-08 10:47:21.423259 [DEBUG] switch_rtp.c:6654 Correct audio
>>>> ip/port confirmed.
>>>>
>>>>
>>>> *And after 30 seconds:*
>>>> 2016-04-08 10:47:53.343283 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com entering state [terminating][0]
>>>> 2016-04-08 10:47:53.343283 [NOTICE] sofia.c:7779 Hangup sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:699
>>>> sofia/external/8 at sip0.MY_DOMAIN.com ending bridge by request from
>>>> write function
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE
>>>> THREAD DONE [sofia/external/7906*******]
>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_ivr_bridge.c:881 Hangup
>>>> sofia/external/7906******* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:542
>>>> (sofia/external/7906*******) State EXCHANGE_MEDIA going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
>>>> (sofia/external/7906*******) Callstate Change ACTIVE -> HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>> (sofia/external/7906*******) State HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
>>>> sofia/external/7906******* hanging up, cause: NORMAL_CLEARING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:484 Sending BYE to
>>>> sofia/external/7906*******
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/7906******* Standard HANGUP, cause: NORMAL_CLEARING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>> (sofia/external/7906*******) State HANGUP going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
>>>> (sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>> (sofia/external/7906*******) State REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
>>>> sofia/external/7906******* Standard REPORTING, cause: NORMAL_CLEARING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>> (sofia/external/7906*******) State REPORTING going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
>>>> (sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 2
>>>> (sofia/external/7906*******) Locked, Waiting on external entities
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:705
>>>> sofia/external/8 at sip0.MY_DOMAIN.com ending bridge by request from read
>>>> function
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE
>>>> THREAD DONE [sofia/external/8 at sip0.MY_DOMAIN.com]
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:1692
>>>> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message [UNBRIDGE]
>>>> (channel is hungup already)
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:2796
>>>> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message
>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:539
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change ACTIVE -> HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
>>>> sofia/external/8 at sip0.MY_DOMAIN.com hanging up, cause:
>>>> NORMAL_UNSPECIFIED
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard HANGUP, cause:
>>>> NORMAL_UNSPECIFIED
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_HANGUP ->
>>>> CS_REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard REPORTING, cause:
>>>> NORMAL_UNSPECIFIED
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_REPORTING ->
>>>> CS_DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 1
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Locked, Waiting on external
>>>> entities
>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session
>>>> 1 (sofia/external/8 at sip0.MY_DOMAIN.com) Ended
>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
>>>> Channel sofia/external/8 at sip0.MY_DOMAIN.com [CS_DESTROY]
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY going to sleep
>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session
>>>> 2 (sofia/external/7906*******) Ended
>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
>>>> Channel sofia/external/7906******* [CS_DESTROY]
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
>>>> (sofia/external/7906*******) Running State Change CS_DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>> (sofia/external/7906*******) State DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341
>>>> sofia/external/7906******* SOFIA DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
>>>> sofia/external/7906******* Standard DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>> (sofia/external/7906*******) State DESTROY going to sleep
>>>>
>>>>
>>>> 2016-04-08 17:37 GMT+03:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:
>>>>
>>>>> Hi,
>>>>>
>>>>> I would recommend you to capture SIP packets during call on
>>>>> Freeswitch server and send it here, I will take a look on it.
>>>>>
>>>>> With kind regards,
>>>>>
>>>>> Jurijs
>>>>>
>>>>> On Fri, Apr 8, 2016 at 5:34 PM, Стас Тельнов <stasan89 at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> I already tried disabling timers, does not work.
>>>>>>
>>>>>> 2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar at gmail.com>:
>>>>>>
>>>>>>> Try disabling session timers in the sip profile. I think that line
>>>>>>> is commented out by default, so uncomment it.
>>>>>>>
>>>>>>> <param name="enable-timer" value="false"/>
>>>>>>>
>>>>>>> On Fri, Apr 8, 2016 at 6:59 AM, Стас Тельнов <stasan89 at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Hello.
>>>>>>>>
>>>>>>>> When using a call or conference through sip — freeswitch with
>>>>>>>> external provider there is a problem – the call is interrupted in 30
>>>>>>>> seconds. Though the sound goes all right.
>>>>>>>> I think that it caused by the NAT settings for freeswitch, but I
>>>>>>>> don't understand how to adjust it correctly.
>>>>>>>> At start of freeswitch I see the following mistakes in the tracking
>>>>>>>> data:
>>>>>>>> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
>>>>>>>> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>> PMP 1/5
>>>>>>>> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>> PMP 2/5
>>>>>>>> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>> PMP 3/5
>>>>>>>> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>> PMP 4/5
>>>>>>>> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>> PMP 5/5
>>>>>>>> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking
>>>>>>>> for PMP [general error]
>>>>>>>> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for
>>>>>>>> UPnP
>>>>>>>> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP
>>>>>>>> NAT devices detected!
>>>>>>>>
>>>>>>>> Despite of this mistake, conference communication between two
>>>>>>>> internal users works normally. The problem arises at a call through
>>>>>>>> external provider.
>>>>>>>>
>>>>>>>> We have the following architecture:
>>>>>>>> In a cloud of Amazon EC2 there are 2 servers – opensips and
>>>>>>>> freeswitch, both for NAT for external clients, but have an opportunity to
>>>>>>>> work with each other directly.
>>>>>>>> opensips has the internal address 172.31.0.169 and external 52.
>>>>>>>> *.*.177
>>>>>>>> freeswitch has the internal address 172.31.22.124 and external 52.
>>>>>>>> *.*.198
>>>>>>>>
>>>>>>>> In fact, freeswitch acts only for conferences, and is ready for use
>>>>>>>> of a remote DB on opensips.
>>>>>>>> The auto-nat settings by default didn't work. The problem is in the
>>>>>>>> external profile settings as far as I understand.
>>>>>>>>
>>>>>>>> I have filled and created the following configuration:
>>>>>>>> vars.xml
>>>>>>>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto”/>
>>>>>>>> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198”/> <!—
>>>>>>>> public freeswitch ip —>
>>>>>>>> <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198”/> <!—
>>>>>>>> public freeswitch ip —>
>>>>>>>> <!-- External SIP Profile -->
>>>>>>>> <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>>>>>>>> <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>>>>>>>> <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>>>>>>>> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>>>>> <X-PRE-PROCESS cmd="set"
>>>>>>>> data="external_ssl_dir=$${base_dir}/conf/tls"/>
>>>>>>>>
>>>>>>>> sip_profile/external.xml
>>>>>>>> <param name="rtp-ip" value="$${local_ip_v4}"/>
>>>>>>>> <param name="sip-ip" value="$${local_ip_v4}"/>
>>>>>>>>
>>>>>>>> <param name="ext-rtp-ip" value=“52.*.*.198”/> <!— public
>>>>>>>> freeswitch ip —>
>>>>>>>> <param name="ext-sip-ip" value=“52.*.*.198”/> <!— public
>>>>>>>> freeswitch ip —>
>>>>>>>> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
>>>>>>>> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
>>>>>>>> (that would be logical), but in that case conferences didn't work at all
>>>>>>>> and errors below appeared:
>>>>>>>> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
>>>>>>>> Also I tried to put such configuration:
>>>>>>>> <param name="rtp-ip" value="auto"/>
>>>>>>>> <param name="sip-ip" value="52.*.*.198”/>
>>>>>>>> but it also hasn't helped to solve the problem.
>>>>>>>>
>>>>>>>> autoload_configs/switch.conf.xml
>>>>>>>> <param name="rtp-start-port" value="16384"/>
>>>>>>>> <param name="rtp-end-port" value="32768"/>
>>>>>>>>
>>>>>>>> "sofia status" looks as follows:
>>>>>>>> Name Type
>>>>>>>> Data State
>>>>>>>>
>>>>>>>> =================================================================================================
>>>>>>>> 172.31.22.124 alias
>>>>>>>> internal ALIASED
>>>>>>>> external profile sip:mod_sofia at 52.*.*.198:5060
>>>>>>>> RUNNING (0)
>>>>>>>> external profile sip:mod_sofia at 52.*.*.198:5061
>>>>>>>> RUNNING (0) (TLS)
>>>>>>>> external::*********.com gateway sip:USER@*********.com
>>>>>>>> REGED
>>>>>>>> internal profile sip:mod_sofia at 52.*.*.198:5080
>>>>>>>> RUNNING (0)
>>>>>>>> internal profile sip:mod_sofia at 52.*.*.198:5081
>>>>>>>> RUNNING (0) (TLS)
>>>>>>>>
>>>>>>>> =================================================================================================
>>>>>>>> 2 profiles 1 alias
>>>>>>>>
>>>>>>>> "sofia status profile external" looks as follows:
>>>>>>>>
>>>>>>>> =================================================================================================
>>>>>>>> Name external
>>>>>>>> Domain Name N/A
>>>>>>>> Auto-NAT false
>>>>>>>> DBName sofia_reg_external
>>>>>>>> Pres Hosts
>>>>>>>> Dialplan XML
>>>>>>>> Context public
>>>>>>>> Challenge Realm auto_to
>>>>>>>> RTP-IP 172.31.22.124
>>>>>>>> Ext-RTP-IP 52.*.*.198
>>>>>>>> SIP-IP 172.31.22.124
>>>>>>>> Ext-SIP-IP 52.*.*.198
>>>>>>>> URL sip:mod_sofia at 52.*.*.198:5060
>>>>>>>> BIND-URL sip:mod_sofia at 52.
>>>>>>>> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
>>>>>>>> TLS-URL sip:mod_sofia at 52.*.*.198:5061
>>>>>>>> TLS-BIND-URL sips:mod_sofia at 52.
>>>>>>>> *.*.198:5061;maddr=172.31.22.124;transport=tls
>>>>>>>> HOLD-MUSIC local_stream://moh
>>>>>>>> OUTBOUND-PROXY N/A
>>>>>>>> CODECS IN PCMA
>>>>>>>> CODECS OUT PCMA
>>>>>>>> TEL-EVENT 101
>>>>>>>> DTMF-MODE rfc2833
>>>>>>>> CNG 13
>>>>>>>> SESSION-TO 0
>>>>>>>> MAX-DIALOG 0
>>>>>>>> NOMEDIA false
>>>>>>>> LATE-NEG true
>>>>>>>> PROXY-MEDIA false
>>>>>>>> ZRTP-PASSTHRU true
>>>>>>>> AGGRESSIVENAT false
>>>>>>>> CALLS-IN 0
>>>>>>>> FAILED-CALLS-IN 0
>>>>>>>> CALLS-OUT 0
>>>>>>>> FAILED-CALLS-OUT 0
>>>>>>>> REGISTRATIONS 0
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> What do I adjust wrong? Whether there is some opportunity, to tell
>>>>>>>> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Arthur
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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