[Freeswitch-users] Call is interrupted in 30 seconds, configuring NAT on Amazon EC2
Jurijs Ivolga
jurijs.ivolga at gmail.com
Mon Apr 11 16:04:44 MSD 2016
Hi Stas,
I assume:
UA => OpenSIPS => Freeswitch => VoIP provider
172.31.0.169:5060 - OpenSIPS
172.31.22.124:5060 - Freeswitch
178.*.*.12:5060 - VoIP provider
>From log what you provided it looks like OpenSIPS is not configured
properly...
If you check RFC:
https://tools.ietf.org/html/rfc3665#section-3.1
Successful call looks like this:
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 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.
>
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 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.
>
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,
Jurijs
On Mon, Apr 11, 2016 at 12:59 PM, Стас Тельнов <stasan89 at gmail.com> wrote:
> >>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
>>
>
>
> _________________________________________________________________________
> 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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