Hi, I am using the example "gw-pstn" from Getting started. I followed the guide to setup mediaproxy for handle NAT function. However, I can't make call between Internal IPphone to PSTN. Anyone can help me to analyze why it doesn't work.
After using mediaproxy, I found that some data are stored in the table "Location" compared not using mediaproxy. The field "contact" contains "sip:number@internalIP:port" and the field "received" contains "sip:externalIP:port". Is it correct after using mediaproxy?
I have a look for the log from ngrep and some extra parameters "rport and receieved" in the header.
When I try to make a call from internal IP phone to PSTN, the error "500 FQDN in SDP Media cannot be resolved.". I think it is still a NAT problem and it seems to use the internal IP instead of external IP. As the same time, Cisco shows (no route to destination (3)).
When I try to make a call from PSTN to internal IP phone, the error " Too Many Hops", which is the error message in the openser.cfg file, shown. I do think it is the same reason as the above case.
Below are the log from ngrep during making a call. openser.cfg and mediaproxy.ini are also listed below. Please help to figure it out the problem. Many thanks!
>>>>>>>-------start of the log ( make a call from internal IP
phone to PSTN) --------- interface: eth1 (INTERFACE_IP/255.255.255.240) filter: (ip) and ( port 5060 ) # U NAT_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 INVITE sip:934163634@owthkems01.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 10.0.0.63:5060;branch=z9hG4bKwgnpocaCevAIebRx. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Contact: sip:36418497@10.0.0.63:5060. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 290. . v=0. o=36418497 63608336 94989291 IN IP4 10.0.0.63. s=SIP CALL. c=IN IP4 10.0.0.63. t=0 0. m=audio 6000 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
# U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.0.0.63:5060;branch=z9hG4bKwgnpocaCevAIebRx;rport=5060;received=NAT_PUBLIC_IP. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. CSeq: 1 INVITE. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 OPENSER_PUBLIC_IP:5060 "Noisy feedback tells: pid=30084 req_src_ip=NAT_PUBLIC_IP req_src_port=5060 in_uri=sip:934163634@owthkems01.mydomain.com out_uri=sip:934163634@CISCO_PUBLIC_IP via_cnt==1". .
# U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 INVITE sip:934163634@CISCO_PUBLIC_IP SIP/2.0. Record-Route: sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0. Via: SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Contact: sip:36418497@NAT_PUBLIC_IP:5060. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 315. . v=0. o=36418497 63608336 94989291 IN IP4 10.0.0.63. s=SIP CALL. c=IN IP4 OPENSER_PUBLIC_IPOPENSER_PUBLIC_IP. #it is very strange as it displayed 2 IPs together! t=0 0. m=audio 3502435024 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
# U CISCO_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0,SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:17 GMT. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. .
# U CISCO_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0,SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:17 GMT. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 5664. Allow-Events: telephone-event. Contact: sip:934163634@CISCO_PUBLIC_IP:5060. Record-Route: sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 184. . v=0. o=CiscoSystemsSIP-GW-UserAgent 6412 5565 IN IP4 CISCO_PUBLIC_IP. s=SIP Call. c=IN IP4 CISCO_PUBLIC_IP. t=0 0. m=audio 16398 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes.
# U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:17 GMT. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 5664. Allow-Events: telephone-event. Contact: sip:934163634@CISCO_PUBLIC_IP:5060. Record-Route: sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 184. . v=0. o=CiscoSystemsSIP-GW-UserAgent 6412 5565 IN IP4 CISCO_PUBLIC_IP. s=SIP Call. c=IN IP4 OPENSER_PUBLIC_IP. t=0 0. m=audio 35024 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes.
# U NAT_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 PRACK sip:934163634@CISCO_PUBLIC_IP:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.63:5060;branch=z9hG4bK6CCN6qgnaIMWA8ww. Route: sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Contact: sip:36418497@10.0.0.63:5060. CSeq: 2 PRACK. RAck: 5664 1 INVITE. Content-Length: 0. .
# U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 PRACK sip:934163634@CISCO_PUBLIC_IP:5060 SIP/2.0. Record-Route: sip:OPENSER_PUBLIC_IP;ftag=EInGXmoIRNHvq87x;lr=on. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKc09e.ba203413.0. Via: SIP/2.0/UDP 10.0.0.63:5060;received=NAT_PUBLIC_IP;branch=z9hG4bK6CCN6qgnaIMWA8ww. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Contact: sip:36418497@10.0.0.63:5060. CSeq: 2 PRACK. RAck: 5664 1 INVITE. Content-Length: 0. .
# U CISCO_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKc09e.ba203413.0,SIP/2.0/UDP 10.0.0.63:5060;received=NAT_PUBLIC_IP;branch=z9hG4bK6CCN6qgnaIMWA8ww. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:18 GMT. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 PRACK. Content-Length: 0. .
# U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.63:5060;received=NAT_PUBLIC_IP;branch=z9hG4bK6CCN6qgnaIMWA8ww. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:18 GMT. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 PRACK. Content-Length: 0. .
# U NAT_PUBLIC_IP:15061 -> OPENSER_PUBLIC_IP:5060 ................ # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 .... # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:15061 .... # U OPENSER_PUBLIC_IP:5060 -> 210.22.7.147:2185 .... # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:15060 .... # U CISCO_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 SIP/2.0 500 FQDN Query in SDP Media cannot be resolved. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0,SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:18 GMT. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. .
# U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 ACK sip:934163634@CISCO_PUBLIC_IP SIP/2.0. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. CSeq: 1 ACK. User-Agent: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. .
# U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 SIP/2.0 500 FQDN Query in SDP Media cannot be resolved. Via: SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:18 GMT. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. .
# U NAT_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 ACK sip:934163634@owthkems01.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 10.0.0.63:5060;branch=z9hG4bKwgnpocaCevAIebRx. Route: sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Contact: sip:36418497@10.0.0.63:5060. CSeq: 1 ACK. Content-Length: 0. .
# U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:5060 SIP/2.0 500 FQDN Query in SDP Media cannot be resolved. Via: SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Date: Sat, 17 Dec 2005 06:12:18 GMT. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. .
# U NAT_PUBLIC_IP:5060 -> OPENSER_PUBLIC_IP:5060 ACK sip:934163634@owthkems01.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 10.0.0.63:5060;branch=z9hG4bKwgnpocaCevAIebRx. Route: sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com;tag=2D902FF8-1CD1. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Contact: sip:36418497@10.0.0.63:5060. CSeq: 1 ACK. Content-Length: 0. .
exit 21 received, 0 dropped <<<<<<<<<<<<<<<<<<<-------end of the log ( make a call from internal IP phone to PSTN) ---------
>>>>>>>>>>>>>-------start of the log (make a call from PSTN to
internal IP phone) ----------- interface: eth1 (203.193.46.224/255.255.255.240) filter: (ip) and ( port 5060 ) # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:15061 .... # U OPENSER_PUBLIC_IP:5060 -> NAT_PUBLIC_IP:15060 .... # U CISCO_PUBLIC_IP:56976 -> OPENSER_PUBLIC_IP:5060 INVITE sip:36418498@OPENSER_PUBLIC_IP:5060 SIP/2.0. Via: SIP/2.0/UDP CISCO_PUBLIC_IP:5060. From: sip:34163634@CISCO_PUBLIC_IP;tag=2DA393B0-1816. To: sip:36418498@OPENSER_PUBLIC_IP. Date: Sat, 17 Dec 2005 06:33:27 GMT. Call-ID: DDE5838A-6DFD11DA-82FEA518-578F6E21@CISCO_PUBLIC_IP. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 3722686073-1845301722-2197529880-1469017633. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 15. Remote-Party-ID: sip:34163634@CISCO_PUBLIC_IP;party=calling;screen=yes;privacy=off. Timestamp: 1134801207. Contact: sip:34163634@CISCO_PUBLIC_IP:5060. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 304. . v=0. o=CiscoSystemsSIP-GW-UserAgent 366 9285 IN IP4 CISCO_PUBLIC_IP. s=SIP Call. c=IN IP4 CISCO_PUBLIC_IP. t=0 0. m=audio 18044 RTP/AVP 3 18 4 2 15. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=yes. a=rtpmap:2 G726-32/8000. a=rtpmap:15 G728/8000.
# U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP CISCO_PUBLIC_IP:5060. From: sip:34163634@CISCO_PUBLIC_IP;tag=2DA393B0-1816. To: sip:36418498@OPENSER_PUBLIC_IP. Call-ID: DDE5838A-6DFD11DA-82FEA518-578F6E21@CISCO_PUBLIC_IP. CSeq: 101 INVITE. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 OPENSER_PUBLIC_IP:5060 "Noisy feedback tells: pid=30091 req_src_ip=CISCO_PUBLIC_IP req_src_port=56976 in_uri=sip:36418498@OPENSER_PUBLIC_IP:5060 out_uri=sip:36418498@OPENSER_PUBLIC_IP:5060 via_cnt==1". .
# U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 SIP/2.0 483 Too Many Hops. Via: SIP/2.0/UDP CISCO_PUBLIC_IP:5060. From: sip:34163634@CISCO_PUBLIC_IP;tag=2DA393B0-1816. To: sip:36418498@OPENSER_PUBLIC_IP;tag=b5ab8e75af536221ca172f8fc39505eb.bdf6. Call-ID: DDE5838A-6DFD11DA-82FEA518-578F6E21@CISCO_PUBLIC_IP. CSeq: 101 INVITE. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 OPENSER_PUBLIC_IP:5060 "Noisy feedback tells: pid=30091 req_src_ip=OPENSER_PUBLIC_IP req_src_port=5060 in_uri=sip:36418498@OPENSER_PUBLIC_IP:5060 out_uri=sip:36418498@OPENSER_PUBLIC_IP:5060 via_cnt==16". .
# U CISCO_PUBLIC_IP:56976 -> OPENSER_PUBLIC_IP:5060 ACK sip:36418498@OPENSER_PUBLIC_IP:5060 SIP/2.0. Via: SIP/2.0/UDP CISCO_PUBLIC_IP:5060. From: sip:34163634@CISCO_PUBLIC_IP;tag=2DA393B0-1816. To: sip:36418498@OPENSER_PUBLIC_IP;tag=b5ab8e75af536221ca172f8fc39505eb.bdf6. Date: Sat, 17 Dec 2005 06:33:27 GMT. Call-ID: DDE5838A-6DFD11DA-82FEA518-578F6E21@CISCO_PUBLIC_IP. Max-Forwards: 15. Content-Length: 0. CSeq: 101 ACK. .
exit 7 received, 0 dropped <<<<<<<<<<<<-------start of the log (make a call from PSTN to internal IP phone) -----------
>>>---------start of the openser.cfg ----------------------
# $Id: gw-pstn.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger@onsip.org $ debug=3 fork=yes log_stderror=no
listen=OPENSER_IP port=5060 children=4
dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:[hidden]/openser" log_facility=LOG_LOCAL0 alias=owthkems01.mydomain.com
loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" #loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/xlog.so"
modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://openser:[hidden]/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password")
modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0)
modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients")
modparam("usrloc", "db_mode", 2)
modparam("registrar", "nat_flag", 6)
modparam("rr", "enable_full_lr", 1)
#The default INVITE timer will be allowed 27 seconds to connect. #If we are going to call a PSTN destination then we will set the 'inv_timeout' #AVP to a higher value in order to allow additional timer for the call to #connect. This is show in the PSTN route handler. modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout")
#modparam("permissions", "db_mode", 1) #modparam("permissions", "trusted_table", "trusted")
route {
#xlog("L_INFO","SIP Request: method [$rm] from [$si] [$sp] to [$Ri] [$Rp]\n"); # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; };
if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; };
# ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("OPENSER_IP:5060;nat=yes"); #} else if (method=="ACK") { #zzz # record_route(); } else if (method!="REGISTER") { record_route(); };
# ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); };
# ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) {
if (has_totag() && (method=="INVITE" || method=="ACK")) {
if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; };
route(1); return; };
# ----------------------------------------------------------------- # Call Type Processing Section # is_uri_host_local will determine if the domain portion of the # Request URI is a domain that our SIP proxy is responsible for. # ----------------------------------------------------------------- if (!is_uri_host_local()) { #xlog("L_INFO","SIP route4: line 121\n"); #if (is_from_local() || allow_trusted()) { route(4); route(1); #} else { #sl_send_reply("403", "Forbidden"); #}; return; };
if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; };
lookup("aliases"); if (uri!=myself) { route(4); route(1); return; };
if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; };
route(1); }
route[1] {
# ----------------------------------------------------------------- # Default Message Handler # -----------------------------------------------------------------
# when dealing with NATed clients we must correctly handle reponse # messages that may be heading back to the client. These # response messages are accessible in SER by using a reply_route # block t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") { end_media_session(); };
sl_reply_error(); }; }
route[2] {
# ----------------------------------------------------------------- # REGISTER Message Handler # -----------------------------------------------------------------
sl_send_reply("100", "Trying");
if (!search("^Contact:[ ]**") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); };
if (!www_authorize("","subscriber")) { www_challenge("","0"); return; };
if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; };
consume_credentials();
if (!save("location")) { sl_reply_error(); }; }
route[3] {
xlog("L_INFO","SIP : line 212\n"); # ----------------------------------------------------------------- # INVITE Message Handler # -----------------------------------------------------------------
#if (!allow_trusted()) {
#if (!proxy_authorize("","subscriber")) { #proxy_challenge("","0"); #return; #} else if (!check_from()) { #sl_send_reply("403", "Use From=ID"); #return; #};
#consume_credentials(); #};
if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); };
#if (uri=~"^sip:1[0-9]{10}@") { # strip(1); #};
lookup("aliases"); if (uri!=myself) { route(4); route(1); return; };
#if (uri=~"^sip:011[0-9]*@") { # International PSTN #route(4); #route(5); #return; #};
if (!lookup("location")) { if (uri=~"^sip:9[0-9]{8}@") { # Domestic PSTN route(4); route(5); return; };
sl_send_reply("404", "User Not Found"); return; };
route(4); route(1); }
route[4] {
# ----------------------------------------------------------------- # NAT Traversal Section # -----------------------------------------------------------------
#xlog("L_INFO","SIP route4: line 274\n"); if (isflagset(6) || isflagset(7)) { xlog("L_INFO","SIP : line 275 use_media_proxy\n"); use_media_proxy(); }; }
route[5] {
# ----------------------------------------------------------------- # PSTN Handler # -----------------------------------------------------------------
rewritehost("203.193.46.242"); # INSERT YOUR PSTN GATEWAY IP ADDRESS #xlog("L_INFO","SIP route5: line 287\n");
avp_write("i:45", "inv_timeout");
route(4); route(1); }
onreply_route[1] {
if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) {
if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; };
if (client_nat_test("1")) { fix_nated_contact(); }; } <<<<<<<<<<<<--------------end of openser.cfg------------------
>>>>>------------start of mediaproxy.ini----------------
; ; Configuration file for MediaProxy ;
[Dispatcher] ; ; Section for configuring the proxy dispatcher ; ; The following options are available here: ; ; start Boolean value that specifies if to start the dispatcher. ; Default value: Yes ; ; socket Path to the UNIX socket where the dispatcher receives ; commands from SER. This should match the value for ; mediaproxy_socket in ser.cfg ; Default value: /var/run/proxydispatcher.sock ; ; group Put the socket in this group and make it group writable. ; Default value: ser ; ; defaultProxy Default mediaproxy to use in case the From/To domains ; involved in the call don't define any. ; Valid values for this are: ; ; - None ; don't use any default proxies. domains without ; mediaproxy SRV records won't work ; - /path/to/unix/socket ; use a single MediaProxy server identified by the given ; UNIX socket path ; - IP_or_hostname[:port] ; use a single MediaProxy server identified by its network ; address. The network address consists of an IP address ; or a hostname and an optional port number separated by ; a double colon. If port is missing 25060 will be assumed. ; Examples: ; 10.0.0.1 (connect to 10.0.0.1 on port 25060) ; 10.0.0.1:90 (connect to 10.0.0.1 on port 90) ; mp1.mydomain.com ; mp1.mydomain.com:7000 ; - domain://domain_name ; Use all MediaProxies defined by domain_name, honoring ; their priority and weight to create a cluster of proxies ; with fallback and load balancing capabilities. ; ; Default value: /var/run/mediaproxy.sock ; ;start = yes ;socket = /var/run/proxydispatcher.sock ;group = ser ;defaultProxy = /var/run/mediaproxy.sock
[MediaProxy] ; ; Section for configuring the MediaProxy server ; ; The following options are available here: ; ; start Boolean value that specifies if to start the RTP proxy server. ; Default value: Yes ; ; socket Path to the UNIX socket where MediaProxy receives ; commands from the dispatcher or SER. ; Default value: /var/run/mediaproxy.sock ; ; group Put the socket in this group and make it group writable. ; Default value: ser ; ; listen Network address where MediaProxy receives commands from ; a remote dispatcher. ; Valid values for this are: ; ; - None ; don't listen for network connections at all ; - address[:port] ; listen on the specified address and port ; address can be an IP a hostname or the keyword Any ; (in which case it will listen on 0.0.0.0). If address is ; a hostname, that should map in DNS to an IP address ; present on the machine through an A record. ; If port is missing assume 25060. ; ; Default value: None ; ; allow List of addresses that are allowed to connect to this ; MediaProxy server and send commands. ; They are specified as a comma separated list of entries, with ; each entry being specified in the CIDR network/mask notation ; (ex. 10.0.0.0/8) ; ; In addition simple IP addresses or hostnames are allowed, in ; which case the mask is considered to be 32. ; ; In addition to network ranges/addresses 2 keywords can be used ; for this option: ; None to specify that none is allowed to connect (not very ; useful but this is the default for security reasons) ; Any to specify that anyone is allowed to connect ; (dangerous!) ; ; Example: allow = 10.0.0.0/24, home-pc.mydomain.com, 1.2.3.4 ; ; Default value: None ; ; proxyIP IP address to use to talk to the phones. If not specified, the ; first found will be used. However first found usually means ; first defined in /etc/hosts which may not be what you want. ; If you find that the address that's automatically selected is ; not the one you want, you can specify the right one using this ; option. The address must be one that's present on one of the ; host's interfaces. ; ; portRange The range of ports to use for proxying the rtp streams. ; This option is specified as minport:maxport with minport and ; maxport being even numbers in the range 1024-65536 ; Default value: 35000:65000 ; ; TOS Unless you know what TOS means, leave this option alone. ; The TOS value can be specified either as a decimal number or ; as a hex number in the 0xnn format. ; Default value: 0xb8 ; ; idleTimeout Expire idle sessions after this much time. ; Default 60 seconds ; ; holdTimeout Expire calls on hold after this much time. ; Default value is 3600 seconds ; ; forceClose Forcibly close a RTP session after this many seconds even if ; it's still active. If forceClose is 0, then a session is never ; closed no matter how long it lasts. ; Default value: 0 ; ;start = yes ;socket = /var/run/mediaproxy.sock ;group = ser ;listen = None ;allow = None ;proxyIP = 10.0.0.1 ;portRange = 35000:65000 ;TOS = 0xb8 ;idleTimeout = 60 ;holdTimeout = 3600 ;forceClose = 0 accounting = off
[Accounting] user = dbuser password = dbpass host = dbhost database = radius table = radacct <<<<<<<<<<<-----------end of mediaproxy.ini-------------
U OPENSER_PUBLIC_IP:5060 -> CISCO_PUBLIC_IP:5060 INVITE sip:934163634@CISCO_PUBLIC_IP SIP/2.0. Record-Route: sip:934163634@OPENSER_PUBLIC_IP:5060;nat=yes;ftag=EInGXmoIRNHvq87x;lr=on. Via: SIP/2.0/UDP OPENSER_PUBLIC_IP;branch=z9hG4bKf09e.6b2c5156.0. Via: SIP/2.0/UDP 10.0.0.63:5060;rport=5060;received=NAT_PUBLIC_IP;branch=z9hG4bKwgnpocaCevAIebRx. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-57. From: "36418497" sip:36418497@owthkems01.mydomain.com;tag=EInGXmoIRNHvq87x. To: "934163634" sip:934163634@owthkems01.mydomain.com. Call-ID: xZyhO2TPQ3VEaazQ@10.0.0.63. Contact: sip:36418497@NAT_PUBLIC_IP:5060. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 315. . v=0. o=36418497 63608336 94989291 IN IP4 10.0.0.63. s=SIP CALL. c=IN IP4 OPENSER_PUBLIC_IPOPENSER_PUBLIC_IP. #it is very strange
You have an error in your routing logic. Looks like you call use_media_proxy twice.
regards klaus
as it displayed 2 IPs together! t=0 0. m=audio 3502435024 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
unplug wrote:
Hi, I am using the example "gw-pstn" from Getting started. I followed the guide to setup mediaproxy for handle NAT function. However, I can't make call between Internal IPphone to PSTN. Anyone can help me to analyze why it doesn't work.
After using mediaproxy, I found that some data are stored in the table "Location" compared not using mediaproxy. The field "contact" contains "sip:number@internalIP:port" and the field "received" contains "sip:externalIP:port". Is it correct after using mediaproxy?
I have a look for the log from ngrep and some extra parameters "rport and receieved" in the header.
When I try to make a call from internal IP phone to PSTN, the error "500 FQDN in SDP Media cannot be resolved.". I think it is still a NAT problem and it seems to use the internal IP instead of external IP. As the same time, Cisco shows (no route to destination (3)).
When I try to make a call from PSTN to internal IP phone, the error " Too Many Hops", which is the error message in the openser.cfg file, shown. I do think it is the same reason as the above case.
Probably openser does not recognize that it should handle the SIP request thus it forwards it (again to itself).
You will see the looped requests when sniffing on the loopback device too: ngrep -d any port 5060
This is a problem if uri=myself does not match
If you are using the domain module, maybe you have to add the IP address of openser into the domain table. You could also use is_from_gw from lcr module to detect requests from your gateway.
regards klaus
Thanks klaus. After getting your clue, I have fixed, I think, the problem. As you mentioned, I have inserted the IP in the domain table. Moreover, I have remarked line 238 from the original document to exclude one of the use_media_proxy(). It seems to work fine. Next, I will move on the call forwarding.
On 12/19/05, Klaus Darilion klaus.mailinglists@pernau.at wrote:
unplug wrote:
Hi, I am using the example "gw-pstn" from Getting started. I followed the guide to setup mediaproxy for handle NAT function. However, I can't make call between Internal IPphone to PSTN. Anyone can help me to analyze why it doesn't work.
After using mediaproxy, I found that some data are stored in the table "Location" compared not using mediaproxy. The field "contact" contains "sip:number@internalIP:port" and the field "received" contains "sip:externalIP:port". Is it correct after using mediaproxy?
I have a look for the log from ngrep and some extra parameters "rport and receieved" in the header.
When I try to make a call from internal IP phone to PSTN, the error "500 FQDN in SDP Media cannot be resolved.". I think it is still a NAT problem and it seems to use the internal IP instead of external IP. As the same time, Cisco shows (no route to destination (3)).
When I try to make a call from PSTN to internal IP phone, the error " Too Many Hops", which is the error message in the openser.cfg file, shown. I do think it is the same reason as the above case.
Probably openser does not recognize that it should handle the SIP request thus it forwards it (again to itself).
You will see the looped requests when sniffing on the loopback device too: ngrep -d any port 5060
This is a problem if uri=myself does not match
If you are using the domain module, maybe you have to add the IP address of openser into the domain table. You could also use is_from_gw from lcr module to detect requests from your gateway.
regards klaus
-
Klaus Darilion -
unplug