Alice Bob
| |
| INVITE F1 |
|----------------------->|
| 180 Ringing F2 |
|<-----------------------|
| |
| 200 OK F3 |
|<-----------------------|
| ACK F4 |
|----------------------->|
| Both Way RTP Media |
|<======================>|
| |
| BYE F5 |
|<-----------------------|
| 200 OK F6 |
|----------------------->|
| |
In your case somehow Freeswitch never receive ACK from OpenSIPS.
As you can see following packet was sent several times by Freeswitch:
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@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.
And on this OK Freeswitch should receive ACK, but never receive it from Opensips and because of this FreeSWITCH hang-up call:
U 172.31.22.124:5060 -> 52.*.*.177:5060
BYE sip:8@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@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.
As you can see BYE is strange too... Bye should be sent back to OpenSIPS, but not to 52.*.*.177:5060.
On this point I think you have incorrect Record-Route records:
Good example you can find below, please check 16.12.1.1.
https://www.ietf.org/rfc/rfc3261.txt
Somehow you have following record-route record:
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr> but your opensips never use 5061 port, it is using 5060, so something is incorrect here...
On this point I think you have miss-configured OpenSIPS.
With kind regards,
#All packages on call:I suppose that this is because the server has not received confirmation from one of the customers that the call took place.But phone clients do not send this command.About NORMAL_CLEARING in freeswitch documentation (https://wiki.freeswitch.org/wiki/Hangup_Causes):First BYE packet include reason: NORMAL_CLEARING. BYE packet for caller include reason: ACK Timeout.I analize packets in ngrep.I understand, but I don`t know how fix it.>>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...
Jurijs Ivolga,
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.
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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@192.168.0.110:10977;ob;transport=tls>;+sip.instance="<urn:uuid:E458598F-FBED-B020-670D-167CE2ADB38A>".
To: "8"<sip:8@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@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@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@gmail.com>:With kind regards,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/JurijsOn Fri, Apr 8, 2016 at 6:02 PM, Стас Тельнов <stasan89@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@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@sip0.MY_DOMAIN.com) Running State Change CS_NEW
2016-04-08 10:47:10.503262 [DEBUG] sofia.c:9248 sofia/external/8@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@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@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@sip0.MY_DOMAIN.com) State NEW
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8@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@sip0.MY_DOMAIN.com) State INIT
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88 sofia/external/8@sip0.MY_DOMAIN.com SOFIA INIT
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40 sofia/external/8@sip0.MY_DOMAIN.com Standard INIT
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48 (sofia/external/8@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@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@sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
2016-04-08 10:47:10.503262 [DEBUG] switch_channel.c:2247 (sofia/external/8@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@sip0.MY_DOMAIN.com) State ROUTING
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/8@sip0.MY_DOMAIN.com SOFIA ROUTING
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166 sofia/external/8@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@sip0.MY_DOMAIN.com parsing [public->from_opensips] continue=false
Dialplan: sofia/external/8@sip0.MY_DOMAIN.com Regex (PASS) [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/ break=on-false
Dialplan: sofia/external/8@sip0.MY_DOMAIN.com Action transfer(${destination_number} XML default)
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216 (sofia/external/8@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@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@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@sip0.MY_DOMAIN.com) State EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/8@sip0.MY_DOMAIN.com SOFIA EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258 sofia/external/8@sip0.MY_DOMAIN.com Standard EXECUTE
EXECUTE sofia/external/8@sip0.MY_DOMAIN.com transfer(7906******* XML default)
2016-04-08 10:47:10.503262 [DEBUG] switch_ivr.c:2085 (sofia/external/8@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@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@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@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@sip0.MY_DOMAIN.com) State ROUTING
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/8@sip0.MY_DOMAIN.com SOFIA ROUTING
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166 sofia/external/8@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@sip0.MY_DOMAIN.com parsing [default->unloop] continue=false
Dialplan: sofia/external/8@sip0.MY_DOMAIN.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/8@sip0.MY_DOMAIN.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/8@sip0.MY_DOMAIN.com parsing [default->tod_example] continue=true
Dialplan: sofia/external/8@sip0.MY_DOMAIN.com Date/Time Match (PASS) [tod_example] break=on-false
Dialplan: sofia/external/8@sip0.MY_DOMAIN.com Action set(open=true)
Dialplan: sofia/external/8@sip0.MY_DOMAIN.com parsing [default->outbound_calls_to_freelycall] continue=false
Dialplan: sofia/external/8@sip0.MY_DOMAIN.com Regex (PASS) [outbound_calls_to_freelycall] destination_number(7906*******) =~ /^(.+)/ break=on-true
Dialplan: sofia/external/8@sip0.MY_DOMAIN.com Action set(hangup_after_bridge=true)
Dialplan: sofia/external/8@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@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@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@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@sip0.MY_DOMAIN.com) State EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/8@sip0.MY_DOMAIN.com SOFIA EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258 sofia/external/8@sip0.MY_DOMAIN.com Standard EXECUTE
EXECUTE sofia/external/8@sip0.MY_DOMAIN.com set(open=true)
2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/8@sip0.MY_DOMAIN.com [open]=[true]
EXECUTE sofia/external/8@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@sip0.MY_DOMAIN.com [hangup_after_bridge]=[true]
EXECUTE sofia/external/8@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@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@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@8000
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec sofia/external/8@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@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@8000
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485 sofia/external/8@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@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@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@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@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@sip0.MY_DOMAIN.com!
2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468 (sofia/external/8@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@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@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@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@sip0.MY_DOMAIN.com] has been answered
2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767 (sofia/external/8@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@sip0.MY_DOMAIN.com entering state [terminating][0]
2016-04-08 10:47:53.343283 [NOTICE] sofia.c:7779 Hangup sofia/external/8@sip0.MY_DOMAIN.com [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:699 sofia/external/8@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@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@sip0.MY_DOMAIN.com]
2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:1692 sofia/external/8@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@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@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@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@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@sip0.MY_DOMAIN.com) State HANGUP
2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel sofia/external/8@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@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@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@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@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@sip0.MY_DOMAIN.com) State REPORTING
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104 sofia/external/8@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@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@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@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@sip0.MY_DOMAIN.com) Ended
2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close Channel sofia/external/8@sip0.MY_DOMAIN.com [CS_DESTROY]
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630 (sofia/external/8@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@sip0.MY_DOMAIN.com) State DESTROY
2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341 sofia/external/8@sip0.MY_DOMAIN.com SOFIA DESTROY
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111 sofia/external/8@sip0.MY_DOMAIN.com Standard DESTROY
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640 (sofia/external/8@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 sleep2016-04-08 17:37 GMT+03:00 Jurijs Ivolga <jurijs.ivolga@gmail.com>:With kind regards,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.JurijsOn Fri, Apr 8, 2016 at 5:34 PM, Стас Тельнов <stasan89@gmail.com> wrote:I already tried disabling timers, does not work.2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar@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@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@52.*.*.198:5060 RUNNING (0)
external profile sip:mod_sofia@52.*.*.198:5061 RUNNING (0) (TLS)
external::*********.com gateway sip:USER@*********.com REGED
internal profile sip:mod_sofia@52.*.*.198:5080 RUNNING (0)
internal profile sip:mod_sofia@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@52.*.*.198:5060
BIND-URL sip:mod_sofia@52.*.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
TLS-URL sip:mod_sofia@52.*.*.198:5061
TLS-BIND-URL sips:mod_sofia@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org