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@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, Jurijs On Mon, Apr 11, 2016 at 12:59 PM, Стас Тельнов <stasan89@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@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>:
Hi,
This is not what I need, please use ngrep:
http://jonathanmanning.com/2009/11/17/how-to-sip-capture-using-ngrep-debug-s...
With kind regards,
Jurijs
On 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 sleep
2016-04-08 17:37 GMT+03:00 Jurijs Ivolga <jurijs.ivolga@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@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