- sr-users - kamailio.org

[Serusers] SERWEB Installation
by vincenzo mennella 16 Jan '07

16 Jan '07

Can someone please help me to explain this error? When I try to open firefox ...http://lolcalhost/serweb there is the following error in /var/log/apache2/erro_log: [Mon Jan 15 13:48:00 2007] [error] [client 127.0.0.1] client denied by server configuration: /var/www/html/error_log thanks vincenzo ___________________________________ Vinci i biglietti per FIFA World Cup in Germania! yahoo.it/concorso_messenger

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[Serusers] Ottendorf compilation errors
by raj gopal 16 Jan '07

16 Jan '07

Hi, I am newbie to linux and SER. While installing Ottendorf version of SER , I am facing following compilation errors /usr/bin/ld: cannot find -l_ser_cds collect2: ld returned 1 exit status make[1]: *** [dialog.so] Error 1 /usr/bin/ld: cannot find -l_ser_xcap collect2: ld returned 1 exit status make[1]: *** [pa.so] Error 1 /usr/bin/ld: cannot find -l_ser_xcap collect2: ld returned 1 exit status make[1]: *** [presence_b2b.so] Error 1 /usr/bin/ld: cannot find -l_ser_xcap collect2: ld returned 1 exit status make[1]: *** [rls.so] Error 1 What would be the problem??? -Venkat

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[Serusers] SER 0.10.99 (Carpo) - Cancel from CP-7912G not recognized
by Theodor Meier 16 Jan '07

16 Jan '07

Hello, the VoIP Provider Carpo (www.carpo.com) uses Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)). When using the CP-7912G with this server, the CANCEL reuqests aren't processed. When testing myself, has the openser (it was easier to set up with deb-packages) the same behaviour like the SER? Following a log from the Cisco CP-7912G with the 4 CANCEL that SER doesn't recognize. Is it due to configuration or aren't the SIP-Messages from 7912g not right? This log was recorded without the CP-7912G ConnectMode Parameter set to 0×00400000 (set bit 22 to use the received ip address for NAT) Before the 2 INVITE: ---- to SER INVITE sip:01608438483@sip.carpo.de;user=phone SIP/2.0 ---- from SER SIP/2.0 407 Proxy Authentifikation erforderlich ---- To SER ACK sip:01608438483@sip.carpo.de;user=phone SIP/2.0 ---- [0:0] Tx Msg to 80.95.252.5:5060 INVITE sip:01608438483@sip.carpo.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 INVITE Contact: Use of a Display Name <sip:wa.mayer@192.168.178.41:5062;transport=udp> User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="77323afc4bea0fb5e2cceb50bff9e404" Expires: 300 Content-Length: 285 Content-Type: application/sdp v=0 o=wa.mayer 2928 2928 IN IP4 192.168.178.41 s=Cisco 7912 SIP Call c=IN IP4 192.168.178.41 t=0 0 m=audio 30004 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ---- [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.178.41:5062;rport=5062;received=84.150.194.222;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 INVITE P-Nat: Yes Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.14:5070 "Noisy feedback tells: pid=22183 req_src_ip=80.95.252.5 req_src_port=5060 in_uri=sip:01608438483@sip.carpo.de;user=phone out_uri=sip:491608438483@217.188.44.79 via_cnt==2" Response=100; Trying 0:30;2,0,0,0, ---- [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 183 Sitzungsfortschritt Record-Route: <sip:80.95.252.14:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on> Record-Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;rport=5062;received=84.150.194.222;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone>;tag=e0aef7928df800c0a39d563783d2a332 Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 INVITE Contact: <sip:aa+wa.mayer@80.95.252.29:5090> Content-Type: application/sdp Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 382 Warning: 392 80.95.252.29:5090 "Noisy feedback tells: pid=5490 req_src_ip=80.95.252.14 req_src_port=5070 in_uri=sip:491608438483@217.188.44.79 out_uri=sip:491608438483@217.188.44.79 via_cnt==0" v=0 o=- 4067017 0 IN IP4 62.53.226.18 s=Cisco SDP 0 c=IN IP4 80.95.247.69 t=0 0 m=audio 2544 RTP/AVP 0 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 a=X-cap: 2 image udptl t38 a=nortpproxy:yes AVT: payload=101 Response=183; Session Progress 18x Response has SDP Frames per packet=2 SetG723Mode: 2 0 [0]Enable encoder 0 RTP TX[0]:SSRC_ID = 9e8b6574 RTP Tx Init: 0, 0 [0]Enable encoder 0 [0]Start/Resume Media-> 80.95.247.69:2544 (pt=0) ---- [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 180 Klingelt Record-Route: <sip:80.95.252.14:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on> Record-Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;rport=5062;received=84.150.194.222;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone>;tag=e0aef7928df800c0a39d563783d2a332 Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 INVITE Content-Type: application/sdp Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.29:5090 "Noisy feedback tells: pid=5490 req_src_ip=80.95.252.14 req_src_port=5070 in_uri=sip:491608438483@217.188.44.79 out_uri=sip:491608438483@217.188.44.79 via_cnt==0" Response=180; Ringing ---- [0]DPKT 1st: 91375780 91375620, pt 0 SCC Cmd[0:0]CANCEL [0:0]Cancel INVITE [0]CANCEL Retry 0 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip::80.95.252.14:5070;avp=B50DCA SIP/2.0 Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="31b050a88b23b31a915df6f6ef8cff28" Content-Length: 0 [0]CANCEL Retry 1 ----- [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip::80.95.252.14:5070;avp=B50DCA SIP/2.0 Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="31b050a88b23b31a915df6f6ef8cff28" Content-Length: 0 [0]CANCEL Retry 2 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip::80.95.252.14:5070;avp=B50DCA SIP/2.0 Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="31b050a88b23b31a915df6f6ef8cff28" Content-Length: 0 [0]CANCEL Retry 3 ----- [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip::80.95.252.14:5070;avp=B50DCA SIP/2.0 Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="31b050a88b23b31a915df6f6ef8cff28" Content-Length: 0 [0]CANCEL Retry 4 ---- [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip::80.95.252.14:5070;avp=B50DCA SIP/2.0 Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="31b050a88b23b31a915df6f6ef8cff28" Content-Length: 0 BYE/CAN Failed (Proxy No Resp) SCC Cmd[0:0]RELEASE >>> SIP Soft Reset [0:0] <<< ___________________________________________________________ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de

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[Serusers] SER 0.10.99 (Carpo) - Cancel from CP-7912G not recognized - second
by Theodor Meier 15 Jan '07

15 Jan '07

I found another CANCEL problem where the Request-URI has not two double dots (sip:: ..) The call goes from Carpo to Carpo, wheras in my previous mail the call went to a gateway. What I found in the log is, that the branch Parameter is Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK68fec26c25c87141 till PRACK, there and in the OK from the server this changes. Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK4abf1bf17596f52 the CANCEL request from the phone has the initial branch value. Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 finally the CANCEL is rejected SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 What is the reason for the rejection of the CANCEL? Is this a server or a phone problem? --------------------------- calling 23200302 SCC Cmd[0:0]CLIP SCC Cmd[0:0]CALL proxy = sip.carpo.de:5060 [0]INVITE Retry 0 [0:0] Tx Msg to 80.95.252.5:5060 INVITE sip:23200302@sip.carpo.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK56e50100b962975c From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 1 INVITE Contact: <sip:paula.mayer@84.150.187.204:5066;transport=udp> User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 289 Content-Type: application/sdp v=0 o=paula.mayer 17339 17339 IN IP4 192.168.178.42 s=Cisco 7912 SIP Call c=IN IP4 84.150.187.204 t=0 0 m=audio 30006 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 407 Proxy Authentifikation erforderlich Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK56e50100b962975c From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=68e836eb5e2cffe6b7aefb548a3e0a9b.43de Call-ID: 2749835686(a)84.150.187.204 CSeq: 1 INVITE P-Nat: Yes Proxy-Authenticate: Digest realm="carpo", nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8" Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.14:5070 "Noisy feedback tells: pid=22174 req_src_ip=80.95.252.5 req_src_port=5060 in_uri=sip:23200302@sip.carpo.de;user=phone out_uri=sip:23200302@sip.carpo.de;user=phone via_cnt==2" Response=407; Proxy Authentication Required [0]TxMsg to 80.95.252.5:5060 ACK sip:23200302@sip.carpo.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK56e50100b962975c From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=68e836eb5e2cffe6b7aefb548a3e0a9b.43de Call-ID: 2749835686(a)84.150.187.204 CSeq: 1 ACK User-Agent: Cisco-CP7912/8.0.0-060111A Content-Length: 0 [0]INVITE Retry 0 [0:0] Tx Msg to 80.95.252.5:5060 INVITE sip:23200302@sip.carpo.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 INVITE Contact: <sip:paula.mayer@84.150.187.204:5066;transport=udp> User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="271e020ff6474a1304077083c9beca99" Expires: 300 Content-Length: 289 Content-Type: application/sdp v=0 o=paula.mayer 17358 17358 IN IP4 192.168.178.42 s=Cisco 7912 SIP Call c=IN IP4 84.150.187.204 t=0 0 m=audio 30006 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 INVITE P-Nat: Yes Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Response=100; Trying [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 180 Klingelt Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK68fec26c25c87141 Record-Route: <sip:80.95.252.5;lr=on> Record-Route: <sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on> Record-Route: <sip:80.95.252.5;lr=on> From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 INVITE Require: 100rel RSeq: 1 Contact: <sip:wa.mayer@84.150.187.204:5062;transport=udp> Server: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 Response=180; Ringing [0]PRACK Retry 0 [0:0] Tx Msg to 80.95.252.5:5060 PRACK sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK4abf1bf17596f52 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 3 PRACK RAck: 1 2 INVITE User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="34b5d81e9d984c39b09f586b09b12f04" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK4abf1bf17596f52 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 3 PRACK Server: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 0 SCC Cmd[0:0]CANCEL [0:0]Cancel INVITE [0]CANCEL Retry 0 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="5a04c4c23e034a4192e964fa468c9461" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=a8c29d96e482b9ddbb33ac37669b77b1.da3a Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.5:5060 "Noisy feedback tells: pid=16875 req_src_ip=84.150.187.204 req_src_port=5066 in_uri=sip:wa.mayer@84.150.187.204:5062 out_uri=sip:wa.mayer@84.150.187.204:5062 via_cnt==1" [0]CANCEL Retry 1 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="5a04c4c23e034a4192e964fa468c9461" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=a8c29d96e482b9ddbb33ac37669b77b1.da3a Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.5:5060 "Noisy feedback tells: pid=16872 req_src_ip=84.150.187.204 req_src_port=5066 in_uri=sip:wa.mayer@84.150.187.204:5062 out_uri=sip:wa.mayer@84.150.187.204:5062 via_cnt==1" [0]CANCEL Retry 2 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="5a04c4c23e034a4192e964fa468c9461" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=a8c29d96e482b9ddbb33ac37669b77b1.da3a Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.5:5060 "Noisy feedback tells: pid=16874 req_src_ip=84.150.187.204 req_src_port=5066 in_uri=sip:wa.mayer@84.150.187.204:5062 out_uri=sip:wa.mayer@84.150.187.204:5062 via_cnt==1" [0]CANCEL Retry 3 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="5a04c4c23e034a4192e964fa468c9461" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=a8c29d96e482b9ddbb33ac37669b77b1.da3a Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux3:00;2,0,0,0, )) Content-Length: 0 Warning: 392 80.95.252.5:5060 "Noisy feedback tells: pid=16871 req_src_ip=84.150.187.204 req_src_port=5066 in_uri=sip:wa.mayer@84.150.187.204:5062 out_uri=sip:wa.mayer@84.150.187.204:5062 via_cnt==1" [0]CANCEL Retry 4 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="5a04c4c23e034a4192e964fa468c9461" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=a8c29d96e482b9ddbb33ac37669b77b1.da3a Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.5:5060 "Noisy feedback tells: pid=16868 req_src_ip=84.150.187.204 req_src_port=5066 in_uri=sip:wa.mayer@84.150.187.204:5062 out_uri=sip:wa.mayer@84.150.187.204:5062 via_cnt==1" BYE/CAN Failed (Proxy No Resp) SCC Cmd[0:0]RELEASE >>> SIP Soft Reset [0:0] <<< 3:30;2,0,0,0, [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 480 Temporarily Not Available Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 INVITE Server: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 [0]TxMsg to 80.95.252.5:5060 ACK sip:23200302@sip.carpo.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 ACK User-Agent: Cisco-CP7912/8.0.0-060111A Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 480 Temporarily Not Available Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 INVITE Server: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 ___________________________________________________________ Der frühe Vogel fängt den Wurm. Hier gelangen Sie zum neuen Yahoo! Mail: http://mail.yahoo.de

2 1

[Serusers] Re: Users presence status
by Vaclav Kubart 15 Jan '07

15 Jan '07

According to PIDF RFC there MUST be <basic> with value "open" or "closed". I guess, that this PUBLISH request should be rejected with non-2xx response. Is it? BTW, PA is not able to handle "partial PIDF" - you have to use PIDF. Presence status can be stored into database if you set "use_db" module parameter to 1. You can publish document like this: <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" entity="pres:venkat@10.187.83.218"> <tuple id="000s3"> <status> <basic>open</basic> </status> <contact>venkat(a)10.187.83.218</contact> </tuple> <dm:person id="p1"> <rpid:activities> <busy/> </rpid:activities> </dm:person> </presence> I'm not sure if it is strictly according to RPID specification, but might be. Much simpler it could be like this: <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" entity="pres:venkat@10.187.83.218"> <tuple id="000s3"> <status> <basic>open</basic> </status> <contact>venkat(a)10.187.83.218</contact> <note>busy</note> </tuple> </presence> Vaclav On Mon, Jan 15, 2007 at 01:34:04PM +0530, raj gopal wrote: > Thanks for reply, > > I am sending PUBLISH method with following simplified PIDF content. > > <pidf-part:presence xmlns:pidf-part="urn:ietf:params:xml:ns:pidf-partial" > xmlns="urn:ietf:params:xml:ns:pidf" > xmlns:et="urn:ietf:params:xml:ns:pidf:rpid-tuple" > xmlns:ep="urn:ietf:params:xml:ns:pidf:status:rpid-status" > xmlns:dc="urn:ietf:params:xml:ns:pidf:status:devcaps" entity=" > venkat(a)10.187.83.218" version="1" state="full"> > <tuple id="000s3"> > <status> > <basic>busy</basic> > </status> > <contact>venkat(a)10.187.83.218</contact> > </tuple> > > But, server sends following NOTIFY msg > > <?xml version="1.0" encoding="UTF-8"?> > <presence xmlns="urn:ietf:params:xml:ns:pidf" entity=" > pres:venkat@10.187.83.218"> > <tuple id="0xb626fba4x52c779c5x45ab1b6d"> > <status><basic>open</basic></status> > <contact priority="0.00">sip:10.187.83.222:5060</contact> > </tuple> > </presence> > > The server sends either closed/open. But it dont send busy status. > > Am i doing something wrong? is status is stored in database?? > > -venkat > > > On 1/15/07, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > > > >Hello, > >SER understands only basic status what means open/closed, but it allows > >clients to publish other info in extension elements (look into PIDF > >specification - RFC 3863). Such elements are not parsed by SER, but they > >are sent in NOTIFY to watchers. > > > >If you don't want to use PIDF extension elements, you can send > >additional status info in <note> elements - many clients do it in this > >way. > > > > Vaclav > > > >On Mon, Jan 15, 2007 at 12:11:31PM +0530, raj gopal wrote: > >> Hi Vaclav, > >> > >> I am using SER as sip and presence server. According to my understanding > >SER > >> supports "open" and "close" presence status. But, my application > >required > >> presence status like busy, tolunch etc.. When client sends PUBLISH with > >his > >> presence status , how does SER process it? while server sending NOTIFY, > >does > >> it consider the status published by client?? > >> > >> Please suggest me to achieve above requirement. > >> > >> -Venkat > >

1 0

[Serusers] Users presence status
by raj gopal 15 Jan '07

15 Jan '07

Hi, Does presence module supports user presence status other than closed/open. -venkat

2 1

Re: [Users] Multi Call Legs Accounting
by Daniel-Constantin Mierla 14 Jan '07

14 Jan '07

Hello, are you proxying the media stream all the time now? Or just when the one of the party is behind the nat? I have in mynd the sceanrio when the initial callee is behind nat, it sends back redirect, and the new destination is public... Cheers, Daniel On 01/12/07 14:36, Howard Tang wrote: > Hi Daniel, > > Thank you very much, the problem fixed. > > Regards, > Howard > > On 1/11/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> Hello, >> >> as i see in your failure route you close mediaproxy nat handling -- try >> to keep it on. ALso, the contacts in redirect message might be natted >> even when initial destination wasn't -- you have to deal with this case >> as well (start nat handling). >> >> Cheers, >> Daniel >> >> >> On 01/09/07 14:23, Howard Tang wrote: >> > Hi Daniel, >> > >> > by using the following code in the failure_route block, i am able to >> > have a better accounting record for billing. However, i am having one >> > way audio problem which i believe it is related to NAT. I have tested >> > forward a call on a UA which is on public ip, which works prefectly. >> > However, when i tested it with UA which is behind NAT, one way audio >> > happen when the UA send 302 reply to forward a call.. >> > >> > failure_route[1] >> > { >> > end_media_session(); >> > # Handle 302 Redirects >> > if (t_check_status("302")) { >> > get_redirects("*"); >> > t_relay(); >> > }; >> > } >> > >> > Anyone has an idea on how i can fix this? >> > >> > Regards, >> > Howard >> > >> > >> > On 1/9/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> >> Hello, >> >> >> >> when the call A -> B is forwarded to C, then you need to have in your >> >> script: >> >> >> >> avp_write("A", "$avp(i:110)"); >> >> avp_write("B", "$avp(i:111)"); >> >> avp_write("B", "$avp(i:110)"); >> >> avp_write("C", "$avp(i:111)"); >> >> >> >> A, B, C are the SIP addresses of the parties involved in this call. >> >> >> >> Cheers, >> >> Daniel >> >> >> >> On 01/08/07 23:14, Howard Tang wrote: >> >> > Hi, >> >> > >> >> > Did anyone successfully setup call accounting for call forward? >> I have >> >> > been searching for a working example for few days on google, I >> still >> >> > couldn't find any working example. >> >> > >> >> > The config I have at the moment is >> >> > >> >> > modparam("acc", "multi_leg_enabled", 1) >> >> > modparam("acc", "src_leg_avp_id", 110) >> >> > modparam("acc", "dst_leg_avp_id", 111) >> >> > >> >> > In the main route, >> >> > if (!method=="REGISTER" && !method=="NOTIFY") { >> >> > avp_write("$from/username", "$avp(i:110)"); >> >> > avp_write("$ruri/username", "$avp(i:111)"); >> >> > }; >> >> > >> >> > Let say A call B and B forward call to C, the record in the acc >> table >> >> > is all >> >> > A to C without B in any one of the record for billing purpose. The >> >> > Call forward was set on the UA. >> >> > >> >> > Can anyone give me a hand on this please? >> >> > >> >> >> > >> >

2 1

[Serusers] proxy_challenge and E60
by Stefan Tichy 14 Jan '07

14 Jan '07

Hi, a nokia E60 is registered to SER 0.9.6. The response to a INVITE by the E60 is not a proper 407 response but just a small UDP paket (46 bytes). I cannot be shure that it is really the response and not some data send from time to time, but there is no other response. Ethereal does not identify this paket as SIP if (!proxy_authorize("domain","subscriber")) { log(3,"proxy_challenge"); proxy_challenge("domain","0"); break; } The phone is registered and can be called. Just INVITE does not work. Compression and security flags in phones config are set to off. The firmware of the phone is somewhat old, but probably I am not the only one using E60 with old firmware. The INVITE itself looks not bad and does work for asterisk 1.4 server and other systems. Even if the INVITE would contain garbage SER should send some SIP response with 5XX code. It is strange. Any help appreciated -- Stefan Tichy <seruser(a)pi4tel.de>

4 4

[Users] prevent double registration
by Howard Tang 14 Jan '07

14 Jan '07

Hi, What do i need to do if i want to prevent same user using two different UAs to register to openser in different location? e.g. A has 2 ATA, one at 222.222.222.222 and one in 111.111.111.111, he has put same login detials in the ATA, and try to register openser on 123.123.123.123 Both ATA will successfully register to the proxy, i want to prevent this, what can be done? Regards, Howard

3 4

[Users] Reduce Retry Time
by Arun Kumar 14 Jan '07

14 Jan '07

Hi, I've configured the ser with my lcr and its working fine. If first gateway fails after a wait of 30sec it loads another gw and passes the call to my asterisk server. Now I want to reduce this 30sec retry time to 5sec so in case my first gw is not reachable then after 5sec it should load the next gateway. thanks arun

4 5

[Serusers] Status of SER 2.0 (Ottendorf) documentation
by Greger V. Teigre 14 Jan '07

14 Jan '07

Just a quick summary of documentation status for SER 2.0 (Ottendorf): ------------------- * http://www.iptel.org/ser/doc/010whatsnew contains a set of book pages introducing new features * ser-basic.cfg and ser.cfg in etc/ directory of source package contains a non-database and mysql database example that should be a good basis for testing * http://www.iptel.org/new_major_ser_pre_release_ottendorf_is_out_for_testing provides and overview of new stuff and a step-wise guide to get started * Module documentation is available online at http://www.iptel.org/views/moduledocs (I personally prefer the table overview for quick browsing: http://www.iptel.org/views/tableofmodules). Most of the important modules are available online, but see below for module documentation status What's left? ---------------- * Overall review of all readme files in source tree, as well as review of completeness/correctness of module documentation * Completion of module documentation * Update SER - Getting Started document and configuration files * Complete What's New 0.10.x as a migration guide; evaluate what to extract in a separate manual for SER functionality (updated admin guide) * Make sure all new features have been documented properly either in module docs or migration guide If you want to contribute in some way, please volunteer! (ex. to review module documentation for your favorite module...) g-) ---------------------------------------------------------------------------- Most of the modules are documented online at iptel.org. The documentation is checked out every night (4am CET) from the cvs and generated, so any changes made the previous day will be available online the next day. The modules below are not yet available online and the following have no readme or docbook documentation in CVS (anybody with documentation for these, please make a sound!): acc_db acc_radius ctl dialog fifo xmlrpc These modules have README files available in CVS, but have not been converted to docbook format yet (in the works): db_ops diversion eval mediaproxy postgres timer tls

1 0

[Users] get_hdr_field: bad to header...
by Lars Sundqvist 14 Jan '07

14 Jan '07

Hello, We have a few hundred users registered to a openser we are running and everything works fine, however i get som error messages in the logs I would like to get rid of... //sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: ERROR:parse_from_header: bad from header /sbin/openser[28867]: xl_get_from: ERROR cannot parse FROM header /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: xl_get_to: ERROR cannot parse TO header /sbin/openser[28867]: [REGISTER] from [<null>] to [<null>] /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: ERROR:maxfwd:is_maxfwd_present : parsing MAX_FORWARD header failed! /sbin/openser[28867]: ERROR:parse_from_header: bad from header /sbin/openser[28867]: insert_RR(): From parsing failed /sbin/openser[28867]: record_route(): Error while inserting Record-Route line /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: find_first_route: Error while parsing headers /sbin/openser[28867]: analyzing REGISTER request /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: find_credentials(): Error while parsing headers /sbin/openser[28867]: pre_auth(): Error while looking for credentials /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed /sbin/openser[28867]: pre_auth(): Error while sending 400 reply / Anyone has a clue what I can do to debug this to find out where the request are coming from or what can be done to stop the invalid(?) requests from being processed and clutter my logs? Thanks Lars Sundqvist

6 9

[Serusers] How to create sip account Ser 0.10
by jean sigue 12 Jan '07

12 Jan '07

All, I have a fresh install ser 0.10 running with mysql. The utility serctl cannot be use to create sip account because the credentials table replaces the subscriber table. Can someone tell me how to create an account ? [root@itiah MySQL-python-1.2.1_p2]# ps -ef | grep ser root 87 7 0 Jan08 ? 00:00:00 [kseriod] root 1696 1 0 Jan08 ? 00:00:00 /usr/bin/hidd --server mysql 9972 9936 0 Jan08 pts/0 00:00:01 /usr/libexec/mysqld --defaults-file=/etc/my.cnf --basedir=/usr --datadir=/var/lib/mysql --user=myl --pid-file=/var/run/mysqld/mysqld.pid --skip-external-locking --socket=/var/lib/mysql/mysql.sock root 21031 1 0 Jan09 ? 00:00:00 ser root 21038 21031 0 Jan09 ? 00:00:00 ser root 21040 21031 0 Jan09 ? 00:00:00 ser root 21042 21031 0 Jan09 ? 00:00:00 ser root 21044 21031 0 Jan09 ? 00:00:00 ser root 21046 21031 0 Jan09 ? 00:00:00 ser root 21048 21031 0 Jan09 ? 00:00:00 ser root 21050 21031 0 Jan09 ? 00:00:00 ser root 21052 21031 0 Jan09 ? 00:00:00 ser root 21053 21031 0 Jan09 ? 00:00:00 ser root 21056 21031 0 Jan09 ? 00:00:00 ser root 21058 21031 0 Jan09 ? 00:00:00 ser root 21060 21031 0 Jan09 ? 00:00:00 ser root 21062 21031 0 Jan09 ? 00:00:00 ser root 21064 21031 0 Jan09 ? 00:00:00 ser root 21066 21031 0 Jan09 ? 00:00:00 ser root 21068 21031 0 Jan09 ? 00:00:00 ser root 21070 21031 0 Jan09 ? 00:00:10 ser root 21072 21031 0 Jan09 ? 00:00:00 ser root 21075 21031 0 Jan09 ? 00:00:00 ser root 21076 21031 0 Jan09 ? 00:00:00 ser root 21077 21031 0 Jan09 ? 00:00:00 ser root 21078 21031 0 Jan09 ? 00:00:00 ser root 21079 21031 0 Jan09 ? 00:00:00 ser root 21080 21031 0 Jan09 ? 00:00:00 ser root 21081 21031 0 Jan09 ? 00:00:00 ser root 21082 21031 0 Jan09 ? 00:00:00 ser root 21083 21031 0 Jan09 ? 00:00:00 ser root 21084 21031 0 Jan09 ? 00:00:00 ser root 17014 31475 0 19:24 pts/2 00:00:00 grep ser

3 3

[Serusers] SER + Asterisk Voice Mail, delay before redirect
by inge 12 Jan '07

12 Jan '07

Hi all, I work on an existing configuration of SER. I'm not able to find the parameter wich define the time in seconds before SER redirect call to Asterisk Voice Mail. In fact, I want to change this timing to put it in SER DB with a usr_preferences table like. Someone have an idea about the kind of line which could be define it ? Thanks for your help.

3 6

[Users] adding aliases does not work
by Cristian Caprar 12 Jan '07

12 Jan '07

Hi, Adding aliases with openserctl does not seem to be working anymore in the latest CVS snapshot. I run the command: /usr/local/sbin/openserctl alias add 00162123456 sip:user@domain.local And the output is: database engine 'MYSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 check_db_alias: alias counter=0 entering fifo_cmd ul_add aliases 00162123456(a)domain.local sip:user@domain.local 0 1.00 0 0 4294967295 ERROR: Error opening openser's FIFO /tmp/openser_fifo ERROR: Make sure you have line fifo=/tmp/openser_fifo in your config The same command used to work with a snapshot I got around 20 dec. OpenSER starts correctly. Thanks, Cristian

2 2

[Users] Meaning of CS_DIRTY
by Bill Neely 12 Jan '07

12 Jan '07

I get the follwing response from openserctl show ul ~~~Contact(0x284a6880)~~~ domain : 'location' aor : '1020111' Contact : 'sip:1020111@192.168.0.198:5060' Expires : 3592 q : Call-ID : '4ef617d2-8e70ad17(a)192.168.0.198' CSeq : 4249 User-Agent: 'Linksys/SPA3102-5.1.5(GWa)' received : 'sip:67.188.231.188:62292' Path : '' State : CS_DIRTY Flags : 0 Sock : 66.224.20.35:5060 (0x8123fb0) Methods : 5183 next : 0x0 prev : 0x0 ~~~/Contact~~~~ Can anybody tell me the meaning of the CS_DIRTY state reply? -- 702-874-3833 1-866-553-3833

2 1

[Serusers] ser logging into the file? - ser 0.10.99
by TZieleniewski 12 Jan '07

12 Jan '07

Hi ! Is there in ser 0.10.99 a way to send the output to the file instead of syslog?? Previously for ser 0.9.7 I had it done throught the serctl tool. I had the logstderror=yes in my ser.cfg and the output was redirected to log file pointed by SER_LOGFILE inside serctl. Tomasz

4 4

[Serusers] SER with LCR Config
by Arun Kumar 12 Jan '07

12 Jan '07

Hi All, I've posted the working config of LCR with SER and NAT on voipbhiwani.blogspot.com b'coz list is not allowing this much big config file. thanks Arun

1 0

Re: [Serusers] ser-0.10.99-dev64 (ser_ctl)]
by TZieleniewski 11 Jan '07

11 Jan '07

jean sigue napisał(a): > I create the database but the credentials table replaces the old > subscriber table 0.9.6 > > >> From: Hendrik Scholz <hendrik.scholz(a)freenet-ag.de> >> To: jean sigue <jeansigue(a)hotmail.com> >> CC: serusers(a)lists.iptel.org >> Subject: Re: [Serusers] ser-0.10.99-dev64 (ser_ctl) >> Date: Thu, 11 Jan 2007 14:46:36 +0100 >> >> Hi! >> >> jean sigue wrote: >> > MySql password: >> > ERROR 1146 (42S02) at line 1: Table 'ser.subscriber' doesn't exist >> >> Did you create the MySQL tables? >> ser_mysql.sh create (or the *.sql directly) should create them. >> Either way make sure you populate the domain table as well. >> >> Just my $.02, >> Hendrik >> >> -- >> freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany >> Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 >> Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > create the database with -n option (name of the new database) the same db user but new database for other ser instance there are some other usefull options too:)

1 0

[Serusers] ser-0.10.99-dev64 (ser_ctl)
by jean sigue 11 Jan '07

11 Jan '07

Hi, I am trying to create a sip account. When I enter the command [root@itiah runkit-0.9]# serctl add 7777777(a)vaboomz.com mypass ge(a)vaboomz.com MySql password: ERROR 1146 (42S02) at line 1: Table 'ser.subscriber' doesn't exist error: 500 Command 'ul_show_contact' not found Can anyome tell me where to find ser_ctl or more importantly how to fix it .

2 2

RE: [Users] using multiple DNS servers
by Will Quan 11 Jan '07

11 Jan '07

Yes, I think its DNS related because the callflows look perfect when both my DNS servers are up and running. This scenario only occurs when the primary dns server is offline. I thought maybe my debug trace would tell more of the story. Sorry I dont have a ngrep trace. << primary dns is offline; assuming res_search() considers the secondary in resolv.conf >>Jan 10 20:55:20 www openser[21124]: get_record: lookup(5934.rdtw1.qa2.l1.voiplabs.net, 35) failedJan 10 20:55:20 www openser[21124]: DEBUG:sip_resolvehost2: no valid NAPTR record found for 5934.rdtw1.qa2.l1.voiplabs.net, trying direct SRV lookup...Jan 10 20:55:20 www openser[21346]: get_record: lookup(5934.rdtw1.qa2.l1.voiplabs.net, 35) failedJan 10 20:55:20 www openser[21346]: DEBUG:sip_resolvehost2: no valid NAPTR record found for 5934.rdtw1.qa2.l1.voiplabs.net, trying direct SRV lookup...Jan 10 20:55:20 www openser[21140]: is_from_user_enum(): order 100, pref 100, flen 1, flags 'u', slen 7, services 'SIP+E2U', rlen 43, regexp '!^(.*)$!sip:\1@5934.rdtw1.qa2.l1.voiplabs.net!'Jan 10 20:55:20 www openser[21140]: reg_replace(): pattern: '^(.*)$', replacement: 'sip:\1@5934.rdtw1.qa2.l1.voiplabs.net', string: '+14556660002'Jan 10 20:55:20 www openser[21140]: is_from_user_enum(): resulted in replacement: 'sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net'Jan 10 20:55:20 www openser[21140]: DEBUG:sip_resolvehost2: no port, no proto -> do NAPTR lookup!Jan 10 20:55:21 www openser[21233]: get_record: lookup(5934.rdtw1.qa2.l1.voiplabs.net, 35) failedJan 10 20:55:21 www openser[21233]: DEBUG:sip_resolvehost2: no valid NAPTR record found for 5934.rdtw1.qa2.l1.voiplabs.net, trying direct SRV lookup...<<processing 200 OK (invite) for the first time >>Jan 10 20:55:22 www openser[21124]: SIP Reply (status):Jan 10 20:55:22 www openser[21124]: version: <SIP/2.0>Jan 10 20:55:22 www openser[21124]: status: <200>Jan 10 20:55:22 www openser[21124]: reason: <OK>Jan 10 20:55:22 www openser[21124]: parse_headers: flags=2Jan 10 20:55:22 www openser[21124]: Found param type 232, <branch> = <z9hG4bK7d34.d8cbc746.0>; state=16Jan 10 20:55:22 www openser[21124]: end of header reached, state=5Jan 10 20:55:22 www openser[21124]: parse_headers: Via found, flags=2Jan 10 20:55:22 www openser[21124]: parse_headers: this is the first viaJan 10 20:55:22 www openser[21124]: After parse_msg...Jan 10 20:55:22 www openser[21124]: forward_reply: found module tm, passing reply to itJan 10 20:55:22 www openser[21124]: DEBUG: t_check: msg id=19 global id=18 T start=0xffffffffJan 10 20:55:22 www openser[21124]: parse_headers: flags=22Jan 10 20:55:22 www openser[21124]: Found param type 232, <branch> = <z9hG4bK175-64.101.130.210>; state=16Jan 10 20:55:22 www openser[21124]: end of header reached, state=5Jan 10 20:55:22 www openser[21124]: parse_headers: Via found, flags=22Jan 10 20:55:22 www openser[21124]: parse_headers: this is the second viaJan 10 20:55:22 www openser[21124]: DEBUG: add_param: tag=35Jan 10 20:55:22 www openser[21124]: DEBUG:parse_to:end of header reached, state=29Jan 10 20:55:22 www openser[21124]: DBUG:parse_to: display={"18001033799"}, ruri={sip:+18001033799@rdtw1.qa2.l1.voiplabs.net}Jan 10 20:55:22 www openser[21124]: DEBUG: get_hdr_field: <To> [64]; uri=[sip:+18001033799@rdtw1.qa2.l1.voiplabs.net] Jan 10 20:55:22 www openser[21124]: DEBUG: to body ["18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>]Jan 10 20:55:22 www openser[21124]: get_hdr_field: cseq <CSeq>: <1> <INVITE>Jan 10 20:55:22 www openser[21124]: parse_headers: flags=8Jan 10 20:55:22 www openser[21124]: DEBUG: t_reply_matching: hash 17367 label 1685896333 branch 0Jan 10 20:55:22 www openser[21124]: DEBUG:tm:REF_UNSAFE: after is 1Jan 10 20:55:22 www openser[21124]: DEBUG: t_reply_matching: reply matched (T=0x283fe3a8)!Jan 10 20:55:22 www openser[21124]: parse_headers: flags=8Jan 10 20:55:22 www openser[21124]: DBG: trans=0x283fe3a8, callback type 2, id 0 enteredJan 10 20:55:22 www openser[21124]: parse_headers: flags=8Jan 10 20:55:22 www openser[21124]: DEBUG: t_check: msg id=19 global id=19 T end=0x283fe3a8Jan 10 20:55:22 www openser[21124]: DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 is_invite=1)Jan 10 20:55:22 www openser[21124]: DEBUG:tm:t_should_relay_response: T_code=180, new_code=200Jan 10 20:55:22 www openser[21124]: DEBUG:tm:relay_reply: branch=0, save=0, relay=0Jan 10 20:55:22 www openser[21124]: old size: 844, new size: 783Jan 10 20:55:22 www openser[21124]: build_res_from_sip_res: copied size: orig:289, new: 228, rest: 555 msg= SIP/2.0 200 OK^M Record-Route: <sip:10.88.128.76;lr;ftag=64.101.130.210>^M Record-Route: <sip:10.88.128.77;lr;ftag=64.101.130.210>^M Record-Route: <sip:10.88.128.74;lr;ftag=64.101.130.210>^M Record-Route: <sip:64.101.130.210:5062>^M Via: SIP/2.0/UDP 64.101.130.210:5062;branch=z9hG4bK175-64.101.130.210^M From: "14556660002" <sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net>;tag=64.101.130.210^M To: "18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>;tag=35^M Call-ID: 1-1140(a)64.101.130.210^M CSeq: 1 INVITE^M Max-Forwards: 70^M Content-Length: 145^M Contact: "18001033799" <sip:+118001033799@10.88.129.252:6001>^M Content-Type: application/sdp^M ^M v=0^M o=user1 53655765 2353687637 IN IP4 10.88.129.252^M s=SIP Call^M c=IN IP4 10.88.129.252^M t=0 0^M m=audio 10800 RTP/AVP 0^M a=rtpmap:0 PCMU/8000^M Jan 10 20:55:22 www openser[21124]: DEBUG: update_totag_set: new totag Jan 10 20:55:22 www openser[21124]: DEBUG: add_to_tail_of_timer[2]: 0x283fe3f0Jan 10 20:55:22 www openser[21124]: DEBUG:tm:relay_reply: sent buf=0x81351c8: SIP/2.0 2..., shmem=0x283fe090: SIP/2.0 2Jan 10 20:55:22 www openser[21124]: DBG: trans=0x283fe3a8, callback type 128, id 0 enteredJan 10 20:55:22 www openser[21124]: ACC: transaction answered: method=INVITE, i-uri=sip:+18001033799@rdtw1.qa2.l1.voiplabs.net, o-uri=sip:+18001033799@10.88.128.77:5060, call_id=1-1140(a)64.101.130.210, from="14556660002" <sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net>;tag=64.101.130.210, code=200, uid=n/a, fromtag=64.101.130.210, fromuri=sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net, fromuri=sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net, fromuser=+14556660002, fromdomain=5934.rdtw1.qa2.l1.voiplabs.net, to="18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>;tag=35, totag=35, touri=sip:+18001033799@rdtw1.qa2.l1.voiplabs.net, touser=+18001033799, userpart=+18001033799, domain=rdtw1.qa2.l1.voiplabs.netJan 10 20:55:22 www openser[21124]: DEBUG: cleanup_uac_timers: RETR/FR timers resetJan 10 20:55:22 www openser[21124]: DEBUG:tm:UNREF_UNSAFE: after is 0Jan 10 20:55:22 www openser[21124]: DEBUG:destroy_avp_list: destroying list 0x0Jan 10 20:55:22 www openser[21124]: receive_msg: cleaning up<< processing ACK from orig >>Jan 10 20:55:22 www openser[21124]: SIP Request:Jan 10 20:55:22 www openser[21124]: method: <ACK>Jan 10 20:55:22 www openser[21124]: uri: <sip:+18001033799@10.88.129.253>Jan 10 20:55:22 www openser[21124]: version: <SIP/2.0>Jan 10 20:55:22 www openser[21124]: parse_headers: flags=2Jan 10 20:55:22 www openser[21124]: end of header reached, state=5Jan 10 20:55:22 www openser[21124]: parse_headers: Via found, flags=2Jan 10 20:55:22 www openser[21124]: parse_headers: this is the first viaJan 10 20:55:22 www openser[21124]: After parse_msg...Jan 10 20:55:22 www openser[21124]: preparing to run routing scripts...Jan 10 20:55:22 www openser[21124]: DEBUG : sl_filter_ACK: to late to be a local ACK!Jan 10 20:55:22 www openser[21124]: parse_headers: flags=100Jan 10 20:55:22 www openser[21124]: DEBUG: add_param: tag=35Jan 10 20:55:22 www openser[21124]: DEBUG:parse_to:end of header reached, state=29Jan 10 20:55:22 www openser[21124]: DBUG:parse_to: display={"18001033799"}, ruri={sip:+18001033799@rdtw1.qa2.l1.voiplabs.net}Jan 10 20:55:22 www openser[21124]: DEBUG: get_hdr_field: <To> [64]; uri=[sip:+18001033799@rdtw1.qa2.l1.voiplabs.net] Jan 10 20:55:22 www openser[21124]: DEBUG: to body ["18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>]Jan 10 20:55:22 www openser[21124]: get_hdr_field: cseq <Cseq>: <2> <ACK>Jan 10 20:55:22 www openser[21124]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Jan 10 20:55:22 www openser[21124]: parse_headers: flags=200Jan 10 20:55:22 www openser[21124]: is_preloaded: NoJan 10 20:55:22 www openser[21124]: grep_sock_info - checking if host==us: 13==12 && [10.88.129.253] == [10.88.128.74]Jan 10 20:55:22 www openser[21124]: grep_sock_info - checking if port 5060 matches port 5060Jan 10 20:55:22 www openser[21124]: check_self: host != meJan 10 20:55:22 www openser[21124]: grep_sock_info - checking if host==us: 12==12 && [10.88.128.74] == [10.88.128.74]Jan 10 20:55:22 www openser[21124]: grep_sock_info - checking if port 5060 matches port 5060Jan 10 20:55:22 www openser[21124]: after_loose: Topmost route URI: 'sip:10.88.128.74;lr;ftag=64.101.130.210' is meJan 10 20:55:22 www openser[21124]: after_loose: URI to be processed: 'sip:10.88.128.77;lr;ftag=64.101.130.210'Jan 10 20:55:22 www openser[21124]: after_loose: Next URI is a loose routerJan 10 20:55:22 www openser[21124]: DEBUG: t_newtran: msg id=20 , global msg id=19 , T on entrance=0xffffffffJan 10 20:55:22 www openser[21124]: parse_headers: flags=ffffffffffffffffJan 10 20:55:22 www openser[21124]: DEBUG: get_hdr_body : content_length=0Jan 10 20:55:22 www openser[21124]: found end of headerJan 10 20:55:22 www openser[21124]: parse_headers: flags=78Jan 10 20:55:22 www openser[21124]: t_lookup_request: start searching: hash=17364, isACK=1Jan 10 20:55:22 www openser[21124]: DEBUG: proceeding to pre-RFC3261 transaction matchingJan 10 20:55:22 www openser[21124]: DEBUG: t_lookup_request: no transaction foundJan 10 20:55:22 www openser[21124]: DEBUG:tm:t_relay: forwarding ACKJan 10 20:55:22 www openser[21124]: DEBUG: mk_proxy: doing DNS lookup...Jan 10 20:55:22 www openser[21124]: check_via_address(64.101.130.210, 64.101.130.210, 0)Jan 10 20:55:22 www openser[21124]: Sending: ACK sip:+18001033799@10.88.129.253 SIP/2.0^M Via: SIP/2.0/UDP 10.88.128.74;branch=0^M Via: SIP/2.0/UDP 64.101.130.210:5062^M From: "14556660002" <sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net>;tag=64.101.130.210^M To: "18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>;tag=35^M Call-ID: 1-1140(a)64.101.130.210^M Cseq: 2 ACK^M Contact: "14556660002" <sip:14556660002@64.101.130.210>^M Max-Forwards: 69^M Record-Route: <sip:64.101.130.210:5062>^M Route: <sip:10.88.128.77;lr;ftag=64.101.130.210>,<sip:10.88.128.76;lr;ftag=64.101.130.210>,"18001033799" <sip:+118001033799@10.88.129.252:6001>^M Content-Length: 0^M ^M .Jan 10 20:55:22 www openser[21124]: orig. len=608, new_len=605, proto=1Jan 10 20:55:22 www openser[21124]: DEBUG:destroy_avp_list: destroying list 0x0Jan 10 20:55:22 www openser[21124]: receive_msg: cleaning up<< strange behavior starts here ... >>Jan 10 20:55:22 www openser[21346]: DEBUG:do_srv_lookup: SRV(_sip._udp.5934.rdtw1.qa2.l1.voiplabs.net) = proxy.5934.rdtw1.qa2.l1.voiplabs.net:5060Jan 10 20:55:22 www openser[21346]: rdtw1.qa2.l1.voiplabs.net-outbound: time_t=1168484116 ^LINE=180 ^FILE=/var/local/ser/ROUTLET/config.pre ^call_id=1-1140(a)64.101.130.210 ^cseq=1 ^contact="14556660002" <sip:+14556660002@64.101.130.210> ^from=sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net ^fromtag=64.101.130.210 ^to=sip:+18001033799@rdtw1.qa2.l1.voiplabs.net ^totag=<null> ^method=INVITE ^ruri=sip:+18001033799@rdtw1.qa2.l1.voiplabs.net ^messageid=8^remark=sJan 10 20:55:22 www openser[21346]: rdtw1.qa2.l1.voiplabs.net-RECORDROUTE: time_t=1168484116 ^LINE=180 ^FILE=/var/local/ser/ROUTLET/config.pre ^call_id=1-1140(a)64.101.130.210 ^cseq=1 ^contact="14556660002" <sip:+14556660002@64.101.130.210> ^from=sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net ^fromtag=64.101.130.210 ^to=sip:+18001033799@rdtw1.qa2.l1.voiplabs.net ^totag=<null> ^method=INVITE ^ruri=sip:+18001033799@10.88.128.77:5060 ^messageid=8^remark=rdtw1.qa2.l1.voiplabs.netJan 10 20:55:22 www openser[21346]: DEBUG: t_newtran: msg id=8 , global msg id=7 , T on entrance=0xffffffffJan 10 20:55:22 www openser[21346]: parse_headers: flags=ffffffffffffffffJan 10 20:55:22 www openser[21346]: parse_headers: flags=78Jan 10 20:55:22 www openser[21346]: t_lookup_request: start searching: hash=17367, isACK=0Jan 10 20:55:22 www openser[21346]: DEBUG: RFC3261 transaction matched, tid=175-64.101.130.210Jan 10 20:55:22 www openser[21346]: DEBUG:tm:REF_UNSAFE: after is 1Jan 10 20:55:22 www openser[21346]: DEBUG: t_lookup_request: transaction found (T=0x283fe3a8)<< about to reTX the 200 OK (invite) *even though we just finished processing ACK* >>Jan 10 20:55:22 www openser[21346]: DEBUG:tm:t_retransmit_reply: buf=0x28229b40: SIP/2.0 2..., shmem=0x283fe090: SIP/2.0 2 ! ! ! ! ! ! ! !!!!!!!!Jan 10 20:55:22 www openser[21346]: DEBUG:tm:UNREF_UNSAFE: after is 0Jan 10 20:55:22 www openser[21346]: DEBUG:destroy_avp_list: destroying list 0x0Jan 10 20:55:22 www openser[21346]: receive_msg: cleaning up<< orig sends another ACK; to be expected since we just reTX the 200 OK >>Jan 10 20:55:22 www openser[21124]: SIP Request:Jan 10 20:55:22 www openser[21124]: method: <ACK>Jan 10 20:55:22 www openser[21124]: uri: <sip:+18001033799@10.88.129.253>Jan 10 20:55:22 www openser[21124]: version: <SIP/2.0>Jan 10 20:55:22 www openser[21124]: parse_headers: flags=2Jan 10 20:55:22 www openser[21124]: end of header reached, state=5Jan 10 20:55:22 www openser[21124]: parse_headers: Via found, flags=2Jan 10 20:55:22 www openser[21124]: parse_headers: this is the first viaJan 10 20:55:22 www openser[21124]: After parse_msg...Jan 10 20:55:22 www openser[21124]: preparing to run routing scripts...Jan 10 20:55:22 www openser[21124]: DEBUG : sl_filter_ACK: to late to be a local ACK!Jan 10 20:55:22 www openser[21124]: parse_headers: flags=100Jan 10 20:55:22 www openser[21124]: DEBUG: add_param: tag=35Jan 10 20:55:22 www openser[21124]: DEBUG:parse_to:end of header reached, state=29Jan 10 20:55:22 www openser[21124]: DBUG:parse_to: display={"18001033799"}, ruri={sip:+18001033799@rdtw1.qa2.l1.voiplabs.net}Jan 10 20:55:22 www openser[21124]: DEBUG: get_hdr_field: <To> [64]; uri=[sip:+18001033799@rdtw1.qa2.l1.voiplabs.net] Jan 10 20:55:22 www openser[21124]: DEBUG: to body ["18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>]Jan 10 20:55:22 www openser[21124]: get_hdr_field: cseq <Cseq>: <2> <ACK>Jan 10 20:55:22 www openser[21124]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Jan 10 20:55:22 www openser[21124]: parse_headers: flags=200Jan 10 20:55:22 www openser[21124]: is_preloaded: NoJan 10 20:55:22 www openser[21124]: grep_sock_info - checking if host==us: 13==12 && [10.88.129.253] == [10.88.128.74]Jan 10 20:55:22 www openser[21124]: grep_sock_info - checking if port 5060 matches port 5060Jan 10 20:55:22 www openser[21124]: check_self: host != meJan 10 20:55:22 www openser[21124]: grep_sock_info - checking if host==us: 12==12 && [10.88.128.74] == [10.88.128.74]Jan 10 20:55:22 www openser[21124]: grep_sock_info - checking if port 5060 matches port 5060Jan 10 20:55:22 www openser[21124]: after_loose: Topmost route URI: 'sip:10.88.128.74;lr;ftag=64.101.130.210' is meJan 10 20:55:22 www openser[21124]: after_loose: URI to be processed: 'sip:10.88.128.77;lr;ftag=64.101.130.210'Jan 10 20:55:22 www openser[21124]: after_loose: Next URI is a loose routerJan 10 20:55:22 www openser[21124]: DEBUG: t_newtran: msg id=21 , global msg id=20 , T on entrance=0xffffffffJan 10 20:55:22 www openser[21124]: parse_headers: flags=ffffffffffffffffJan 10 20:55:22 www openser[21124]: DEBUG: get_hdr_body : content_length=0Jan 10 20:55:22 www openser[21124]: found end of headerJan 10 20:55:22 www openser[21124]: parse_headers: flags=78Jan 10 20:55:22 www openser[21124]: t_lookup_request: start searching: hash=17364, isACK=1Jan 10 20:55:22 www openser[21124]: DEBUG: proceeding to pre-RFC3261 transaction matchingJan 10 20:55:22 www openser[21124]: DEBUG: t_lookup_request: no transaction foundJan 10 20:55:22 www openser[21124]: DEBUG:tm:t_relay: forwarding ACKJan 10 20:55:22 www openser[21124]: DEBUG: mk_proxy: doing DNS lookup...Jan 10 20:55:22 www openser[21124]: check_via_address(64.101.130.210, 64.101.130.210, 0)Jan 10 20:55:22 www openser[21124]: Sending: ACK sip:+18001033799@10.88.129.253 SIP/2.0^M Via: SIP/2.0/UDP 10.88.128.74;branch=0^M Via: SIP/2.0/UDP 64.101.130.210:5062^M From: "14556660002" <sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net>;tag=64.101.130.210^M To: "18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>;tag=35^M Call-ID: 1-1140(a)64.101.130.210^M Cseq: 2 ACK^M Contact: "14556660002" <sip:14556660002@64.101.130.210>^M Max-Forwards: 69^M Record-Route: <sip:64.101.130.210:5062>^M Route: <sip:10.88.128.77;lr;ftag=64.101.130.210>,<sip:10.88.128.76;lr;ftag=64.101.130.210>,"18001033799" <sip:+118001033799@10.88.129.252:6001>^M Content-Length: 0^M ^M .Jan 10 20:55:22 www openser[21124]: orig. len=608, new_len=605, proto=1Jan 10 20:55:22 www openser[21124]: DEBUG:destroy_avp_list: destroying list 0x0Jan 10 20:55:22 www openser[21124]: receive_msg: cleaning upJan 10 20:55:23 www openser[21376]: DEBUG: timer routine:1,tl=0x283fe4d4 next=0x0Jan 10 20:55:23 www openser[21376]: DEBUG: timer routine:4,tl=0x283fe4c4 next=0x0<< repeat reTX 200 OK (invite) / ACK several times >>Any Ideas?Will> Date: Thu, 11 Jan 2007 17:23:10 +0100> From: klaus.mailinglists(a)pernau.at> To: sip(a)voiplabs.net> CC: users(a)openser.org> Subject: Re: [Users] using multiple DNS servers> > Will Quan wrote:> > I have a primary and secondary dns server setup in resolv.conf, with> > a timeout option of 1 second.The idea is when the primary fails, my> > secondary server picks up the workload.When openser is running, and> > primary dns is down, I am seeing several re-transmissions of 200 OK> > (invite) to my originator *after* openser has finished processing> > ACK. I was thinking the final responses would stop reTX after the ACK> > was t_relayed. Have you seen such behavior? And better yet- Is there> > a solution for this?Thanks, Will> > Are you sure this problem is related to DNS failover?> > regards> klaus> > > > > > > ------------------------------------------------------------------------> > > > > > _______________________________________________ Users mailing list > > Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users> > > -- > Klaus Darilion> nic.at>

1 0

[Users] NEW OPTION: "auto_aliases" core parameter
by Bogdan-Andrei Iancu 11 Jan '07

11 Jan '07

Hi everybody, I added a nee core parameter named "auto_aliases" - it controls if aliases should be automatically discovered and added during fixing listening sockets. Formally, this was done with no control over it (to turn it off), which was a security risk (in my opinion) as the server was using auto-discover aliases that were not configured. To maintain backward compatibility, the default value is "on"; Ex: auto_aliases=no or auto_aliases=0 regards, bogdan

2 4

[Users] msg_exec program in background
by Dimo 11 Jan '07

11 Jan '07

Hi, I am having trouble with msg_exec. I want it to start the execution of a command which may wait for some time but i do not need openser to wait for the command to end in order to proceed with script execution. I tried this: msg_exec("command &"); and this makes openser process hang forever with a defunct child. Is there any way someone may suggest that i can make openser run a command and not wait for it to end? nohup does not do the trick also. Best, Dimo

2 2

[Users] presence module database tables
by Cristian Caprar 11 Jan '07

11 Jan '07

Hi, I am having trouble creating the xcap_xml table of the new presence module in mysql (latest 4.1.22 on RHEL 4). The table definition from the docs is: CREATE TABLE `xcap_xml` ( `user` varchar(256) NOT NULL, `domain` varchar(256) NOT NULL, `xcap` text NOT NULL, `doc_type` varchar(64) NOT NULL, PRIMARY KEY (`user`,`domain`,`doc_type`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1; I get the following error: ERROR 1170 (42000): BLOB/TEXT column 'user' used in key specification without a key length In my opinion, the problem is the definition of the user and domain columns, which are varchar(256), instead of varchar(255) and are automatically promoted to text columns. If changing the size to 255, everything is fine. Does the presence db code do some length checks that can be affected by this change? Is the table definition correct or am I missing something? Thanks, Cristian

2 2

[Serusers] New version of SER with presence support
by raj gopal 11 Jan '07

11 Jan '07

Hi, When the new version of stable SER with complete presence support is planned for release???? Venkat

3 5

[Users] How can i add my own events on KphoneSI
by Kc Ayuba 11 Jan '07

11 Jan '07

Hi, I'm using the latest openser version with presence supports. My User Agent=kphoneSI. My question is: How can i add my own events on the KphoneSI? Kind regards, Kc

3 4

[Users] Multi Call Legs Accounting
by Howard Tang 11 Jan '07

11 Jan '07

Hi, Did anyone successfully setup call accounting for call forward? I have been searching for a working example for few days on google, I still couldn't find any working example. The config I have at the moment is modparam("acc", "multi_leg_enabled", 1) modparam("acc", "src_leg_avp_id", 110) modparam("acc", "dst_leg_avp_id", 111) In the main route, if (!method=="REGISTER" && !method=="NOTIFY") { avp_write("$from/username", "$avp(i:110)"); avp_write("$ruri/username", "$avp(i:111)"); }; Let say A call B and B forward call to C, the record in the acc table is all A to C without B in any one of the record for billing purpose. The Call forward was set on the UA. Can anyone give me a hand on this please? -- Howard Tang

2 4

[Users] Updating call-info...
by Premnath, KN 11 Jan '07

11 Jan '07

Hello, How to update call-info in openser (for an INVITE sip message)? I want the call-info to contain the actual IP-address of called party. (that we normally do for $ruri). Thank You. With warm regards, Premnath, KN The new company Nokia Siemens Networks is expected to start operations in the first quarter of 2007, subject to customary regulatory approvals, the completion of standard closing conditions, and the agreement of a number of detailed implementation steps. Disclaimer: This e-mail may contain trade secrets or privileged, undisclosed or otherwise confidential information. If you have received this e-mail in error, you are hereby notified that any review, copying or distribution of it is strictly prohibited. Please inform us immediately and destroy the original transmittal. Thank you for your cooperation.

2 5

[Serusers] No Sound when use LCR
by Arun Kumar 11 Jan '07

11 Jan '07

Hi, I've configured my SER to send calls to my Asterisk server when user connects to ser and dial a number ser forwards this to my Asterisk server and my asterisk server ask for my pin number saying "Please enter your pin number". In this config I'm using rewritehostport(ip:port) to forward the calls and its working fine. But, Now I've implemented the LCR when I see ngrep calls are going to my asterisk server but I'm not able to hear any sound from my asterisk server saying "enter your pin number". Please help in this issue. Here is my ser.cfg file : debug=3 fork=no log_stderror=yes listen=192.168.2.210 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:arun@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/lcr.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", " mysql://ser:arun@localhost/ser") 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) modparam("tm", "fr_inv_timer", 120) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("lcr","db_url", "mysql://ser:arun@localhost/ser") modparam("lcr","gw_table","gw") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","grp_id_column","grp_id") modparam("lcr","lcr_table","lcr") modparam("lcr","prefix_column","prefix") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","contact_avp","1401") modparam("lcr","fr_inv_timer_avp","fr_inv_timer_avp") modparam("lcr","fr_inv_timer",90) modparam("lcr","fr_inv_timer_next",60) # ------ Account and Radius modules-------------- modparam("acc", "db_url", "mysql://ser:arun@localhost/ser") modparam("acc","log_level",1) modparam("acc","log_flag",1) modparam("acc", "db_flag", 1) modparam("acc","radius_flag",1) modparam("acc","service_type",15) #modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; if (!load_gws()) { sl_send_reply("500", "Server Internal Error - Cannot load gateways"); break; } else { xlog("L_INFO", "\n[SER]: - [$Tf] Method: <$rm> R-uri: <$ru> From: <$fu>To <$tu> : Successfully Loaded Gateway List\n"); }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.168.2.210:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden Cant cross 1st bridge"); break; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { sl_send_reply("404", "I am trusted host, "); route(4); route(1); } else { sl_send_reply("403", "Forbidden, I am here"); }; break; }; if (method=="ACK") { setflag(1); route(1); break; } if (method=="CANCEL") { setflag(1); route(1); break; } else if (method=="INVITE") { setflag(1); route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { setflag(1); 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"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (!is_uri_host_local()) { route(4); route(1); break; }; if (!next_gw()) { t_reply("503", "Service not available - No more gateways"); break; } else { t_on_failure("2"); t_relay(); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } #route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- # rewritehostport("192.168.2.202:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS # avp_write("i:45", "inv_timeout"); # route(1); #} onreply_route[1] { xlog("L_INFO", "In onreply route"); 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(); }; } failure_route[2] { if (!next_gw()) { t_reply("503", "Service not available - No more gateways"); break; } else { t_on_failure("2"); t_relay(); break; } } thanks arun

2 2

[Serusers] RTPProxy and SER, Multiple Interface Server
by Gerardo Amaya 11 Jan '07

11 Jan '07

Hello All. After investigating quite a while about this issue I'm kind of desperate now, I will appreciate your help on this. I have a SER Server with multiple interfaces, I'm able to handle all the SIP packets with the multi homed enabled. Now the part that I'm having real trouble with is the RTP audio streams. After struggling with MediaProxy I finally started using RTPProxy for this, since it has bridge mode available. Now the problem is that I have more than one network to connect audio from, so the internal and external configuration is not enough for me. So my questions are: 1.-Can I have RTPProxy Bridge mode working for all the interfaces in my server? 2.-Can I have more than one RTPProxy socket available for each of the multiple interfaces I'm connecting in order to have the audio working, so each route will use the proxy of their own network? 3.-If #2 is true, is this where the force_socket and force_rtp_proxy functions play a role in my configuration? Please help!! Thanks you so much in advance Gerardo Amaya

2 3

[Serusers] 0.10.99 - ERROR: authorize.c:359: Cannot obtain digest realm from parameter '$fd.digest_realm'
by "tzieleniewski" 11 Jan '07

11 Jan '07

HI All! I have encountered the following error and don't know waht to do with it:) ERROR: authorize.c:359: Cannot obtain digest realm from parameter '$fd.digest_realm' the situation takes places when the following sip massage arives at ser: # U 2007/01/09 11:43:02.274949 127.0.0.1:7000 -> 127.0.0.1:5060 REGISTER sip:scscf.ims.touk.pl SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:7000;rport;branch=z9hG4bK2575034C94BE793615C5BEE814464E5B. From: tzl <sip:tzl@scscf.ims.touk.pl>;tag=1829574745. To: tzl <sip:tzl@scscf.ims.touk.pl>. Contact: "tzl" <sip:tzl@127.0.0.1:7000>. Call-ID: 3E67EAEFDC8BFC73844E4D0F849DF676(a)scscf.ims.touk.pl. CSeq: 38402 REGISTER. Expires: 1800. Max-Forwards: 70. User-Agent: X-Lite release 1105d. Content-Length: 0. . My Register logic looks like this: route[REGISTER] { if (method=="REGISTER") { if (!$t.did) { sl_send_reply("403", "Register forwarding forbidden"); drop; } if (!www_authenticate("$fd.digest_realm", "credentials")) { if ($? == -2) { sl_send_reply("500", "Internal Server Error"); } else if ($? == -3) { sl_send_reply("400", "Bad Request"); } else { if ($digest_challenge) { append_to_reply("%$digest_challenge"); } sl_send_reply("401", "Unauthorized"); } drop; }; } and here are my database contensts: mysql> select * from uri; +------------------------------+--------------------------+----------+-------+--------+ | uid | did | username | flags | scheme | +------------------------------+--------------------------+----------+-------+--------+ | tzl(a)sipproxy.hellboy.touk.pl | sipproxy.hellboy.touk.pl | tzl | 61 | sip | +------------------------------+--------------------------+----------+-------+--------+ mysql> select * from domain; +--------------------------+--------------------------+-------+ | did | domain | flags | +--------------------------+--------------------------+-------+ | sipproxy.hellboy.touk.pl | sipproxy.hellboy.touk.pl | 37 | +--------------------------+--------------------------+-------+ mysql> select * from domain_attrs; +--------------------------+--------------+------+--------------------------+-------+ | did | name | type | value | flags | +--------------------------+--------------+------+--------------------------+-------+ | sipproxy.hellboy.touk.pl | sw_admin | 2 | 0 | 32 | | sipproxy.hellboy.touk.pl | digest_realm | 2 | sipproxy.hellboy.touk.pl | 32 | +--------------------------+--------------+------+--------------------------+-------+ mysql> select * from credentials; +---------------+--------------------------+--------------------------+----------+-------+----------------------------------+----------------------------------+------------------------------+ | auth_username | did | realm | password | flags | ha1 | ha1b | uid | +---------------+--------------------------+--------------------------+----------+-------+----------------------------------+----------------------------------+------------------------------+ | tzl | sipproxy.hellboy.touk.pl | sipproxy.hellboy.touk.pl | tzl | 33 | df4a49c51b082f68b792d7e2177c6ed8 | 8ba18e21048f2f587ad3386b54b99e5e | tzl(a)sipproxy.hellboy.touk.pl | | admin | sipproxy.hellboy.touk.pl | sipproxy.hellboy.touk.pl | heslo | 32 | | | 0 | +---------------+--------------------------+--------------------------+----------+-------+----------------------------------+----------------------------------+------------------------------+ Thanku for any help Bests Tomasz

6 8

[Serusers] xlite and ser
by zahra bahar 11 Jan '07

11 Jan '07

my configure in xlite:in system setting in network I put these: Auto detect IP:yes Primary stun server:blank in sip proxy I selected default and: enabled:yes display name:user username:1234 Aotorisation user:1234 password:1234 Domain/Realm:10.10.10.138 sip proxy:10.10.10.138:5060 use outband proxy:default send internal ip:Always register:default Is it any wrong?this xlite phone is on a pc that ser server is.another one is on another PC and only in username and display and password differs with this one. I am in the back of a NAT but my two clients and server are there. what is so important in setting xlite? thanks. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com --------------------------------- Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.

2 1

[Serusers] documentation generation - ser 0.10.99
by TZieleniewski 10 Jan '07

10 Jan '07

Hi, Is there some method to generate modules documentation from the cvs source?? Im running make install-modules-doc and nothing happens?? Bests Tomasz

3 3

[Serusers] Routing Based on Codec
by Nathan E. Pralle 10 Jan '07

10 Jan '07

Has anyone ever tried routing calls based on codec in SER? If so, do you have pointers for getting started? Thanks, Nathan -- -------------------- www.nathanpralle.com --------------------

1 0

[Users] Logging levels
by Cristian Caprar 10 Jan '07

10 Jan '07

Can anyone explain the link between the debug core parameter and the logging levels used in the log functions (log and xlog)? The levels start from -3 (ALERT) and go up to 5 (DEBUG), but the parameter value is from 0 to 9. Thanks, Cristian

2 3

[Users] Multiple networks source address
by T.R. Missner 10 Jan '07

10 Jan '07

Hello, I ran into a seemingly strange issue using openser to proxy an invite from the public internet to a private ipsec tunnel on my network. The problem appears to be caused by the fact that the packet is sent out to on the correct interface ( eth1 in my case ) but still is using the source address of et0 in the ip header. This is causing my ipsec tunnel to ignore the packet. Is there anyway to control the source addr in the IP header? Using a simple python script to send a upd packet across the same path I have verified this is not a tunnel issue. Example et0: 4.X.X.229 packet arrives on this interface eth1: 10.X.X.13 packet is sent out this interface RURI is re-written and t_relay is used to send the packet to destination: 10.X.X.115 Here is a partially redacted packet as captured by tethereal: Frame 1 (982 bytes on wire, 982 bytes captured) Arrival Time: Jan 2, 2007 16:49:35.536412000 [Time delta from previous packet: 0.000000000 seconds] [Time since reference or first frame: 0.000000000 seconds] Frame Number: 1 Packet Length: 982 bytes Capture Length: 982 bytes [Frame is marked: False] [Protocols in frame: eth:ip:udp:sip:sdp] Ethernet II, Src: DellComp_f6:27:19 (00:06:5b:f6:27:19), Dst: DellComp_ec:8d:a3 (00:06:5b:ec:8d:a3) Destination: DellComp_ec:8d:a3 (00:06:5b:ec:8d:a3) Address: DellComp_ec:8d:a3 (00:06:5b:ec:8d:a3) .... ...0 .... .... .... .... = IG bit: Individual address (unicast) .... ..0. .... .... .... .... = LG bit: Globally unique address (factory default) Source: DellComp_f6:27:19 (00:06:5b:f6:27:19) Address: DellComp_f6:27:19 (00:06:5b:f6:27:19) .... ...0 .... .... .... .... = IG bit: Individual address (unicast) .... ..0. .... .... .... .... = LG bit: Globally unique address (factory default) Type: IP (0x0800) Internet Protocol, Src: 4.X.X.229 (4.X.X.229), Dst: 10.X.X.108 (10.X.X.108) ------> should source from 10.X.X.13 not 4.X.X.229 Version: 4 Header length: 20 bytes Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00) 0001 00.. = Differentiated Services Codepoint: Unknown (0x04) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 968 Identification: 0x0076 (118) Flags: 0x04 (Don't Fragment) 0... = Reserved bit: Not set .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x1dfe [correct] [Good: True] [Bad : False] Source: 4.X.X.229 (4.X.X.229) Destination: 10.X.X.108 (10.X.X.108) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 6060 (6060) Source port: 5060 (5060) Destination port: 6060 (6060) Length: 948 Checksum: 0x1c67 [incorrect, should be 0x222a] Session Initiation Protocol Request-Line: INVITE sip:+1720560XXXX@10.X.X.108:6060 SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: <sip:4.X.X.229;lr;ftag=as0473b81f> Via: SIP/2.0/UDP 4.X.X.229;branch=z9hG4bK7a42.e0b9bac5.0 Transport: UDP Sent-by Address: 4.X.X.229 ..... Any clues on how to fix this problem? Thanks, T.R.

4 4

Re: [Serusers] Please help with LCR
by Steve Blair 10 Jan '07

10 Jan '07

Great! The only other advice I'd give you is to think about the gateway list, from matching and prefix. I ran into a few odd cases after deploying LCR where I had to define additional rules to handle URIs with slightly different syntax. -Steve Arun Kumar wrote: > Hi Steve, > > thanks very much your code is working fine now ser is able to use my > both gw and able to send calls on both gw if first fail then it will > send to other. > > thanks very much for helping me. > > thanks > arun > > On 1/10/07, *Steve Blair* <blairs(a)isc.upenn.edu > <mailto:blairs@isc.upenn.edu>> wrote: > > > Arun: > > This depends a little on your configuration. Generally speaking the > !load_gws and !next_gw would be in route block 0. You will most likely > want to do any per-call checks or rewrites before determining which > gateway should receive the call. The failure_route can be inserted > after > your route blocks. > > -Steve > > Arun Kumar wrote: > > Hi, > > > > thanks, One more thing can you please guide me where do I've to > place > > this code in my ser.cfg. > > > > thanks > > > > arun > > > > On 1/10/07, *Steve Blair * <blairs(a)isc.upenn.edu > <mailto:blairs@isc.upenn.edu> > > <mailto: blairs(a)isc.upenn.edu <mailto:blairs@isc.upenn.edu>>> wrote: > > > > > > Arun: > > > > I would try something like the following: > > > > if (!load_gws()) { > > sl_send_reply("500", "Server Internal Error - > Cannot load > > gateways"); > > break; > > } else { > > xlog("L_INFO", "\n[SER]: - [$Tf] Method: <$rm> > R-uri: <$ru> > > From: <$fu> > > To <$tu> : Successfully Loaded Gateway List\n"); > > }; > > > > ...... > > > > if (!next_gw()) { > > t_reply("503", "Service not available - No more > gateways"); > > break; > > } else { > > t_on_failure("2"); <--- This will be where you > load the > > next > > gateway > > t_relay(); > > break; > > }; > > > > failure_route[2] { > > > > if (!next_gw()) { > > xlog("L_CRIT","No more gateways for <$tu>\n"); > > t_reply("503", "Service not available - No more > gateways"); > > break; > > } else { > > t_on_failure("2"); > > t_relay(); > > break; > > } > > } > > > > -Steve > > > > Arun Kumar wrote: > > > Hi > > > > > > thanks for your prompt reply but I've already checked and I'm > > not able > > > to make my lcr working by seeing these two pages I got > this conf but > > > > > > This config is sending the call to my first gateway but > when My > > first > > > gateway is down its not sending the calls to other > gateways. Plese > > > help me to make this working. Here is my config file. > > > > > > ------------------------ > > > debug=3 > > > fork=no > > > log_stderror=yes > > > > > > listen= 192.168.2.210 <http://192.168.2.210> > <http://192.168.2.210> > > <http://192.168.2.210/> # INSERT YOUR IP > > > ADDRESS HERE > > > port=5060 > > > children=4 > > > > > > dns=no > > > rev_dns=no > > > fifo="/tmp/ser_fifo" > > > fifo_db_url="mysql://ser:pass@localhost/ser" > > > > > > loadmodule "/usr/local/lib/ser/modules > > > /mysql.so" > > > loadmodule "/usr/local/lib/ser/modules/sl.so" > > > loadmodule "/usr/local/lib/ser/modules/tm.so" > > > loadmodule "/usr/local/lib/ser/modules/rr.so" > > > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > > > loadmodule "/usr/local/lib/ser/modules/usrloc.so" > > > loadmodule "/usr/local/lib/ser/modules/registrar.so" > > > loadmodule "/usr/local/lib/ser/modules/auth.so" > > > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > > loadmodule "/usr/local/lib/ser/modules/uri.so" > > > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > > > loadmodule "/usr/local/lib/ser/modules/domain.so" > > > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > > > loadmodule "/usr/local/lib/ser/modules/avpops.so" > > > loadmodule "/usr/local/lib/ser/modules/textops.so" > > > loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" > > > loadmodule "/usr/local/lib/ser/modules/permissions.so" > > > loadmodule "/usr/local/lib/ser/modules/acc.so" > > > loadmodule "/usr/local/lib/ser/modules/auth_radius.so" > > > loadmodule "/usr/local/lib/ser/modules/lcr.so" > > > > > > > > > modparam("auth_db|permissions|uri_db|usrloc|domain", > "db_url", " > > > mysql://ser:pass@localhost/ser") > > > 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) > > > > > > modparam("tm", "fr_inv_timer", 120) > > > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > > > > > > modparam("permissions", "db_mode", 1) > > > modparam("permissions", "trusted_table", "trusted") > > > > > > modparam("lcr","db_url", "mysql://ser:pass@localhost/ser") > > > modparam("lcr","gw_table","gw") > > > modparam("lcr","gw_name_column","gw_name") > > > modparam("lcr","ip_addr_column","ip_addr") > > > modparam("lcr","port_column","port") > > > modparam("lcr","grp_id_column","grp_id") > > > modparam("lcr","lcr_table","lcr") > > > modparam("lcr","prefix_column","prefix") > > > modparam("lcr","from_uri_column","from_uri") > > > modparam("lcr","priority_column","priority") > > > modparam("lcr","contact_avp","1401") > > > modparam("lcr","fr_inv_timer_avp","fr_inv_timer_avp") > > > modparam("lcr","fr_inv_timer",90) > > > modparam("lcr","fr_inv_timer_next",60) > > > # ------ Account and Radius modules-------------- > > > modparam("acc", "db_url", " mysql://ser:pass@localhost/ser") > > > > > > modparam("acc","log_level",1) > > > modparam("acc","log_flag",1) > > > modparam("acc", "db_flag", 1) > > > > > > modparam("acc","radius_flag",1) > > > modparam("acc","service_type",15) > > > > > > #modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") > > > > > > > > > > route { > > > > > > # > > > ----------------------------------------------------------------- > > > # Sanity Check Section > > > # > > > ----------------------------------------------------------------- > > > if (!mf_process_maxfwd_header("10")) { > > > sl_send_reply("483", "Too Many Hops"); > > > break; > > > }; > > > > > > if (msg:len > max_len) { > > > sl_send_reply("513", "Message Overflow"); > > > break; > > > }; > > > > > > # > > > ----------------------------------------------------------------- > > > # Record Route Section > > > # > > > ----------------------------------------------------------------- > > > if (method=="INVITE" && client_nat_test("3")) { # > setflag(1); > > > # INSERT YOUR IP ADDRESS HERE > > > record_route_preset(" 192.168.2.210:5060 > <http://192.168.2.210:5060> > > <http://192.168.2.210:5060> > > > <http://192.168.2.210:5060/>;nat=yes"); > > > } else if (method!="REGISTER") { > > > record_route(); > > > }; > > > # > > > ----------------------------------------------------------------- > > > # Call Tear Down Section > > > # > > > ----------------------------------------------------------------- > > > if (method=="BYE" || method=="CANCEL") { setflag(1); > > > end_media_session(); > > > }; > > > > > > # > > > ----------------------------------------------------------------- > > > # Loose Route Section > > > # > > > ----------------------------------------------------------------- > > > if (loose_route()) { > > > > > > if ((method=="INVITE" || method=="REFER") && > !has_totag()) { > > > sl_send_reply("403", "Forbidden Cant cross 1st > bridge"); > > > break; > > > }; > > > > > > if (method=="INVITE") { > > > > > > if (!allow_trusted()) { > > > > > > if (!proxy_authorize("","subscriber")) { > > > proxy_challenge("","0"); > > > break; > > > } else if (!check_from()) { > > > sl_send_reply("403", "Use From=ID"); > > > break; > > > }; > > > > > > consume_credentials(); > > > }; > > > > > > if > (client_nat_test("3")||search("^Route:.*;nat=yes")){ > > > setflag(6); > > > use_media_proxy(); > > > }; > > > }; > > > > > > route(1); > > > break; > > > }; > > > # > > > ----------------------------------------------------------------- > > > # Call Type Processing Section > > > # > > > ----------------------------------------------------------------- > > > if (!is_uri_host_local()) { > > > if (is_from_local() || allow_trusted()) { > > > sl_send_reply("404", "I am trusted host, "); > > > route(4); > > > route(1); > > > } else { > > > sl_send_reply("403", "Forbidden, I am here"); > > > }; > > > break; > > > }; > > > > > > if (method=="ACK") { setflag(1); > > > route(1); > > > break; > > > } if (method=="CANCEL") { setflag(1); > > > route(1); > > > break; > > > } else if (method=="INVITE") { setflag(1); > > > # sl_send_reply("410", "I am sending your call, "); > > > route(3); > > > break; > > > } else if (method=="REGISTER") { > > > route(2); > > > break; > > > }; > > > > > > lookup("aliases"); > > > if (!is_uri_host_local()) { > > > route(4); > > > route(1); > > > break; > > > }; > > > > > > if (!lookup("location")) { > > > sl_send_reply("404", "User Not Found"); > > > break; > > > }; > > > > > > route(1); > > > } > > > > > > route[1] { > > > > > > # > > > ----------------------------------------------------------------- > > > # Default Message Handler > > > # > > > ----------------------------------------------------------------- > > > > > > t_on_reply("1"); > > > > > > if (!t_relay()) { > > > > > > if (method=="INVITE" || method=="ACK") { setflag(1); > > > 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"); > > > break; > > > }; > > > if (!check_to()) { > > > sl_send_reply("401", "Unauthorized"); > > > break; > > > }; > > > > > > consume_credentials(); > > > > > > if (!save("location")) { > > > sl_reply_error(); > > > }; > > > } > > > > > > route[3] { > > > > > > # > > > ----------------------------------------------------------------- > > > # INVITE Message Handler > > > # > > > ----------------------------------------------------------------- > > > > > > if (client_nat_test("3")) { > > > setflag(7); > > > force_rport(); > > > fix_nated_contact(); > > > }; > > > > > > if (!allow_trusted()) { > > > > > > if (!proxy_authorize("","subscriber")) { > > > proxy_challenge("","0"); > > > break; > > > } else if (!check_from()) { > > > sl_send_reply("403", "Use From=ID"); > > > break; > > > }; > > > > > > consume_credentials(); > > > }; > > > > > > route(4); > > > > > > if(!load_gws()){ > > > sl_send_reply("500", "Internal server error, unable to load > > gateways"); > > > break; > > > } > > > if(!next_gw()){ > > > sl_send_reply("503", "Service not available, no > gateways found"); > > > break; > > > }; > > > > > > if (!is_uri_host_local()) { > > > route(4); > > > route(1); > > > break; > > > }; > > > > > > > > > # route(1); > > > } > > > > > > route[4] { > > > > > > # > > > ----------------------------------------------------------------- > > > # NAT Traversal Section > > > # > > > ----------------------------------------------------------------- > > > > > > if (isflagset(6) || isflagset(7)) { > > > if (!isflagset(8)) { > > > setflag(8); > > > use_media_proxy(); > > > }; > > > }; > > > } > > > > > > 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(); > > > }; > > > } > > > > > > > > > thanks very much > > > > > > arun > > > > > > On 1/10/07, * Greger V. Teigre* <greger(a)teigre.com > <mailto:greger@teigre.com> > > <mailto: greger(a)teigre.com <mailto:greger@teigre.com>> > > > <mailto: greger(a)teigre.com <mailto:greger@teigre.com> > <mailto:greger@teigre.com <mailto:greger@teigre.com>>>> wrote: > > > > > > A quick search on iptel.org <http://iptel.org> > <http://iptel.org> > > <http://iptel.org> would have given > > > you these: > > > http://www.iptel.org/lcr_module_drop_in_for_0_9_x > > > http://www.iptel.org/lcr_with_mysql_accounting > <http://www.iptel.org/lcr_with_mysql_accounting> > > <http://www.iptel.org/lcr_with_mysql_accounting> > > > > > > > > > Arun Kumar wrote: > > >> Hi > > >> > > >> please help with LCR can some one show his lcr config > file > > >> please. I'm not able to make it working. > > >> > > >> please help > > >> > > >> thanks > > >> arun > > >> > > > ------------------------------------------------------------------------ > > >> > > >> _______________________________________________ > > >> Serusers mailing list > > >> Serusers(a)lists.iptel.org > <mailto:Serusers@lists.iptel.org> <mailto:Serusers@lists.iptel.org > <mailto:Serusers@lists.iptel.org>> > > <mailto:Serusers@lists.iptel.org > <mailto:Serusers@lists.iptel.org> <mailto:Serusers@lists.iptel.org > <mailto:Serusers@lists.iptel.org>>> > > >> http://lists.iptel.org/mailman/listinfo/serusers > > >> > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > <mailto: Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org>> > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > >

1 0

Re: [Serusers] Please help with LCR
by Steve Blair 10 Jan '07

10 Jan '07

Arun: This depends a little on your configuration. Generally speaking the !load_gws and !next_gw would be in route block 0. You will most likely want to do any per-call checks or rewrites before determining which gateway should receive the call. The failure_route can be inserted after your route blocks. -Steve Arun Kumar wrote: > Hi, > > thanks, One more thing can you please guide me where do I've to place > this code in my ser.cfg. > > thanks > > arun > > On 1/10/07, *Steve Blair * <blairs(a)isc.upenn.edu > <mailto:blairs@isc.upenn.edu>> wrote: > > > Arun: > > I would try something like the following: > > if (!load_gws()) { > sl_send_reply("500", "Server Internal Error - Cannot load > gateways"); > break; > } else { > xlog("L_INFO", "\n[SER]: - [$Tf] Method: <$rm> R-uri: <$ru> > From: <$fu> > To <$tu> : Successfully Loaded Gateway List\n"); > }; > > ...... > > if (!next_gw()) { > t_reply("503", "Service not available - No more gateways"); > break; > } else { > t_on_failure("2"); <--- This will be where you load the > next > gateway > t_relay(); > break; > }; > > failure_route[2] { > > if (!next_gw()) { > xlog("L_CRIT","No more gateways for <$tu>\n"); > t_reply("503", "Service not available - No more gateways"); > break; > } else { > t_on_failure("2"); > t_relay(); > break; > } > } > > -Steve > > Arun Kumar wrote: > > Hi > > > > thanks for your prompt reply but I've already checked and I'm > not able > > to make my lcr working by seeing these two pages I got this conf but > > > > This config is sending the call to my first gateway but when My > first > > gateway is down its not sending the calls to other gateways. Plese > > help me to make this working. Here is my config file. > > > > ------------------------ > > debug=3 > > fork=no > > log_stderror=yes > > > > listen= 192.168.2.210 <http://192.168.2.210> > <http://192.168.2.210/> # INSERT YOUR IP > > ADDRESS HERE > > port=5060 > > children=4 > > > > dns=no > > rev_dns=no > > fifo="/tmp/ser_fifo" > > fifo_db_url="mysql://ser:pass@localhost/ser" > > > > loadmodule "/usr/local/lib/ser/modules > > /mysql.so" > > loadmodule "/usr/local/lib/ser/modules/sl.so" > > loadmodule "/usr/local/lib/ser/modules/tm.so" > > loadmodule "/usr/local/lib/ser/modules/rr.so" > > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > > loadmodule "/usr/local/lib/ser/modules/usrloc.so" > > loadmodule "/usr/local/lib/ser/modules/registrar.so" > > loadmodule "/usr/local/lib/ser/modules/auth.so" > > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > loadmodule "/usr/local/lib/ser/modules/uri.so" > > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > > loadmodule "/usr/local/lib/ser/modules/domain.so" > > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > > loadmodule "/usr/local/lib/ser/modules/avpops.so" > > loadmodule "/usr/local/lib/ser/modules/textops.so" > > loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" > > loadmodule "/usr/local/lib/ser/modules/permissions.so" > > loadmodule "/usr/local/lib/ser/modules/acc.so" > > loadmodule "/usr/local/lib/ser/modules/auth_radius.so" > > loadmodule "/usr/local/lib/ser/modules/lcr.so" > > > > > > modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", " > > mysql://ser:pass@localhost/ser") > > 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) > > > > modparam("tm", "fr_inv_timer", 120) > > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > > > > modparam("permissions", "db_mode", 1) > > modparam("permissions", "trusted_table", "trusted") > > > > modparam("lcr","db_url", "mysql://ser:pass@localhost/ser") > > modparam("lcr","gw_table","gw") > > modparam("lcr","gw_name_column","gw_name") > > modparam("lcr","ip_addr_column","ip_addr") > > modparam("lcr","port_column","port") > > modparam("lcr","grp_id_column","grp_id") > > modparam("lcr","lcr_table","lcr") > > modparam("lcr","prefix_column","prefix") > > modparam("lcr","from_uri_column","from_uri") > > modparam("lcr","priority_column","priority") > > modparam("lcr","contact_avp","1401") > > modparam("lcr","fr_inv_timer_avp","fr_inv_timer_avp") > > modparam("lcr","fr_inv_timer",90) > > modparam("lcr","fr_inv_timer_next",60) > > # ------ Account and Radius modules-------------- > > modparam("acc", "db_url", " mysql://ser:pass@localhost/ser") > > > > modparam("acc","log_level",1) > > modparam("acc","log_flag",1) > > modparam("acc", "db_flag", 1) > > > > modparam("acc","radius_flag",1) > > modparam("acc","service_type",15) > > > #modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") > > > > > > route { > > > > # > ----------------------------------------------------------------- > > # Sanity Check Section > > # > ----------------------------------------------------------------- > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483", "Too Many Hops"); > > break; > > }; > > > > if (msg:len > max_len) { > > sl_send_reply("513", "Message Overflow"); > > break; > > }; > > > > # > ----------------------------------------------------------------- > > # Record Route Section > > # > ----------------------------------------------------------------- > > if (method=="INVITE" && client_nat_test("3")) { # setflag(1); > > # INSERT YOUR IP ADDRESS HERE > > record_route_preset("192.168.2.210:5060 > <http://192.168.2.210:5060> > > <http://192.168.2.210:5060/>;nat=yes"); > > } else if (method!="REGISTER") { > > record_route(); > > }; > > # > ----------------------------------------------------------------- > > # Call Tear Down Section > > # > ----------------------------------------------------------------- > > if (method=="BYE" || method=="CANCEL") { setflag(1); > > end_media_session(); > > }; > > > > # > ----------------------------------------------------------------- > > # Loose Route Section > > # > ----------------------------------------------------------------- > > if (loose_route()) { > > > > if ((method=="INVITE" || method=="REFER") && !has_totag()) { > > sl_send_reply("403", "Forbidden Cant cross 1st bridge"); > > break; > > }; > > > > if (method=="INVITE") { > > > > if (!allow_trusted()) { > > > > if (!proxy_authorize("","subscriber")) { > > proxy_challenge("","0"); > > break; > > } else if (!check_from()) { > > sl_send_reply("403", "Use From=ID"); > > break; > > }; > > > > consume_credentials(); > > }; > > > > if (client_nat_test("3")||search("^Route:.*;nat=yes")){ > > setflag(6); > > use_media_proxy(); > > }; > > }; > > > > route(1); > > break; > > }; > > # > ----------------------------------------------------------------- > > # Call Type Processing Section > > # > ----------------------------------------------------------------- > > if (!is_uri_host_local()) { > > if (is_from_local() || allow_trusted()) { > > sl_send_reply("404", "I am trusted host, "); > > route(4); > > route(1); > > } else { > > sl_send_reply("403", "Forbidden, I am here"); > > }; > > break; > > }; > > > > if (method=="ACK") { setflag(1); > > route(1); > > break; > > } if (method=="CANCEL") { setflag(1); > > route(1); > > break; > > } else if (method=="INVITE") { setflag(1); > > # sl_send_reply("410", "I am sending your call, "); > > route(3); > > break; > > } else if (method=="REGISTER") { > > route(2); > > break; > > }; > > > > lookup("aliases"); > > if (!is_uri_host_local()) { > > route(4); > > route(1); > > break; > > }; > > > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); > > break; > > }; > > > > route(1); > > } > > > > route[1] { > > > > # > ----------------------------------------------------------------- > > # Default Message Handler > > # > ----------------------------------------------------------------- > > > > t_on_reply("1"); > > > > if (!t_relay()) { > > > > if (method=="INVITE" || method=="ACK") { setflag(1); > > 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"); > > break; > > }; > > if (!check_to()) { > > sl_send_reply("401", "Unauthorized"); > > break; > > }; > > > > consume_credentials(); > > > > if (!save("location")) { > > sl_reply_error(); > > }; > > } > > > > route[3] { > > > > # > ----------------------------------------------------------------- > > # INVITE Message Handler > > # > ----------------------------------------------------------------- > > > > if (client_nat_test("3")) { > > setflag(7); > > force_rport(); > > fix_nated_contact(); > > }; > > > > if (!allow_trusted()) { > > > > if (!proxy_authorize("","subscriber")) { > > proxy_challenge("","0"); > > break; > > } else if (!check_from()) { > > sl_send_reply("403", "Use From=ID"); > > break; > > }; > > > > consume_credentials(); > > }; > > > > route(4); > > > > if(!load_gws()){ > > sl_send_reply("500", "Internal server error, unable to load > gateways"); > > break; > > } > > if(!next_gw()){ > > sl_send_reply("503", "Service not available, no gateways found"); > > break; > > }; > > > > if (!is_uri_host_local()) { > > route(4); > > route(1); > > break; > > }; > > > > > > # route(1); > > } > > > > route[4] { > > > > # > ----------------------------------------------------------------- > > # NAT Traversal Section > > # > ----------------------------------------------------------------- > > > > if (isflagset(6) || isflagset(7)) { > > if (!isflagset(8)) { > > setflag(8); > > use_media_proxy(); > > }; > > }; > > } > > > > 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(); > > }; > > } > > > > > > thanks very much > > > > arun > > > > On 1/10/07, * Greger V. Teigre* <greger(a)teigre.com > <mailto:greger@teigre.com> > > <mailto: greger(a)teigre.com <mailto:greger@teigre.com>>> wrote: > > > > A quick search on iptel.org <http://iptel.org> > <http://iptel.org> would have given > > you these: > > http://www.iptel.org/lcr_module_drop_in_for_0_9_x > > http://www.iptel.org/lcr_with_mysql_accounting > <http://www.iptel.org/lcr_with_mysql_accounting> > > > > > > Arun Kumar wrote: > >> Hi > >> > >> please help with LCR can some one show his lcr config file > >> please. I'm not able to make it working. > >> > >> please help > >> > >> thanks > >> arun > >> > ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > <mailto:Serusers@lists.iptel.org <mailto:Serusers@lists.iptel.org>> > >> http://lists.iptel.org/mailman/listinfo/serusers > >> > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > > http://lists.iptel.org/mailman/listinfo/serusers > > > >

1 0

[Serusers] Please help with LCR
by Arun Kumar 10 Jan '07

10 Jan '07

Hi please help with LCR can some one show his lcr config file please. I'm not able to make it working. please help thanks arun

3 3

[Serusers] OT? Duplicated digits.
by Juan 10 Jan '07

10 Jan '07

My system is DIDprovider ---SER---Cisco AS5350---Calling card platform. Since December the 1st we noticed a large increase in the amount of wrong numbers received at the CC platform. As this problem arrises once the call is established and there is no RTP relay I don't think is a SER question but I resort to your major knowledge to see if someone can enlight me about where to look to solve the problem. BTW. No changes were made in the equipment at neither end. Thanks in advance Juan

2 1

[Serusers] BYE and SER.cfg
by Benito Caracuel 10 Jan '07

10 Jan '07

Hello all, In my ser.cfg I do: route{.... if (method=="BYE"){log(1,"enter in method bye")do ABCD.....} ....} When SER receives a BYE, SER doesn´t do "ABCD...." and if I put SER in debug mode, I can`t read the log "enter in method bye" but I can read the line"method: <BYE>" why doesn`t work "if method="BYE"? Thanks. _________________________________________________________________ Llama a tus amigos de PC a PC: ¡Es GRATIS! http://get.live.com/messenger/overview

3 2

Re: [Serusers] One way audio
by Greger V. Teigre 10 Jan '07

10 Jan '07

You need to look at when NAT is detected (nat_uac_test) for INVITE and for 200 OK in onreply route for the same call. Make sure NAT is detected for both, if not, find out why it is not detected. g-) Gareth Owen wrote: > > Thanks Greger, I've already worked through the getting started doc and > think I've got a good understanding but am still getting the one-way > audio. Can you suggest anything to help me understand how I can track > the problem? Thanks gareth. > > > > ------------------------------------------------------------------------ > > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* 04 January 2007 06:5 > *To:* gareth.owen(a)bevoice.net > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] One way audio > > > > Try gettingstarted doc. http://iptel.org/ser/doc/gettingstarted > g-) > > Gareth Owen wrote: > > Can someone help? > > > > I am fairly new to SER and Mediaproxy and can't find a detailed > explanation of how nathelper etc work, so I'm struggling! > > > > My config is:- > > > > SER 0.9.6 > > Mediaproxy 1.7.2 > > Fedora Core 4 86x64 > > > > SER and Mediaproxy run on the same server which acts as a gateway to a > remote Sonus PSTN gateway over a VPN. > > > > I also use Asterisk on my local network. The SER server acts as a > router between Asterisk servers and the PSTN via the Sonus. This works > fine but my problem is when I attempt to use X-Lite 3.0 UA behind NAT > on a remote network. Asterisk is not used in this config but proves > that two way audio is fine using SER and the Sonus gateway. > > > > Problem: X-Lite behind NAT on remote network to Sonus PSTN gateway via > SER and Mediaproxy, I get only one ay audio. The audio is fine from > X-Lite to PSTN but not in the reverse direction. > > > > Thanks for your help > > ** ** > > ** ** > > ** ** > > ** ** > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] Can I see your PSTN with Accounting?
by Bart Fisher 10 Jan '07

10 Jan '07

Noob here, sorry :) I've been working with SER for about a month and I have a working config using what's provided by iptel.org 'getting started' gw-pstn.cfg - But here's were I'm stuck. I'd like to add accounting, but I haven't found a good example based on gw-pstn.cfg where accounting has been added. Google turns up old updated information. I'm hoping someone can post or email their gw-pstn.cfg with acc enabled: Note: Basically the gw-pstn.cfg script provides a good example of using SER with nathelper and mediaproxy as well as rewriting to a PSTN Gateway Can you help me? TIA Bart bart(a)icpage.com

2 1

[Serusers] Question
by Bill 10 Jan '07

10 Jan '07

I am setting up a SER/Mediaproxy/MySQL box. I have it setup and running except for NAT. When I put up a call this is what my session look like. The 127 are the fakeconversations.py. The other session is my live traffic but the other end has all ????????????????. Any ideas. I am using the basic SER/Mediaproxy config below. Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------- ---------------------------------------- 127.0.0.1:10000 - 10.0.1.100:60004 - 127.0.0.1:10002 active 0'10" Dynamic(127) Audio 94.38k/188.58k/94.38k 10.0.2.1:15040 - 10.0.1.100:60000 - ?.?.?.?:? inactive 4'41" Dynamic(107) Audio 1.58M/0/0 Total traffic: 120.55kbps/74.61kbps/149.22kbps (in1/in2/out) Session count: 2 Proxy version: 1.8.0 #####################SER.CFG [root@voipproxy1 mediaproxy]# cat /usr/local/etc/ser/ser.cfg # $Id: nat-mediaproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=7 fork=yes log_stderror=no listen=10.0.1.100 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 sock_mode=0666 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") 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) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("10.0.1.100:5060;nat=yes"); } 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 ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- 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"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } 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(); }; } ######################MEDIAPROXY.INI [Dispatcher] defaultProxy = /var/run/mediaproxy.sock [MediaProxy] ;start = yes ;socket = /var/run/mediaproxy.sock ;group = openser ;listen = None ;allow = None proxyIP = 10.0.1.100 portRange = 60000:65000 ;TOS = 0xb8 ;idleTimeout = 60 ;holdTimeout = 3600 ;forceClose = 0 [Accounting] ; one of none, radius or database accounting = database [Database] user = ser password = heslo host = localhost database = ser table = acc [Radius] secret = secret server = localhost authport = 1812 acctport = 1813 dictionaries = /etc/radiusclient-ng/dictionary, /etc/openser/radius/dictionary, /usr/local/mediaproxy/dictionary retries = 2 timeout = 3 Bill

2 1

[Serusers] ser_ctl
by Jean Sigue 10 Jan '07

10 Jan '07

Hi, After compiling the pre-release 0.10 , I still have the serctl not ser_ctl. How do i fix it Jean R. Sigue Senior Network Engineer CCNP, CSS1, NSA OMX Global Services USA Direct: 1 908-673-6004 Mobile: (917-257-9742) Fax: 1 908-673-6057 Visiting address: 2 Connell Drive Berkeley Heights, NJ 07922 <http://www.omxgroup.com/>www.omxgroup.com No. 1 in marketplace solutions ******************************************************************************** This e-mail and the information it contains may be privileged and/or confidential. It is for the intended addressee(s) only. The unauthorised use, disclosure or copying of this e-mail, or any information it contains, is prohibited. If you are not an intended recipient, please contact the sender and delete the material from your computer. ********************************************************************************

4 4

[Serusers] Error with MTU size in SER with MIPv6
by Jose Angel Calvo 10 Jan '07

10 Jan '07

Thank you for your help and sorry for my English. I have a problem with mipv6 and a server SIP, SER (SIP Express Router). The mipv6 path is "mipv6-2.0.2" from http://www.mobile-ipv6.org/ ,in kernel 2.6.16.24-Ubuntu. The problem occurs when a packet is too big (it must be fragmented), and go to the node what makes function of P-CSCF (Proxy in SIP) and Correspondent Node. If the P-CSCF is not a CN, all is OK. But if the node is CN, then i have the problem. The SER logs are: ############################################################## ................ Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: SER: new INVITE Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: parse_headers: flags=-1 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: check_via_address(2001:720:1500:1B:FCFD:FF:FE0C:103, [2001:720:1500:1B:FCFD:FF:FE0C:103], 3) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: ERROR: warning_builder: buffer size exceeded Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: WARNING: warning skipped -- too big Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: WARNING:vqm_resize: resize(0) called Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: reply sent out. buf=0x8106270: SIP/2.0 1..., shmem=0xb5e8b330: SIP/2.0 1 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: _reply_light: finished Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: DEBUG: mk_proxy: doing DNS lookup... Nov 20 18:28:25 pcscfD /usr/sbin/ser[8051]: check_via_address(2001:720:1500:1B:FCFD:FF:FE0C:103, [2001:720:1500:1B:FCFD:FF:FE0C:103], 3) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8068]: DBG: tcp_main_loop: dead child 2 (shutting down?) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: child process 8051 exited by a signal 11 Nov 20 18:28:25 pcscfD kernel: [4296472.851000] <0>skb_under_panic: text:c02d2aad len:321 put:14 head:d62cfc00 data:d62cfbea tail:d62cfd2b end:d62cfd80 dev:eth0 Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: core was not generated Nov 20 18:28:25 pcscfD kernel: [4297162.639000] ------------[ cut here ]------------ Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: INFO: terminating due to SIGCHLD Nov 20 18:28:25 pcscfD kernel: [4297162.639000] kernel BUG at net/core/skbuff.c:112! Nov 20 18:28:25 pcscfD /usr/sbin/ser[8059]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8052]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8053]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8054]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8055]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8056]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8057]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8058]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8060]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8061]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8062]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8063]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8064]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8065]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8066]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8067]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8068]: INFO: signal 15 received Nov 20 18:28:25 pcscfD /usr/sbin/ser[8050]: INFO: signal 15 received Nov 20 18:28:25 pcscfD kernel: [4297162.639000] invalid opcode: 0000 [#3] Nov 20 18:28:25 pcscfD /usr/sbin/ser[8059]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8052]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8053]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8054]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8055]: Memory status (pkg): Nov 20 18:28:25 pcscfD /usr/sbin/ser[8056]: Memory status (pkg): ....................... Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: hash = 2055 fragments no.: 1, unused: 0 ^I^I bucket size: 1048576 - 2097152 (first 1572864) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: hash = 2059 fragments no.: 1, unused: 0 ^I^I bucket size: 16777216 - 33554432 (first 31934160) Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: TOTAL: 26 free fragments = 33537656 free bytes Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: ----------------------------- Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: shm_mem_destroy Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: destroying the shared memory lock Nov 20 18:28:25 pcscfD /usr/sbin/ser[8049]: terminating due to SIGCHLD ############################################################## The problem is in the MTU size, because if the packet is not fragmented, there is not a problem. But the INVITE packet needs to fragment, what it means, i always have the same problem. My SER version is version: ser 0.9.6 (i386/linux) and it's a vanilla version. The packets is sent over UDP. My questions are: - Have anybody the same o similar problem? - The MTU size is a problem in SER? - Have anybody sent fragmented packets with SER? Thanks for your help. __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.es

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[Serusers] SER LCR Config
by Arun Kumar 10 Jan '07

10 Jan '07

Hi, I'm trying to configure LCR with ser here is my config file. This config is sending the call to my first gateway but when My first gateway is down its not sending the calls to other gateways. Plese help me to make this working. ------------------------ debug=3 fork=no log_stderror=yes listen=192.168.2.210 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:pass@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/lcr.so" modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", " mysql://ser:pass@localhost/ser") 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) modparam("tm", "fr_inv_timer", 120) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("lcr","db_url", "mysql://ser:pass@localhost/ser") modparam("lcr","gw_table","gw") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","grp_id_column","grp_id") modparam("lcr","lcr_table","lcr") modparam("lcr","prefix_column","prefix") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","contact_avp","1401") modparam("lcr","fr_inv_timer_avp","fr_inv_timer_avp") modparam("lcr","fr_inv_timer",90) modparam("lcr","fr_inv_timer_next",60) # ------ Account and Radius modules-------------- modparam("acc", "db_url", "mysql://ser:pass@localhost/ser") modparam("acc","log_level",1) modparam("acc","log_flag",1) modparam("acc", "db_flag", 1) modparam("acc","radius_flag",1) modparam("acc","service_type",15) #modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # setflag(1); # INSERT YOUR IP ADDRESS HERE record_route_preset("192.168.2.210:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden Cant cross 1st bridge"); break; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { sl_send_reply("404", "I am trusted host, "); route(4); route(1); } else { sl_send_reply("403", "Forbidden, I am here"); }; break; }; if (method=="ACK") { setflag(1); route(1); break; } if (method=="CANCEL") { setflag(1); route(1); break; } else if (method=="INVITE") { setflag(1); # sl_send_reply("410", "I am sending your call, "); route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { setflag(1); 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"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; route(4); if(!load_gws()){ sl_send_reply("500", "Internal server error, unable to load gateways"); break; } if(!next_gw()){ sl_send_reply("503", "Service not available, no gateways found"); break; }; if (!is_uri_host_local()) { route(4); route(1); break; }; # route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } 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(); }; } ------------------------------- thanks arun

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[Users] uri_radius and radius_does_user_exist() not working
by Toni Heinonen 09 Jan '07

09 Jan '07

Hi, I've been through the OpenSER + RADIUS configuration tutorial many, many times, and it works like a charm, except for the uri_radius module which I can't get to detect user existence with the radius_does_user_exist() function. I'm trying to decide whether to send a 404 or a 480, as such: if (!lookup("location")) { if(radius_does_uri_exist()) { sl_send_reply("480", "User offline"); } else { sl_send_reply("404", "User not found"); }; }; It worked identically with the uri_db module's does_user_exist() function. Now the RADIUS server doesn't seem to understand the Call-Check request. I have a dump here: http://tonih.iki.fi/temp/uri_radius.cap Can anyone guess why OpenSER always gives a 404 even for users that exist, but that are simply offline? The users are currently hand-configured into freeradius's text configuration file users.conf. PS. What have you found to be the best way to authenticate users from a domain? FreeRADIUS using Kerberos 5, LDAP or relaying to a Microsoft RADIUS (IAS) server? -- http://tonih.iki.fi/ ~ http://blogit.helsinki.fi/toni.heinonen/ "The progress of a dynamic civilization depends on the special people who make play out of work. In their all-absorbing passion, they create the variations that, through trial and error, become the sources of progress. They make the discoveries that drive the infinite series." - Virginia Postrel

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[Users] sip register
by sinclair fils 09 Jan '07

09 Jan '07

i am developping one sip dialer using Microsoft RTC in vb6 i want to register in my opnser server put no request of registration are send to my server any help __________________________________________________ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicités http://mail.yahoo.fr Yahoo! Mail

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[Users] NEW FEATURE: error_route - automatic error handling
by Daniel-Constantin Mierla 09 Jan '07

09 Jan '07

Hello, today was introduced in development version a framework for automatic error handling. Based on past experiences and comments on the mailing lists, error handling was a bit deficient from the configuration file. Log files could get full of error messages when SIP messages were incorrect, the processing continued even it should have stopped. With the new framework, the administrator can write a error_route block and there decide what to do when an error is encountered. Right now, error_route is executed for parsing errors in sip requests - one can send back now the appropriate reply. I use this email to start a discussion about how to extend it, there could be system errors (e.g., memory, database access, ...) or errors in replies (here we can use same route block and check the type of sip message not to force reply to reply - a new more flexible and easier way to access pseudo-variables is underway). In the error_route are available 4 new pseudo-variables: - $err.class - the class of error (now is ‘1’ for parsing errors) - $err.level - severity level for the error - $err.info - text describing the error - $err.rcode - recommended reply code - $err.rreason - recommended reply reason phrase With the new sl_reply() from sl module you can reply with the appropriate code and reason from error_route[] - here is an example: error_route { xlog("--- error route class=$(err.class) level=$(err.level) info=$(err.info) rcode=$(err.rcode) rreason=$(err.rreason) ---\n"); xlog("--- error from [$si:$sp]\n+++++\n$mb\n++++\n"); sl_reply("$err.rcode", "$err.rreason"); exit; } In addition to sl_send_reply(), append_to_reply() and xlog functions can be used in error_route. You have to use 'exit' if you do not want to continue the execution when an error occurred. Error level is meant to show if the processing can continue without any issue, or if the error allows to send reply or not. Please note that malformed SIP messages can still go through your proxy in the case the processing didn't discover it (OpenSER has an incremental parser, it parses only what it needs). If no error_route is defined, the functionality is same as before. Looking forward to feedback about this new automatic error handling. Cheers, Daniel

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[Serusers] serweb 0.10.0 - error: Allowed memory size of 8388608 bytes exhausted
by "tzieleniewski" 09 Jan '07

09 Jan '07

Hi After instaling serweb 0.10.0 for ser 0.10.99 when I try to log into user aplication I get the following error: Fatal error: Allowed memory size of 8388608 bytes exhausted (tried to allocate 9699 bytes) in /home/ser/serweb/serweb-0.10.0/smarty/Smarty_Compiler.class.php on line 196 I configured serweb with the multidomain support according to README in modules/multidomain/. What can I do with that? And what information do You need to investigate the probelem? Bests Tomasz

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[Users] Re: [Devel] OpenSER not sending Ack on Request Cancel
by Klaus Darilion 09 Jan '07

09 Jan '07

Rajesh Kalagarla wrote: > Hi Klaus, > > As per rfc 3261, if the proxy is stateful the CANCEL request will be a > hop-by-hop request. in that case Relay is not the correct option right? It is the correct function to handle CANCEL. t_relay will pass the CANCEL to openser's transaction modul. The transaction module usualy will detect that this CANCEL is for an ongoing transaction and takes care of the hop-by-hop CANCEL. This should also take care of the hop-by-hop ACK to the 487 response. regards klaus > > Thanks & Regards, > Rajesh > > ----- Original Message ----- > From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> > To: "Rajesh Kalagarla" <RajeshK(a)idnltd.com> > Cc: <devel(a)openser.org> > Sent: Tuesday, January 09, 2007 10:38 AM > Subject: Re: [Devel] OpenSER not sending Ack on Request Cancel > > >> Hi! >> >> This indeed looks strange. Are you using stateful forwarding (t_relay)? >> >> It loks like your are using lookup() also for CANCEL - try to avoid that >> and just t_relay: >> >> if (is_method("CANCEL")) { >> t_relay(); >> } >> >> regards >> klaus >> >> Rajesh Kalagarla wrote: >>> Hi Klaus, >>> >>> here is the log. >>> >>> Thanks & Regards, >>> Rajesh >>> ----- Original Message ----- >>> From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> >>> To: "Rajesh Kalagarla" <RajeshK(a)idnltd.com> >>> Cc: <devel(a)openser.org> >>> Sent: Monday, January 08, 2007 1:00 PM >>> Subject: Re: [Devel] OpenSER not sending Ack on Request Cancel >>> >>> >>>> Rajesh Kalagarla wrote: >>>>> Hello All, >>>>> >>>>> OpenSER is not responding with a Ack for the call that was terminated >>> with CANCEL because of no response from remote user. because of this SIP >>> client is keep retransmitting the 487 Response for the Invite request. >>>>> is it a bug or something need to be enabled? >>>> Please send a "ngrep -W byline port 5060" dump. >>>> >>>> regards >>>> klaus >>>> >>>> >>>>> Thanks, >>>>> Rajesh >>>>> >>>>> >>>>> >>>>> >>>> ------------------------------------------------------------------------ >>>>> _______________________________________________ >>>>> Devel mailing list >>>>> Devel(a)openser.org >>>>> http://openser.org/cgi-bin/mailman/listinfo/devel >>>> -- >>>> Klaus Darilion >>>> nic.at >>>> >>>> >> >> -- >> Klaus Darilion >> nic.at >> >> > > -- Klaus Darilion nic.at

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RE: [Users] SER as proxy in front of Asterisk
by Thomas Deillon 09 Jan '07

09 Jan '07

Behind a PIX, it works Behind a SonicWall without the SIP_aware option, it doesn't work Behind a Watchguard, it doesn't work too (and there is no SIP_aware option) In fact, I have a centrex IP and many clients in the country with, many different configurations and I have to be sure that all works. It's why I have no enable the SIP_aware option in the sonicwall. Thanks for your help, Thomas -----Message d'origine----- De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Thomas Deillon Envoyé : mercredi, 3. janvier 2007 17:06 À : daniel(a)voice-system.ro Cc : users(a)openser.org Objet : RE: [Users] SER as proxy in front of Asterisk Hi, In fact, now, every phone does not response. The options didn't go thought the FW, so, the phone cannot answer. I don't understand why the traffic don't go thought it !! I have test this configuration with two firewall: a sonicwall and a soho. Cheers, Thomas -----Message d'origine----- De : Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Envoyé : mercredi, 3. janvier 2007 14:52 À : Thomas Deillon Cc : users(a)openser.org Objet : Re: [Users] SER as proxy in front of Asterisk Hello, what is the type of NAT/Firewall you have in front of Thomson? As I can see from the diagram, the phone does not reply to OPTIONS request, and I guess, the nat/firewall needs outward traffic to keep the pinhole open. Check why thomson ignores the options requests. Cheers, Daniel On 01/03/07 15:17, Thomas Deillon wrote: > > Hi, > > I have made more tries and now, all my phones don't work ... > > The firewall opens a connection to REGISTER phone but after a while > the OPTION message from the Asterisk will be DROP by the FW ... > > You can find a diagram of the exchange here: > http://deillon.eu/tmp/exchange_sip.png > > A list of all exchange: http://deillon.eu/tmp/ip_sip.txt > > And all sip message: http://deillon.eu/tmp/message.txt > > Thanks a lot for your help, > > Thomas > > ------------------------------------------------------------------------ > > *De :* users-bounces(a)openser.org [mailto:users-bounces@openser.org] > *De la part de* Thomas Deillon > *Envoyé :* mardi, 2. janvier 2007 11:44 > *À :* users(a)openser.org > *Objet :* [Users] SER as proxy in front of Asterisk > > Hi all, > > I wish you first a happy new year !! > One again, I ask you some help. Thanks a lot for your patience and > your answers that really helped me. > > So, I want to put a SER in front of one Asterisk for the moment (more > after). > All Servers have public IP address and so, I don't care about the > RTP/SDP messages. > All phones are behind NAT somewhere in Switerland :) > > > I have the configuration below in openser: > > > modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") > #modparam("dispatcher", "force_dst", 1) > modparam("dispatcher", "flags", 2) > > modparam("usrloc", "db_mode", 0) > #modparam("rr", "enable_full_lr", 1) > > route{ > > > xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > }; > > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > > # if (search("User-Agent:.*Thomson.*")) { > # }; > > > if ((src_ip==212.xxx.xxx.152) || (src_ip==212.xxx.xxx.153)) { > if(method=="OPTIONS") { > }; > avp_pushto("$ru","$tu"); > forward(); > > }else { > fix_nated_contact(); > force_rport(); > if(method=="REGISTER"){ > ds_select_dst("4", "0"); > t_relay(); > } > else { > ds_select_dst("0", "4"); > forward(); > } > } > } > > > And in my dispatcher.list I have: > > 0 sip:212.xxx.xxx.153:5060 > 4 sip:212.xxx.xxx.153:5060 > > > > With this configuration, a snom phone, a X-lite phone and a Cisco > phone seams to work .... > But with the Thomson ST2030, it doesn't work. > In fact, the Thomson REGISTER on asterisk, the Asterisk send a OPTION > message to the FW IP adress and the right port where the thomson is > and then, after a while, the FW close the connection. The thomson > phone is so "UNREACHABLE" on asterisk status and nobody can call it. > I'm not sure that my configuration is ok in ser but I think that is a > problem with the Thomson2030. Do you had the same kind of problem or > do you understand the problem here ? > > > Thanks for your help, > > Thomas Deillon > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] Failover with messages BYE
by Jose Gil Navarrete 09 Jan '07

09 Jan '07

Hi list. I try to use the module dispatcher like failover, when the call is established and after the proxy is down then resend the BYE message at the next proxy using ds_select_dst("2", "4") and forward the message, but I obtain the message 503 of the failure_route[2], Any idea?? Thanks in advance In my route seccion I made the next: if (loose_route()) { if (search("Path")) { route(1); }; if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); exit; }; route(5); }; route[5] { t_on_failure("2"); t_relay(); exit; } failure_route[2] { #500, 503, 504 if ( t_check_status("408") | t_check_status("500") | t_check_status("503") | t_check_status("504") ) { if (ds_next_domain()) { #if (ds_next_dst()) { # On Failure: Choose next Proxy t_on_failure("2"); # Relay to Proxy t_relay(); } else { t_reply("503", "Service not available"); return; } }; } __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/

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[Serusers] Work with a new column in SER.subscriber
by inge 09 Jan '07

09 Jan '07

Hi all, I add a new column named "client" in subscriber table. Next, I want to do a test wich seems like that in SQL : "SELECT client FROM subscriber WHERE username = "actual calling number" => equivalent of avp_db_load I think ? Then, I think I must do a test on the previous values with an if and avp_check I think. But my problem is to do this both requests with SER syntax in ser.cfg. Someone can help me ? Thanks

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[Serusers] ser_ctl: (111, 'Connection refused'): error
by "tzieleniewski" 09 Jan '07

09 Jan '07

Hi ! I've set up the ser 0.10.99, basic functionality works - ser registers users and establishes the media sessions correctly. Now I tried to use the new serctl tool: version 0.3.17. When I try to invoke for instance #ser_ctl ps while ser is running I get the following: ser_ctl: (111, 'Connection refused'): error my ser.cfg file part: port=5060 ... modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") modparam("ctl", "mode", 0666) and my serctl.conf and serctl/config.py files lines: .. SER_URI = 'http://localhost:5060/' ... Please point me what do I need to do to make it work. Best Tomasz

3 6

Re: Re: Re: [Serusers] ser_ctl: (111, 'Connection refused'): error
by Michal Matyska 08 Jan '07

08 Jan '07

I did copy&paste from one of my testing scripts I have on disk .... as it contained the create_via (which is not neccesary now) it was designed for the older xml-rpc way of handling. You can safely use the code from the CVS ser.cfg example, it's kept up to date. Michal On Mon, 2007-01-08 at 19:55 +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > I have something like this > if ((method=="POST" || method=="GET") & src_ip==127.0.0.1) { > if (msg:len >= 8192) { > sl_send_reply("513", "Request to big"); > drop; > } > dispatch_rpc(); > drop; > } > taken from the latest cvs etc/ser.cfg file. > Is this rewriteuri(...) required? Does it imply that sip:xmlrpc.. is somehow reserved? > > > > The ctl module serves the sercmd (utils/sercmd). > > > > If you want to use ser_ctl then you have to support it in the script > > file like this: > > > > if (method=="POST" || method=="GET") { > > # create_via(); shouldn't be required with the latest CVS source > > rewriteuri("sip:xmlrpc@127.0.0.1:5060"); > > dispatch_rpc(); > > drop; > > } > > > > Michal > > > > On Mon, 2007-01-08 at 17:26 +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > > > Results in the attached files > > > > > > tomasz > > > > > > > Plese, try 'netstat -tupan' and 'ser_ctl -Gg ps' and post results. -- hallik > > > > > > > > On Mon, Jan 08, 2007 at 04:41:59PM +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > > > > > Hi ! > > > > > > > > > > I've set up the ser 0.10.99, basic functionality works - ser registers users > > > > > and establishes the media sessions correctly. Now I tried to use the new serctl > > > > > tool: version 0.3.17. When I try to invoke for instance #ser_ctl ps while ser is > > > > > running I get the following: > > > > > ser_ctl: (111, 'Connection refused'): error > > > > > > > > > > my ser.cfg file part: > > > > > port=5060 > > > > > ... > > > > > modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") > > > > > modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") > > > > > modparam("ctl", "mode", 0666) > > > > > > > > > > and my serctl.conf and serctl/config.py files lines: > > > > > .. > > > > > SER_URI = 'http://localhost:5060/' > > > > > ... > > > > > > > > > > Please point me what do I need to do to make it work. > > > > > Best > > > > > Tomasz > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > Serusers mailing list > > > > > Serusers(a)lists.iptel.org > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

Re: Re: Re: [Serusers] ser_ctl: (111, 'Connection refused'): error
by Michal Matyska 08 Jan '07

08 Jan '07

serctl (note the missing underscore) was the only tool for ser < 0.9.x ser 0.10 introduced xmlrpc interface (basically the dispatch_rpc in the script calls the function) and ser_ctl family of tools was introduced (ser_rpc just for the XML-RPC call, ser_user, ser_domain, ..... specialized (the name suggests what they do), ser_ctl as the shell around all of them. They are written in python. Due to the fact of XML parsing, etc. the binary protocol was re-introduced. You need to load the ctl module, set the protocol, address, port) and use sercmd as cli. All methods which are callable via XML-RPC are visible via sercmd, it is just protocol between ser and controller. So if you have configured both was you can use ser_rpc and sercmd interchangeable. For the methods available use: sercmd system.listMethods Michal P.S. Please keep always the list in To/Cc to follow the thread. Somebody else might be interested as well. On Mon, 2007-01-08 at 20:01 +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > Just to make sure:) > So sercmd is a tool used to send command directly so through ser unix socket and rpc calls are entrily handled by xmlrpc module after dispatch_rpc() method?? > > > > The ctl module serves the sercmd (utils/sercmd). > > > > If you want to use ser_ctl then you have to support it in the script > > file like this: > > > > if (method=="POST" || method=="GET") { > > # create_via(); shouldn't be required with the latest CVS source > > rewriteuri("sip:xmlrpc@127.0.0.1:5060"); > > dispatch_rpc(); > > drop; > > } > > > > Michal > > > > On Mon, 2007-01-08 at 17:26 +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > > > Results in the attached files > > > > > > tomasz > > > > > > > Plese, try 'netstat -tupan' and 'ser_ctl -Gg ps' and post results. -- hallik > > > > > > > > On Mon, Jan 08, 2007 at 04:41:59PM +0100, =?UTF-8?Q? tzieleniewski ?= wrote: > > > > > Hi ! > > > > > > > > > > I've set up the ser 0.10.99, basic functionality works - ser registers users > > > > > and establishes the media sessions correctly. Now I tried to use the new serctl > > > > > tool: version 0.3.17. When I try to invoke for instance #ser_ctl ps while ser is > > > > > running I get the following: > > > > > ser_ctl: (111, 'Connection refused'): error > > > > > > > > > > my ser.cfg file part: > > > > > port=5060 > > > > > ... > > > > > modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") > > > > > modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") > > > > > modparam("ctl", "mode", 0666) > > > > > > > > > > and my serctl.conf and serctl/config.py files lines: > > > > > .. > > > > > SER_URI = 'http://localhost:5060/' > > > > > ... > > > > > > > > > > Please point me what do I need to do to make it work. > > > > > Best > > > > > Tomasz > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > Serusers mailing list > > > > > Serusers(a)lists.iptel.org > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Users] Caller-ID setup help
by Chahn Kim 08 Jan '07

08 Jan '07

Hi, a Happy New Year to all. Looking for help to get our caller-id working for outging PSTN calls. We have 3rd party PSTN gateway service and all our callee gets the gateway phone number as CID for all of our sip calls to PSTN. I had implemented so far sip-rpid, remote-party-id, through avp, append_rpid_hf() and confirmed rpid headers on all pstn outgoing frames. However, callee only see the SP's PSTN gateways number as incoming caller-id. Is there other way of implementing caller-id for sip calls before I bring the subject to PSTN service providor? Or is it most likely SP's policy setup? Thanks in advnace. John K

3 4

[Serusers] Instant Messaging Chat/Presence Client Please
by Gould, Aaron 08 Jan '07

08 Jan '07

Does anyone know of a free SIP IM/Presence Chat client that allows for multiple users conferencing? Aaron

2 1

[Users] Chunking Messages
by Danilo Vilela Dantas 08 Jan '07

08 Jan '07

Hi Congratulations! Does Openser supports chunking messages? Is it support at MSRP Relay? Thanks! Danilo

2 1

Re: [Serusers] Problemswith compiling ser 0.10.99
by Dmitry Isakbayev 08 Jan '07

08 Jan '07

Hi, The OSP toolkit project is being moved from sipfoundry to sourceforge. It should be available shortly. Meanwhile, the latest snapshot of the toolkit can be downloaded from http://www.transnexus.com/OSP%20Toolkit/OSP%20Toolkit.htm Regards, Dmitry > Date: Mon, 08 Jan 2007 11:25:24 +0100 > From: "tzieleniewski" <tzieleniewski(a)o2.pl> > Subject: [Serusers] Problemswith compiling ser 0.10.99 > To: serusers(a)lists.iptel.org > Message-ID: <54c54dce.20e2e287.45a21c14.f0624(a)o2.pl> > Content-Type: text/plain; charset="utf-8" > > Hi All!! > I am trying to compile the lattest cvs version of ser but I get errors > which I cann't handle myself so i kindly ask for your help:) > What I want to achive basically is the ser instance with presence > support. > I am compiling ser in the following way: > make group_include="standard standard-dep stable" all > I read that osp requires osptoolkit i tried to get it from sipfoundry > but i couldn't find any source of that. > I also cann't fight problems with tls i didn't includethe experimental > modules(tls) but still I get some errors. > Below is the make output > > >

1 0

[Serusers] Sipp and SER using TCP protocol
by Zeeshan Sajid 08 Jan '07

08 Jan '07

Hey ppl, I am trying to find out the peak number of calls that ser supports using sipp running the default scenario. I am using SER-0.9.6 and sipp version1.8 for this purpose. Now my setup on a local lan is SIpp (UAC) -----------------------> SER --------------------> Sipp (UAS) 192.168.0.8 192.168.0.200 192.168.0.2 the commands that i have used are for Sipp (UAC) ulimit -n 99999 ./sipp -sn uac -rsa 192.168.0.200:5060 -i 192.168.0.8 -p 5061 192.168.0.2:5060 -t t1 -r 1000 -rp 10 for Sipp (SER) default settings for Sipp (UAS) ulimit -n 99999 ./sipp -sn uas -i 192.168.0.2 -p 5060 -rsa 192.168.0.200:5060 -t tn now the problem is that after a while the UAS crashes saying "Broken pipe on TCP connection, remote peer probably closed the socket." my understanding is that SER is unable to handle the messages burst to the Sipp UAS and so please tell me how can i solve this problem .... thanks in advance regards Zeeshan Sajid __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 2

[Serusers] ser 0.10.99 problems with ser.cfg
by "tzieleniewski" 08 Jan '07

08 Jan '07

Hi!! Right now I am trying to migratemy ser.cfg file from ser0.9.7 to ser 0.10.99 when I try to check the ser.cfg file structure I get errors: 0(28672) parse error (252,16-17): unknown command, missing loadmodule? the errors correspond to the following line in the ser.cfg file: www_challenge("scscf.ims.touk.pl","0"); if (!check_to()) { proxy_challenge("scscf.ims.touk.pl","0"); } else if (!check_from()) { below is the list of my loaded modules: loadmodule "/home/ser/src/ser-0.10.x/modules/ctl/ctl.so" loadmodule "/home/ser/src/ser-0.10.x/modules/mysql/mysql.so" loadmodule "/home/ser/src/ser-0.10.x/modules/sl/sl.so" loadmodule "/home/ser/src/ser-0.10.x/modules/tm/tm.so" loadmodule "/home/ser/src/ser-0.10.x/modules/rr/rr.so" loadmodule "/home/ser/src/ser-0.10.x/modules/maxfwd/maxfwd.so" loadmodule "/home/ser/src/ser-0.10.x/modules/usrloc/usrloc.so" loadmodule "/home/ser/src/ser-0.10.x/modules/registrar/registrar.so" loadmodule "/home/ser/src/ser-0.10.x/modules/auth/auth.so" loadmodule "/home/ser/src/ser-0.10.x/modules/auth_db/auth_db.so" loadmodule "/home/ser/src/ser-0.10.x/modules/uri/uri.so" loadmodule "/home/ser/src/ser-0.10.x/modules/uri_db/uri_db.so" loadmodule "/home/ser/src/ser-0.10.x/modules/domain/domain.so" loadmodule "/home/ser/src/ser-0.10.x/modules/mediaproxy/mediaproxy.so" loadmodule "/home/ser/src/ser-0.10.x/modules/nathelper/nathelper.so" loadmodule "/home/ser/src/ser-0.10.x/modules/textops/textops.so" loadmodule "/home/ser/src/ser-0.10.x/modules/avpops/avpops.so" loadmodule "/home/ser/src/ser-0.10.x/modules/xlog/xlog.so" Are those functions not present in ser 0.10.99 any more? Bests Tomasz

1 0

[Users] 2 Instances of OpenSER on the same server
by Gerardo Amaya 08 Jan '07

08 Jan '07

Hello All. I'm trying to install 2 Instances of openser on the same machine with the following interfaces: Instance #1 -----> 172.16.18.2 and 192.168.1.112 Instance #2 -----> 172.16.1.2 and 192.168.1.113 I made one installation on the default OpenSER folders and one in the /usr/loca/openser_1/ The configuration went quite smooth. The first instance of the server is running with no problem at all, but when I try to start the second instance I got an error that it was unable to resolve the IP addresses I add the following lines to the /etc/hosts file: ### OpenSER 1st interface configuration #### 192.168.1.112 openser1_internal 172.16.18.2 openser1_external ### OpenSER 1st Interface configuration #### ### OpenSER 2nd Interface configuration #### 192.168.1.113 openser1_internal 172.16.1.2 openser1_external ### OpenSER 2nd Interface configuration #### After that Now I have the following error: Jan 5 17:26:30 atel-router ./openser[17772]: ERROR: tcp_init: bind(7, 0x813e194, 16) on 192.168.1.112:5060 : Address already in use Any Help will be appreciated, of better is there a How-to on OpenSER multiple instances please let me know. Thanks in advance Gerardo Amaya Here is the configuration file for each instance: OpenSER.cfg Instance #1 # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.1.112 listen=172.16.18.2 port=5060 mhomed=1 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/textops.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper and mediaproxy params -- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # -- End of nathelper and mediaproxy params -- # -- auth_db database params for domain.so module to work properly -- modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://root:codevoz@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") # -- End of database params -- # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); return; }; if (method == "REGISTER" || ! search("^Record-Route:")) { fix_nated_contact(); # Rewrite contact with source IP of signalling #if (method == "INVITE") { # fix_nated_sdp("1"); # Add direction=active to SDP #}; force_rport(); # Add rport parameter to topmost Via } if (method == "INVITE") { if (dst_ip == 192.168.1.112){ if (force_rtp_proxy("FAII")) t_on_reply("1"); } else if (dst_ip == 172.16.18.2){ if (force_rtp_proxy("FAEI")) t_on_reply("1"); } else if (dst_ip == 192.168.1.112){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); } else if (dst_ip == 172.16.18.2){ if (force_rtp_proxy("FAEE")) t_on_reply("1"); } }; # Record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # This is the Call Tear Down Section if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; route(1); } route[1] { t_on_reply("1"); #force_rtp_proxy(); t_relay("udp:192.168.1.2:5060"); } onreply_route[1] { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; if (nat_uac_test("1")) { fix_nated_contact(); }; } OpenSER.cfg Instance #2 # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=udp:192.168.1.113:5060 listen=udp:172.16.1.2:5060 #port=5060 mhomed=1 children=4 fifo="/tmp/ser_fifo2" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/openser_1/lib/openser/modules/sl.so" loadmodule "/usr/local/openser_1/lib/openser/modules/tm.so" loadmodule "/usr/local/openser_1/lib/openser/modules/rr.so" loadmodule "/usr/local/openser_1/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/openser_1/lib/openser/modules/usrloc.so" loadmodule "/usr/local/openser_1/lib/openser/modules/registrar.so" loadmodule "/usr/local/openser_1/lib/openser/modules/textops.so" loadmodule "/usr/local/openser_1/lib/openser/modules/uri.so"loadmodule "/usr/local/openser_1/lib/openser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/openser_1/lib/openser/modules/auth.so" loadmodule "/usr/local/openser_1/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper and mediaproxy params -- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy_1.sock") # -- End of nathelper and mediaproxy params -- # -- auth_db database params for domain.so module to work properly -- modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://root:codevoz@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") # -- End of database params -- # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); return; }; if (method == "REGISTER" || ! search("^Record-Route:")) { fix_nated_contact(); # Rewrite contact with source IP of signalling #if (method == "INVITE") { # fix_nated_sdp("1"); # Add direction=active to SDP #}; force_rport(); # Add rport parameter to topmost Via } if (method == "INVITE") { if (dst_ip == 192.168.1.113){ if (force_rtp_proxy("FAII")) t_on_reply("1"); } else if (dst_ip == 172.16.1.2){ if (force_rtp_proxy("FAEI")) t_on_reply("1"); } else if (dst_ip == 192.168.1.113){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); t_on_reply("1"); } else if (dst_ip == 192.168.1.113){ if (force_rtp_proxy("FAIE")) t_on_reply("1"); } else if (dst_ip == 172.16.1.2){ if (force_rtp_proxy("FAEE")) t_on_reply("1"); } }; # Record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # This is the Call Tear Down Section if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; route(1); } route[1] { t_on_reply("1"); #force_rtp_proxy(); t_relay("udp:192.168.1.2:5060"); } onreply_route[1] { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; if (nat_uac_test("1")) { fix_nated_contact(); }; } loadmodule "/usr/local/openser_1/lib/openser/modules/uri_db.so" loadmodule "/usr/local/openser_1/lib/openser/modules/domain.so" loadmodule "/usr/local/openser_1/lib/openser/modules/mediaproxy.so" ifconfig output eth0 Link encap:Ethernet HWaddr 00:0E:2E:0A:61:68 inet addr:172.16.18.2 Bcast:172.16.18.255 Mask:255.255.255.0 inet6 addr: fe80::20e:2eff:fe0a:6168/64 Scope:Link UP BROADCAST MULTICAST MTU:1500 Metric:1 RX packets:216465 errors:0 dropped:0 overruns:0 frame:0 TX packets:323925 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:18190685 (17.3 MiB) TX bytes:118579358 (113.0 MiB) Interrupt:185 Base address:0xaf00 eth1 Link encap:Ethernet HWaddr 00:0E:2E:06:EC:17 inet addr:172.16.1.2 Bcast:172.16.1.255 Mask:255.255.255.0 UP BROADCAST MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:0 (0.0 b) TX bytes:0 (0.0 b) Interrupt:209 Base address:0xce00 eth2 Link encap:Ethernet HWaddr 00:0C:76:BE:E5:87 inet addr:192.168.1.112 Bcast:192.168.1.255 Mask:255.255.255.0 inet6 addr: fe80::20c:76ff:febe:e587/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1155321 errors:0 dropped:0 overruns:0 frame:0 TX packets:664343 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:187622098 (178.9 MiB) TX bytes:125573054 (119.7 MiB) Interrupt:217 Base address:0xed00 eth2:0 Link encap:Ethernet HWaddr 00:0C:76:BE:E5:87 inet addr:192.168.1.113 Bcast:192.168.1.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 Interrupt:217 Base address:0xed00

3 3

[Users] Surefire Failure
by Bill Neely 08 Jan '07

08 Jan '07

Does anybody have a way to force an unconditional branch to a failure route. Or, if not that, a command that will surely fail and force a branch to the failure route. Bill -- 702-874-3833 1-866-553-3833

2 3

[Users] problem forwarding ACKs when using IPv6
by Emil Ivov 08 Jan '07

08 Jan '07

Hello all, We are currently performing a number of interoperability tests between OpenSER and SIP Communicator over IPv6. The good news is that we have successfully established IPv6 calls between two SIP Communicator instances registered on our local OpenSER installation (version 1.1.0-7 on Debian testing). There is however something weird we'd appreciate any ideas on what might be causing it: When running OpenSER with a default configuration, any ACKs that the caller tries to proxy through OpenSER don't get forwarded to the callee. Changing syn_branch to "no" (we saw somewhere that this might be related) didn't have any effect. I am attaching a log snippet that seems to be corresponding to the ACK processing. In a desperate attempt to resolve the issue we added the following to our route{} clause: if (method == "ACK") { forward("[2001:660:4701:f009:213:ceff:fea7:a88c]:5060"); exit; } where [2001:660:4701:f009:213:ceff:fea7:a88c] is the IPv6 address of the callee. This worked but it is obviously not (even close to) a "viable" solution. So our question is, what would be the best way to make OpenSER forward ACKs that a caller is trying to send to a callee? Thanks Emil 2(11235) receive_msg: cleaning up 4(11237) SIP Request: 4(11237) method: <ACK> 4(11237) uri: <sip:emcho@ipv6.sip-communicator.org> 4(11237) version: <SIP/2.0> 4(11237) parse_headers: flags=2 4(11237) Found param type 232, <branch> = <z9hG4bKc01a084c923dbc4a5de11530d611fe64>; state=16 4(11237) end of header reached, state=5 4(11237) parse_headers: Via found, flags=2 4(11237) parse_headers: this is the first via 4(11237) After parse_msg... 4(11237) preparing to run routing scripts... 4(11237) DEBUG : sl_filter_ACK: to late to be a local ACK! 4(11237) parse_headers: flags=100 4(11237) get_hdr_field: cseq <CSeq>: <1> <ACK> 4(11237) DEBUG: add_param: tag=b9b9c012 4(11237) DEBUG:parse_to:end of header reached, state=29 4(11237) DBUG:parse_to: display={"emcho"}, ruri={sip:emcho@ipv6.sip-communicator.org} 4(11237) DEBUG: get_hdr_field: <To> [60]; uri=[sip:emcho@ipv6.sip-communicator.org] 4(11237) DEBUG: to body ["emcho" <sip:emcho@ipv6.sip-communicator.org>] 4(11237) DEBUG:maxfwd:is_maxfwd_present: value = 70 4(11237) DEBUG: add_param: tag=b7cb05bc 4(11237) DEBUG:parse_to:end of header reached, state=29 4(11237) DBUG:parse_to: display={"77"}, ruri={sip:77@ipv6.sip-communicator.org} 4(11237) parse_headers: flags=200 4(11237) is_preloaded: No 4(11237) grep_sock_info - checking if host==us: 25==25 && [ipv6.sip-communicator.org] == [ipv6.sip-communicator.org] 4(11237) grep_sock_info - checking if port 5060 matches port 5060 4(11237) after_strict: Next hop: 'sip:[2001:660:4701:1001:20E:7FFF:FEB5:AAA3];ftag=b7cb05bc;lr=on' is loose router 4(11237) parse_headers: flags=ffffffffffffffff 4(11237) DEBUG: get_hdr_body : content_length=0 4(11237) found end of header 4(11237) rewrite_uri: Rewriting Request-URI with 'sip:[2001:660:4701:1001:20E:7FFF:FEB5:AAA3];ftag=b7cb05bc;lr=on' 4(11237) after_strict: The last route URI: 'sip:[2001:660:4701:1001:20E:7FFF:FEB5:AAA3];ftag=b7cb05bc;lr=on' 4(11237) parse_headers: flags=ffffffffffffffff 4(11237) DEBUG: t_newtran: msg id=3 , global msg id=2 , T on entrance=0xffffffff 4(11237) parse_headers: flags=ffffffffffffffff 4(11237) parse_headers: flags=78 4(11237) t_lookup_request: start searching: hash=42196, isACK=1 4(11237) parse_headers: flags=38 4(11237) DEBUG: t_lookup_request: e2e proxy ACK found 4(11237) DEBUG:tm:REF_UNSAFE: after is 1 4(11237) DEBUG:tm:t_newtran: building branch for end2end ACK 4(11237) DEBUG:tm:UNREF_UNSAFE: after is 0 4(11237) DEBUG:tm:t_relay: forwarding ACK 4(11237) DEBUG: mk_proxy: doing DNS lookup... 4(11237) check_via_address(2001:660:4701:F009:215:FF:FE35:EE1F, 0.0.0.0, 0) 4(11237) Sending: ACK sip:[2001:660:4701:1001:20E:7FFF:FEB5:AAA3];ftag=b7cb05bc;lr=on SIP/2.0 Record-Route: <sip:[2001:660:4701:1001:20E:7FFF:FEB5:AAA3];lr=on;ftag=b7cb05bc> Via: SIP/2.0/UDP [2001:660:4701:1001:20E:7FFF:FEB5:AAA3];branch=z9hG4bK4d4a.9db19213.2 Via: SIP/2.0/UDP 0.0.0.0;received=2001:660:4701:F009:215:FF:FE35:EE1F;branch=z9hG4bKc01a084c923dbc4a5de11530d611fe64 CSeq: 1 ACK Call-ID: 39bdf32f765ff3a4b67d1eedc2d538c9(a)0.0.0.0 From: "77" <sip:77@ipv6.sip-communicator.org>;tag=b7cb05bc To: "emcho" <sip:emcho@ipv6.sip-communicator.org>;tag=b9b9c012 User-Agent: SIP Communicator 1.0 CVS-Fri_Jan_05_18-27-08_EET_2007 Max-Forwards: 69 Content-Length: 0 P-hint: rr-enforced . 4(11237) orig. len=490, new_len=679, proto=1 4(11237) DEBUG:destroy_avp_list: destroying list (nil)

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[Users] RTPProxy and SER, Multiple Interface Server
by Gerardo Amaya 08 Jan '07

08 Jan '07

Hello All. After investigating quite a while about this issue I'm kind of desperate now, I will appreciate your help on this. I have a OpenSER Server with multiple interfaces, I'm able to handle all the SIP packets with the multi homed enabled. Now the part that I'm having real trouble with is the RTP audio streams. After struggling with MediaProxy I finally started using RTPProxy for this, since it has bridge mode available. Now the problem is that I have more than one network to connect audio from, so the internal and external configuration is not enough for me. So my questions are: 1.-Can I have RTPProxy Bridge mode working for all the interfaces in my server? 2.-Can I have more than one RTPProxy socket available for each of the multiple interfaces I'm connecting in order to have the audio working, so each route will use the proxy of their own network? 3.-If #2 is true, is this where the force_socket and force_rtp_proxy functions play a role in my configuration? Please help!! Thanks you so much in advance Gerardo Amaya

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[Users] Re: [Devel] openserctl user addition problem
by Klaus Darilion 08 Jan '07

08 Jan '07

Rajesh Kalagarla wrote: > Hi Klaus, > > Thanks for your response. > now i am trying to add user using openserctl only. > but when i am giving a command to add the user " command: openserctl add > 1122 earth rajeshk(a)idnltd.com" > > output is, ERROR 1045(28000): Access denied for user openser@localhost > (using PASSWORD: YES) > > the query inside the openserctl looks to be failing. bingo. you are right. you have to configure openserctl to use the proper username/password. take a look at /etc/openser/openserctlrc regards klaus > > Thanks, > Rajesh > > ----- Original Message ----- > From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at> > To: "Rajesh Kalagarla" <RajeshK(a)idnltd.com> > Cc: <devel(a)openser.org> > Sent: Thursday, January 04, 2007 1:29 PM > Subject: Re: [Devel] openserctl user addition problem > > >> This has nothing todo with your openser.cfg. openserctl needs to know >> the mysql password for the openser mysql user. >> >> IIRC it must be configured somewhere in /etc/openser >> >> regards >> klaus >> >> >> Rajesh Kalagarla wrote: >>> Hello All, >>> >>> I am trying to setup the OpenSER on linux system and the distribution is > RHL. >>> How can we add multiple users credentials as part of configuration file? > i am not able to authorize more than one User Agent thru the config file. >>> when i am adding the second user info, it is simply overwriting the old > info and not authorizing the first user. >>> >>> modparam("auth_db","user_column" "1122") >>> modparam("auth_db", "password_column", "pluto") >>> >>> modparam("auth_db","user_column" "3344") >>> modparam("auth_db", "password_column", "earth") >>> >>> with this info, i am able to register only the second user.if i comment > the second user info, first user was able to register. please suggest how to > handle this, >>> when i am trying to add the user using the "openserctl" i am getting > permission denied error. >>> command: openserctl add 1122 earth rajeshk(a)idnltd.com >>> >>> output is, ERROR 1045(28000): Access denied for user openser@localhost > (using PASSWORD: YES) >>> >>> any help greatly appreciated, >>> >>> Thanks, >>> Rajesh >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Devel mailing list >>> Devel(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/devel >> >> -- >> Klaus Darilion >> nic.at >> >> > > -- Klaus Darilion nic.at

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[Serusers] ser 0.96 # mediaproxy # one-way audio SOLVED!!!
by Gareth Owen 08 Jan '07

08 Jan '07

Hi Davor I notice you had a problem with one way audio. I have the same problem but when the X-Lite client is behind firewall. Do you have any ideas or direction I could follow? Best regards Gareth Gareth Owen The information contained in this email message is intended only for the use of the individual(s) or entity to whom it is addressed, and may contain information that is confidential and may be legally privileged and exempt from disclosure under applicable laws. If the reader of this message is not the intended recipient or the employee or agent responsible to deliver it to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error please immediately notify the sender and destroy the original message from your system. The recipient should check this email and any attachments for the presence of viruses. BeVoice accepts no liability for any damage caused by any virus transmitted by this email

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[Serusers] Anyone like to finish my SER install for a fee?
by Bart Fisher 07 Jan '07

07 Jan '07

Hello all, I have SER 0.9.6 installed, along with Mediaproxy 1.8.0, Nathelper, SEMS 0.9.6, SerWeb 0.9.6 set up as a PSTN gateway on CentOS. However, there are some issues with NAT, Accounting (ACC) and the way I setup the web access. Also, I'd like to get to Voice mail and Conference working. Basically, I'd want you to take what's there and finish the install so everything works together. For an expert, all this is probably a few hours job. There is already SSH access in to the system. I started with a "LiveCD - VOIP" version of SER (http://www.wifi.com.ar/english/voip.html) with all the above (almost) but found it grossly out-dated and the Author unwilling to assist nor update to current releases. FYI: If you are are thinking of purchasing this software, save you money! The only thing nice is there is a prepaid billing script that might be converted to current releases. I just need to 'jump-start' this project to some usable level. I'm willing to pay for your time. If you are interested, please contact me at bart (at) icpage (dot) com. TIA Bart

1 0

[Serusers] SER 0.9.6 and Media-Server Asterisk 1.4
by Stefan Tichy 07 Jan '07

07 Jan '07

Hi all, SER 0.9.6 is used as registrar/proxy and asterisk 1.4 as media server (voicemail, ...) The situation when phone A calls phone B and B is busy is handled by a failure route revert_uri(); rewritehostport("10.11.12.13:5070"); append_branch(); t_relay; This works fine, if for example phone A is some Snom 190. If another asterisk server is used instead of phone A it works if "pedantic=no" is set in the sip.conf of asterisk. If it is "pedantic=yes", then asterisk compares the tags in the To header fields. There is one tag from phone B and a second from the asterisk media server. Asterisk ignores every response not having the tag used by phone B. IMHO asterisk is wrong, but since I can't controll which asterisk servers are used as clients - is there a way to may asterisk happy? Would it be possible to remove tags from 1XX responses? -- Stefan Tichy <seruser(a)pi4tel.de>

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[Serusers] how many calls can SER support?
by zahra bahar 06 Jan '07

06 Jan '07

Hi. do you know how many calls can ser support? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Start SER with xlite
by zahra bahar 06 Jan '07

06 Jan '07

Hi. I installed SER as a proxy server and Xlite as a softphone.but xlite can't make a call. it sends :408 or 404 .is it any trouble with ser?or xlite?how can I find out ? what changes in ser.cfg are needed for having a pc to pc call with xlite? in ser.cfg: should fork=no or yes? is it necessary to use mysql module? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] proxy and redirect servers
by zahra bahar 06 Jan '07

06 Jan '07

Hi. Could any one say me when we use proxy server and when it is necessary to use redirect server? thanks. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] TM : BYE retransmissions
by Jean-François SMIGIELSKI 05 Jan '07

05 Jan '07

Hello list! I observed, on an openser-1.1.0, that stateful relayings of BYE requests absorb retransmissions from the sender but do not retransmit it to the next hop in the Route header. Please, is it normal or is ti a misconfiguration from myself ? Thanks a lot! JF Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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Re: [Serusers] aliases
by Nils Ohlmeier 05 Jan '07

05 Jan '07

As this might be interesting for other users I cc'ed the list again. Just to be sure: you know that their is no alias table any more, right? Alias or not is just a matter of the flag value in the uri table. In general there is no need any more for re-writting a request which was targeted towards an alias. Request towards an alias are handled exactly the same way as "normal" requests, because both will get the same UID value. If you used the alias table for re-distributing the requests outside of your domain, which is actually more a forwarding function then an alias, then you could have this function as one (out of many) possible alternative(s) with AVPs like this: (Note: non-tested pseudo code) lookup_user("$t.uid", "ruri"); ... if ($t.uid) { load_attrs("$tu", "$t.uid"); if ($t.forwarding_target) { attr2uri("$t.forwarding_target"); t_relay(); } } Greetings Nils On Friday 05 January 2007 17:27, you wrote: > Thanks for the reply, Nils! I suppose I am a more of a newbie than I first > appeared as your answer is a little over my head. What I am trying to do > with the new version of SER is what used to be done (in previous versions) > with > > If (lookup("aliases")){ > Route(1); > > To redirect users based on a few entries in the aliases table. I won't be > using the SER server as a registrar or to handle dynamic user locations or > even authenticated accounts -- it will be open to the public without > authentication to access SIP accessible numbers and users. Mostly what I > want to do is just to route calls that SER receives (like john(a)iptn.org, > 5555555555(a)iptn.org etc.) to the domain/uri that used to be accessed in the > aliases table (openser worked this way too). > > Sorry for not explaining better and thanks again for your help! > > Eliott > > > -----Original Message----- > From: Nils Ohlmeier [mailto:nils@iptel.org] > Sent: Friday, January 05, 2007 8:32 AM > To: serusers(a)lists.iptel.org > Cc: Eliott Spencer > Subject: Re: [Serusers] aliases > > Hello, > > On Friday 05 January 2007 04:27, Eliott Spencer wrote: > > I am coming over from OpenSER because I ran some tests and found SER to > > be much faster. There are some things that are new to me that I imagine > > are easy to do. I have a working config with the 12/22/06 development > > snapshot > > > of 0.10 but I would like to lookup aliases to redirect inbound calls to > > another domain. I filled in the domain table and the uri table with a > > user > > > and I looked through the sample config but can't seem to get it to work. > > I > > > have also loaded the domain module, then the uri module, then the avp and > > avpops modules (among others). > > > > > > > > Any idea what I might need to put in my route? This doesn't seem to work > > (stolen from ser.cfg.sample): > > > > > > > > if (lookup_user("$t.uid", "@ruri")) { > > > > setflag(1); > > > > route(1); > > > > break; > > > > }; > > from looking at your code snipset I'm not sure what you really want to > achieve. > In general you should just do the lookup_user and then examine the AVPs > afterwards. I think that it is easier then looking at the result code of > the > > lookup_user function. > If you really want to distingiush if the call was targeted to an alias or > the > non-alias URI of an user then check for the AVP ruri_canonical. If it is > set > > (to the value 1) the URI contained the canonical (=non-alias) URI of the > user. > If you just want to know if the URI is a know one and belongs to a user, > just > check for the presence of the uid AVP after the lookup_user call. If it is > present the user is know to the system under the value of the uid AVP. > Otherwise the user/URI is not known to the system. > > Hope this helps > Nils

1 0

[Users] forking call to another user on same proxy
by Christian Schlatter 05 Jan '07

05 Jan '07

I try to fork a call to another user registered at the same proxy as the caller. I figured out that the easiest method would be to use the append_branch() function, something like route[0] { if (method=="REGISTER") { save("location"); exit; } if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; } # if someone calls test2@domain, fork call to test4@domain # (test2 and test4 are both registered at domain) if (avp_check("$rU", "eq/test2")) { append_branch("sip:test4@127.0.0.1"); } if (!t_relay()) { sl_reply_error(); }; exit; } I know that this could also be achieved by adding permanent location records, but I'd like to load the forking targets from a DB in the future. So my first question would be if using append_branch("sip:test4@127.0.0.1"); is the correct way to do parallel forking to a local user. I'm asking because this method does not work for me as expected. It seems that the two UAC state machines running on the same proxy host that are involved in a forking call do somehow not work well together. Forking the INVITE to both test2 and test4 works well, but when I pick up test2 the call gets terminated after a few seconds. Further investigations have revealed that canceling the second call leg does not work correctly. There is a message flow diagram available at http://www.unc.edu/~cschlatt/openser.html . For some reason, the proxy process sending from 127.0.0.1 is retransmitting the 487 "Request Terminated" although it receives ACKs from the upstream proxy. And this causes the upstream proxy to send a 487 back to the caller (time 15.360), which then terminates the call established between caller and test2. And this only happens if I fork to the same proxy, if I use two proxy hosts everything works fine (??). Any advice is highly appreciated, Christian

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[Users] integrating with Asterisk and OpenSER for Voicemail
by raviprakash sunkara 05 Jan '07

05 Jan '07

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24) modparam("tm","wt_timer",1) #mrodparam("tm", "ruri_matching", 0) #modparam("tm", "via1_matching", 0) modparam("avpops","avp_url","mysql://root:passwd@192.168.2.75/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops","avp_aliases","inv=i:15") ............... route { ...... if (loose_route()) { t_relay(); exit; }; if(is_method("INVITE") && !has_totag()) { xdbg("user [$ruri] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri","$avp(inv)"); setflag(2); }; .............. .... route(1); } route[1] { if(isflagset(2)) { t_on_failure("1"); }; } failure_route[1] { log("----------------------------------------- \n"); if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "$avp(inv)"); prefix("9"); # route to Asterisk Media Server rewritehostport("192.168.2.75:5060"); append_branch(); t_relay("192.168.2.75:5060"); resetflag(2); } ......................... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Users] openser+freeradius:(radius_is_user_in(): Failure)+(DEBUG:avpops:ops_c)
by buptzhaoyuan＠sina.com 05 Jan '07

05 Jan '07

Hi, Lists,happy new year! Probably a stupid question,but I can not solved it correctly.The ua can not register the openser normally.I think the 5 step runs well and the last is down.The debug shows two errors. The first error is " 0(5898) radius_is_user_in(): Failure" ;the other is " 0(5898) DEBUG:avpops:ops_check_avp: no src avp found".So please give me help and some suggestion. And the fellowing is my configure file and debug infomation.Thanks!! HTTP/SIP RADIUS +-----+ (1) +-----+ +-----+ | |==========>| | | | | | (2) | | | | | |<==========| | | | | | (3) | | | | | |==========>| | | | | A | | B | (4) | C | | | | |---------->| | | | | | (5) | | | | | |<----------| | | | (6) | | | | | |<==========| | | | +-----+ +-----+ +-----+ ====> HTTP/SIP ----> RADIUS I setup the system by using the document(http://www.openser.org/docs/openser-radius-1.0.x.html) and my openser is openser-1.1.0-tls, the file openser.cfg is the same as the document. # #$Id$ # # radius config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=udp:192.168.168.3 alias="swifton.org" alias="192.168.168.3" #fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- [root@localhost openser]# cat openser.cfg # #$Id$ # # radius config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=udp:192.168.168.3 alias="swifton.org" alias="192.168.168.3" #fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "avpops.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "uri.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc","db_url","mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- modparam("avpops", "avp_aliases", "day=i:101;time=i:102") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # check if user is suspended if(is_method("REGISTER|INVITE|MESSAGE|OPTIONS|SUBSCRIBE")) { if (radius_is_user_in("From", "suspended")) { sl_send_reply("403", "Forbidden - suspended"); exit; }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); } route(1); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; if (!uri==myself) { # check if user is allowed to do voip calls to other domains if(is_method("INVITE|MESSAGE")) { if (!radius_is_user_in("From", "voip")) { sl_send_reply("403", "Forbidden VoIP"); exit; }; }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { # authenticate registers if (method=="REGISTER") { if (!radius_www_authorize("swifton.org")) { www_challenge("swifton.org", "0"); exit; }; # check the src ip address if(!avp_check("$avp(i:2)", "eq/$src_ip/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; # calls to pstn if(uri=~"sip:00[1-9][0-9]+@") { if(is_method("INVITE") && !has_totag()) { if (!radius_is_user_in("From", "pstn")) { sl_send_reply("403", "Forbidden PSTN"); exit; }; }; # set gateway address rewritehostport("10.10.10.10:5090"); route(1); }; # load callee's avps if(avp_load_radius("callee")) { # check if user has time filter enabled if(avp_check("$avp(i:3)", "eq/i:1")) { # print time in an avp avp_printf("$avp(i:100)", "$Tf"); # extract day avp_subst("$avp(i:100)/$avp(i:101)", "/(.{3}) .+/*\1*/"); if(!avp_check("$avp(i:6)", "fm/$day")) { sl_send_reply("403", "Forbidden - day"); exit; }; # extract 'hours:minutes' avp_subst("$avp(i:100)/$avp(i:102)", "/(.{10}) (.{5}):.+/\2/"); if((is_avp_set("$avp(i:4)") && avp_check("$avp(i:4)", "gt/$time")) || (is_avp_set("$avp(i:5)") && avp_check("$avp(i:5)", "lt/$time"))) { sl_send_reply("403", "Forbidden - time"); exit; }; }; }; ...... route(1); } ...... exit; } # and the openser start as follows: 898) SIP Request: 898) method: <REGISTER> 898) uri: <sip:swifton.org> 898) version: <SIP/2.0> 898) parse_headers: flags=2 898) end of header reached, state=5 898) parse_headers: Via found, flags=2 898) parse_headers: this is the first via 898) After parse_msg... 898) preparing to run routing scripts... 898) parse_headers: flags=100 898) DEBUG:maxfwd:is_maxfwd_present: value = 70 898) parse_headers: flags=10 898) DEBUG: add_param: tag=f59c388b5de348778e76ebda993c2d8e 898) DEBUG: add_param: epid=b6757bf3fc 898) DEBUG:parse_to:end of header reached, state=29 898) DBUG:parse_to: display={}, ruri={sip:zhaoy@swifton.org} 898) radius_is_user_in(): Failure 898) parse_headers: flags=200 898) DEBUG:parse_to:end of header reached, state=10 ...... 0(5898) grep_sock_info - checking if host==us: 11==13 && [swifton.org] == [192.168.168.3] 0(5898) grep_sock_info - checking if port 5060 matches port 5060 0(5898) grep_sock_info - checking if host==us: 11==13 && [swifton.org] == [192.168.168.3] 0(5898) grep_sock_info - checking if port 5060 matches port 5060 0(5898) check_nonce(): comparing [459bb612ad569015987d4849d1ae603515f32e6b] and [459bb612ad569015987d4849d1ae603515f32e6b] 0(5898) DEBUG:auth_radius:radius_authorize_sterman: Success 0(5898) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 0(5898) DEBUG:auth_radius:extract_avp: string is <Sip-Group:voip> 0(5898) DEBUG:auth_radius:extract_avp: AVP name is <Sip-Group> 0(5898) DEBUG:auth_radius:extract_avp: AVP val is <voip> 0(5898) DEBUG:auth_radius:generate_avps: AVP 'Sip-Group'/0='voip'/0 has been added 0(5898) DEBUG:avpops:ops_check_avp: no src avp found 0(5898) parse_headers: flags=ffffffffffffffff 0(5898) check_via_address(192.168.168.30, 192.168.168.30, 0) 0(5898) DEBUG:destroy_avp_list: destroying list 0x422bc440 0(5898) receive_msg: cleaning up the freeradius debug infomation as follows: er-Name = "zhaoy(a)swifton.org" Digest-Attributes = 0x0a077a68616f79 Digest-Attributes = 0x010d73776966746f6e2e6f7267 Digest-Attributes = 0x022a34353962623631326164353639303135393837643438343964316165363033353135663332653662 Digest-Attributes = 0x04117369703a73776966746f6e2e6f7267 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "2ead4f5b9d78013a35f2eb7a792971a5" Service-Type = Sip-Session Sip-Uri-User = "zhaoy" NAS-Port = 5060 NAS-IP-Address = 192.168.168.3 Processing the authorize section of radiusd.confmodcall: entering group authorize for request 167 modcall[authorize]: module "preprocess" returns ok for request 167 modcall[authorize]: module "chap" returns noop for request 167 modcall[authorize]: module "mschap" returns noop for request 167 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "zhaoy" Digest-Realm = "swifton.org" Digest-Nonce = "459bb612ad569015987d4849d1ae603515f32e6b" Digest-URI = "sip:swifton.org" Digest-Method = "REGISTER"rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 167 ....... Processing the authenticate section of radiusd.confmodcall: entering group authenticate for request 167A1 = zhaoy:swifton.org:zhaoyA2 = REGISTER:sip:swifton.orgH(A1) = eff77a9105c5c0973492694f59c944f2H(A2) = d3add39dad2709bea55016ba79eb675aKD = eff77a9105c5c0973492694f59c944f2:459bb612ad569015987d4849d1ae603515f32e6b:d3add39dad2709bea55016ba79eb675aEXPECTED 2ead4f5b9d78013a35f2eb7a792971a5RECEIVED 2ead4f5b9d78013a35f2eb7a792971a5 modcall[authenticate]: module "digest" returns ok for request 167modcall: group authenticate returns ok for request 167radius_xlat: 'Authenticated'Login OK: [zhaoy(a)swifton.org] (from client openser port 5060)Sending Access-Accept of id 13 to 192.168.168.3:33101 SIP-AVP = "Sip-Group:voip" Reply-Message = "Authenticated" Sip-Group = "voip"Finished request 167 and my MSN:buptzhaoyuan@sina.com .I am nice to communicate with you! ------------------------------------------------------------------- 惠普商用台式机dc5750,高性能，低功耗！( http://ad4.sina.com.cn/sina/limeng3/mail_zhuiyu/2006/mail_zhuiyu_20061225.h… ) =================================================================== 注册新浪2G免费邮箱（ http://mail.sina.com.cn/chooseMode.html ）

2 1

[Users] Reg:Instance messages
by venkid 05 Jan '07

05 Jan '07

hai all, i am using X-lite as sip client .i have installed in window and am able to send instance messages and presence info is also fine with openser In redhat Linux i am trying to send instance messages using X-lite , but i am not able send and not able to receive any messages . is X-lite linux version is support for redhat linux Es 3.0 or any other sip client is there for redhat linux Es 3.0 Thanks and Regards venkatesh.d

2 1

[Serusers] aliases
by Eliott Spencer 05 Jan '07

05 Jan '07

I am coming over from OpenSER because I ran some tests and found SER to be much faster. There are some things that are new to me that I imagine are easy to do. I have a working config with the 12/22/06 development snapshot of 0.10 but I would like to lookup aliases to redirect inbound calls to another domain. I filled in the domain table and the uri table with a user and I looked through the sample config but can't seem to get it to work. I have also loaded the domain module, then the uri module, then the avp and avpops modules (among others). Any idea what I might need to put in my route? This doesn't seem to work (stolen from ser.cfg.sample): if (lookup_user("$t.uid", "@ruri")) { setflag(1); route(1); break; }; Thanks!

2 1

[Serusers] PA module segfault
by TERRY ROBERT 05 Jan '07

05 Jan '07

Hi, I'm getting a segfault error while running the latest presence server snapshot (ser-0.10.99-dev35-pa-4.2). The scenario is: UA SER SUBSCRIBE ----------> 200 <---------- NOTIFY <---------- 200 ----------> Everything is fine until the subscription expires, at which point the PA module segfaults and generates a core file. Any suggestions on this? (logfile and backtrace from gdb are attached).

2 1

Re: [Users] LCR db_only mode problem
by Jayesh Nambiar 05 Jan '07

05 Jan '07

Hi Juha, Thanks for your reply. First prominent doubt that I have is; when I am using LCR in db_only mode why does openser try to load the table while starting up? The scenario is as follows: I had 50 rows in LCR table and started openser and it started properly and also routed the calls properly according to the values in lcr and gw table. Now I just appended more 40,000 rows to this table and again the routing worked properly as per my lcr table. This behaviour was correct as I was using query only mode. I did not need any restart or any fifo reload command. Only when I restarted my openser, the openser again died as it was trying load the lcr table while starting. Could this be a bug in lcr module as I am not sure whats happening. I tried increasing MAX_NO_OF_LCRS but even that did not help. Somebody please help if poosible. Thanks, w/regards, jayesh ----- Original Message ---- From: Juha Heinanen <jh(a)tutpro.com> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Tuesday, 26 December, 2006 10:58:01 PM Subject: [Users] LCR db_only mode problem X-VM-v5-Data: ([nil nil nil nil nil nil nil nil nil] the error messages you get are coming from this statement: if (lcr_dbf.query(dbh, NULL, 0, NULL, lcr_cols, 0, 4, 0, &res) < 0) { LOG(L_ERR, "lcr_reload_gws(): Failed to query lcr data\n"); lcr_dbf.close(dbh); return -1; } i don't know which kind of memory db queries reserve. if shared memory, you can start openser with larger -m parameter value. if pkg memory, you need to edit PKG_MEM_POOL_SIZE in config.h and recompile openser. with lots of lcr entries and no caching, performance may be bad. if you use caching then you need to increase value of #define MAX_NO_OF_LCRS 256 in lcr_mod.c and recompile lcr module. increasing MAX_NO_OF_LCRS means that more shared memory will be used so you may also need to start openser with larger -m parameter value. -- juha Send free SMS to your Friends on Mobile from your Yahoo! Messenger. Download Now! http://messenger.yahoo.com/download.php

3 3

[Serusers] Aastra 9133i
by Bill 04 Jan '07

04 Jan '07

Does anyone have a config for an Aastra 9133i IP phone to a SER box? I am trying to configure one. I can make calls to the Aastra but can't make calls from the Aastra. It registers properly. Thanks Bill

1 0

[Users] new CDRTool release 5.0.5
by Adrian Georgescu 04 Jan '07

04 Jan '07

Hello, I have been asked many times where is the best place to obtain support and ask questions for CDRTool. As CDTool is at this moment primarily designed and tested for OpenSER deployments I have decided that OpenSER users mailing is the best place to make announcements for new releases. There is of course no guaranteed support, as you may expect for a free product, support requests will however not go unheard. Unless there is no objection or complaint, I will keep this mailing list updated about CDRTool new releases from now on. Here it is: CDRTool 5.0.5 has been released. The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog: * Fixed determination of previous year in update_raddact_record_mediaproxy Consolidate SET statements to minimize binary log entries. Fixed bug in update_raddact_record_mediaproxy, the record values were reset after execution of update on the first table causing failure to update the previous table. setup/radius/OpenSER/radius_accounting.proc must be reloaded into the MySQL server: mysql -u root -p -h sipdb radius < setup/radius/OpenSER/ radius_accounting.proc * Use same CDR structure for MaxSessionTime and DebitBalance * Change logging for prepaid actions to fit one line * Normalize tries previous month if no record has been updated * Mention how to see prepaid account status * Updated rating docs * Moved E164 class to phplib/local.inc * Show accounts with quota that exceeded a certain treshold scripts/SER/quotaShowAccounts.php treshhold * Added documentation for the Quota system (doc/QuotaSystem.txt) * Use normalization lock per table using GET_LOCK() server function http://dev.mysql.com/doc/refman/5.0/en/miscellaneous-functions.html This provided faster web response when multiple clients access the interface * Fixed confirmation for delete operations in rating tables when global vars are turned off in php.ini Regards, Adrian

2 1

[Users] Strange OK Message after Register
by Martin Klisch 04 Jan '07

04 Jan '07

Hi, yesterday i've began to upgrade openser from 1.0.1 to 1.1.0. Now i got a strange situation when registering with two different clients with the same sip account. in the OK Message from the sip proxy i have a strange contact field: Contact: <sip:012345@xxx:5070>;q=0.9;expires=3600, <sip:012345@xxxx:63036>;expires=25 i think in the OK Message should only be one contact adress?

2 1

[Users] Forward to the "To" field
by Thomas Deillon 04 Jan '07

04 Jan '07

Hi all, Merry Xmas ! I'm just a small question if some people are still working :-) I try to make a load balancing solution with SER and Asterisk. I use the dispatcher module and I set on asterisks servers the default gw as SER. All requests go through the SER to the asterisks and on the SER I'm have a line like If source-ip = Asterisk source IP do ... And I want to do something like forward to the TO FIELD ... Do you know how can I do this ? Thanks a lot for your help, Thomas

3 2

[Users] Secure SIP messages
by Steffen Witt 04 Jan '07

04 Jan '07

Hello, I have a few questions about how to secure SIP messages... IPSec or TLS can be used to guarantee privacy, integrity and authenticity of transmitted data. But SIP is only payload/data for these protocols. Digest Authentication does only guarantee authenticity... S/MIME seems to be interesting but I don't know any implementation in a SIP client. My questions: - Is there any mechanism that prevents me from corrupting or faking SIP messages? - Is it possible to create a kind of binding between the certificates used for TLS/IPSec and the SIP accounts? Thanks for your answers... Best regards, Steffen

2 3

[Serusers] HELP!!! Table acc is full
by Nhadie Ramos 04 Jan '07

04 Jan '07

Hi All, How can i solve this problem? is this a mysql problem? anyone has encountered this error before? Regards, Ronald

1 0

[Users] European and Australian based SIP termination providers
by T.R. Missner 04 Jan '07

04 Jan '07

Hello, Does anyone have recommendations for a SIP based termination provider in Europe and/or Australia? I am looking specifically for a wholesale, carrier grade provider. I have found several which appear to be asterisk based and quite small. What I am looking for is more of a traditional carrier that is SIP enabled as opposed to a smallish mom and pop shop. Thanks in advance. T.R.

3 2

[Serusers] Ser, MediaProxy and Asterisk
by Gerardo Amaya 04 Jan '07

04 Jan '07

Hello all, I'm trying to use a SER server for the following: | | | | ISP #1 ISP #2 Private Link Private LINK NATed SIP SIP Clients 172.18.xxx.yyy 172.16.5.xx Clients | | | | | | | | | | | ---------------------------------------------------------------------------------------------------- | ---------------------------------------------------------------- SER with MEDIAPROXY ---------------------------------------------------------------- | | | ================ Asterisk (172.16.5.1) ================= | | PSTN I need asterisk to handle all the calls to the PSTN and event the SIP calls between them. I need to use the SER server as a tool to receive calls from any of the environments that I draw on top, and all communicating with Asterisk, even I also have a backup Asterisk box, that I can use in load balance with SER. I saw that there is a function "rewritehostport()" which I'm pretty sure it won't work since I will not keep the call in the SER server. I have made the NAT configuration of the Getting Started Guide, but I now that this is just part of what I need to build the complete solution. Can anyone please give me a hint in what is that I need to acomplish such task Thanks in advance Gerardo Amaya

2 1

[Serusers] strange problem ..... /tmp/mysql.sock and /var/lib/mysql/mysql.sock
by Tako Chang 04 Jan '07

04 Jan '07

Hi, all I install mysql server and SER in my FC6 when I run SER with mysql to do authenticate #ser -f auth-mysql.cfg (the SER getting started example) it prompt me that can't connect to local MySQL server through socket 'tmp/mysql.sock' so I modified the /etc/my.cnf' file to change socket=/tmp/mysql.sock but when i start SER and use serctl to add a user ERROR 2002 (HY000): Can't connect to local MySQL server through socke 'var/lib/mysql/mysql.sock' was prompted I don't know if I could open two sock at the same time or how can I let SER or serctr chage the path to fit another. Did someone can show me some clue or solution? thanks!!

2 1

[Serusers] One way audio
by Gareth Owen 04 Jan '07

04 Jan '07

Can someone help? I am fairly new to SER and Mediaproxy and can't find a detailed explanation of how nathelper etc work, so I'm struggling! My config is:- SER 0.9.6 Mediaproxy 1.7.2 Fedora Core 4 86x64 SER and Mediaproxy run on the same server which acts as a gateway to a remote Sonus PSTN gateway over a VPN. I also use Asterisk on my local network. The SER server acts as a router between Asterisk servers and the PSTN via the Sonus. This works fine but my problem is when I attempt to use X-Lite 3.0 UA behind NAT on a remote network. Asterisk is not used in this config but proves that two way audio is fine using SER and the Sonus gateway. Problem: X-Lite behind NAT on remote network to Sonus PSTN gateway via SER and Mediaproxy, I get only one ay audio. The audio is fine from X-Lite to PSTN but not in the reverse direction. Thanks for your help

2 1

[Users] Issues with calls using openser.
by Shane Burrell 04 Jan '07

04 Jan '07

I recently installed the latest version of openser and this time used mediaproxy rather than rtpproxy. Everything seems to work but if a sip device is called the phone rings and is instantally disconnected and the far end is left off-hook. This worked before but I did modify my script to work with mediaproxy. Below is the wireshark decode of the sip messagining. Any help or suggestions on where to look would be great. Calls from the sip device works flawlessly. I am using a MaxTNT as the gateway. |Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 | |22.031 | INVITE SDP ( telephone-event) | |SIP From: sip:8385021101@152.53.16.91:5060 To:sip: 8385024200@ siprt1.me.net:5060 | |(5060) ------------------> (5060) | | |22.031 | 100 Giving a try | |SIP Status | |(5060) <------------------ (5060) | | |22.031 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |22.040 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |22.042 | | 180 Ringing |SIP Status | | |(5060) <------------------ (5060) | |22.042 | 180 Ringing | |SIP Status | |(5060) <------------------ (5060) | | |25.244 | | 200 OK SDP ( telephone-event) |SIP Status | | |(5060) <------------------ (5060) | |25.245 | 200 OK SDP ( telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |25.269 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |25.269 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |25.269 | INVITE SDP ( telephone-event) | |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@ siprt1.me.net:5060 | |(5060) ------------------> (5060) | | |25.270 | 407 Proxy Authentication Required | |SIP Status | |(5060) <------------------ (5060) | | |25.291 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |25.291 | BYE | | |SIP Request | |(5060) ------------------> (5060) | | |25.293 | | BYE | |SIP Request | | |(5060) ------------------> (5060) | |25.326 | | 200 OK | |SIP Status | | |(5060) <------------------ (5060) | |25.327 | 200 OK | | |SIP Status | |(5060) <------------------ (5060) | | Shane

2 6

[Users] How relay/retransmit subscribe, notify or publish messages...
by Marvin Ottley 03 Jan '07

03 Jan '07

Good day all, Currently I'm working on an interface/design where I have openser relaying messages to asterisk servers. The stumbling block I've run into is having openser relay SUBSCRIBE, NOTIFY or PUBLISH messages to Asterisk. So far, I've tried using openser's forward() and t_relay() functions without success; By the way, these work just fine with INVITE and REGISGER messages. Any pointers would be greatly appreciated Thanks, Marvin Ottley Email: marvino(a)nbsvoice.com <mailto:marvino@nbsvoice.com> This email may contain confidential and/or privileged information, and is intended solely for the addressee identified above. Any unauthorized use, disclosure or dissemination of the information contained in this email is strictly prohibited. If you have received this email in error, then please notify the sender immediately, and delete this email and all copies.

2 1

[Users] several presence question
by Klaus Darilion 03 Jan '07

03 Jan '07

Hi! I read the docs of the presence module and have some question. Maybe someone can enlighten me ;-) 1. If I understand correctly, the pua module acts as a "UAC". Thus, it can be used to send PUBLISH or SUBSCRIBE. I wonder what is the usecase for sending SUBSCRIBE? Usually, a client which sends a SUBSCRIBE receives subsequent NOTIFY messages. How can these NOTIFY messages be handled? 2. If an external application sends PUBLISH via the uac_mi interface, how can be distinguished between a "new" PUBLISH or an "update" PUBLISH? For example if the MI interface returns the E-Tag from the 200 Ok response this could be used in the next request to group identify PUBLISH reqeusts which belong together. 3. There is a bug in the pua module docs: the db_url(str) parameter is described twice. 4. The following module documentations lack of default values for the module parameters: pua, pua_usrloc, presence (maybe some example show the default value, but IMO the default value should be explicitely stated) 5. How can I update the presence status from a third party application? Is it possible to write directly into the tables of the presence modules? Or can I fake an update using the pua_mi to loop a PUBLISH to itself for handling it afterwards with handle_publish()? May this cause problems because it interferes the subsequent PUBLISH request from the client (etag)? regards Klaus

3 9

RE: [Users] SER as proxy in front of Asterisk
by Thomas Deillon 03 Jan '07

03 Jan '07

Hi, In fact, now, every phone does not response. The options didn't go thought the FW, so, the phone cannot answer. I don't understand why the traffic don't go thought it !! I have test this configuration with two firewall: a sonicwall and a soho. Cheers, Thomas -----Message d'origine----- De : Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Envoyé : mercredi, 3. janvier 2007 14:52 À : Thomas Deillon Cc : users(a)openser.org Objet : Re: [Users] SER as proxy in front of Asterisk Hello, what is the type of NAT/Firewall you have in front of Thomson? As I can see from the diagram, the phone does not reply to OPTIONS request, and I guess, the nat/firewall needs outward traffic to keep the pinhole open. Check why thomson ignores the options requests. Cheers, Daniel On 01/03/07 15:17, Thomas Deillon wrote: > > Hi, > > I have made more tries and now, all my phones don't work ... > > The firewall opens a connection to REGISTER phone but after a while > the OPTION message from the Asterisk will be DROP by the FW ... > > You can find a diagram of the exchange here: > http://deillon.eu/tmp/exchange_sip.png > > A list of all exchange: http://deillon.eu/tmp/ip_sip.txt > > And all sip message: http://deillon.eu/tmp/message.txt > > Thanks a lot for your help, > > Thomas > > ------------------------------------------------------------------------ > > *De :* users-bounces(a)openser.org [mailto:users-bounces@openser.org] > *De la part de* Thomas Deillon > *Envoyé :* mardi, 2. janvier 2007 11:44 > *À :* users(a)openser.org > *Objet :* [Users] SER as proxy in front of Asterisk > > Hi all, > > I wish you first a happy new year !! > One again, I ask you some help. Thanks a lot for your patience and > your answers that really helped me. > > So, I want to put a SER in front of one Asterisk for the moment (more > after). > All Servers have public IP address and so, I don't care about the > RTP/SDP messages. > All phones are behind NAT somewhere in Switerland :) > > > I have the configuration below in openser: > > > modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") > #modparam("dispatcher", "force_dst", 1) > modparam("dispatcher", "flags", 2) > > modparam("usrloc", "db_mode", 0) > #modparam("rr", "enable_full_lr", 1) > > route{ > > > xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > }; > > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > > # if (search("User-Agent:.*Thomson.*")) { > # }; > > > if ((src_ip==212.xxx.xxx.152) || (src_ip==212.xxx.xxx.153)) { > if(method=="OPTIONS") { > }; > avp_pushto("$ru","$tu"); > forward(); > > }else { > fix_nated_contact(); > force_rport(); > if(method=="REGISTER"){ > ds_select_dst("4", "0"); > t_relay(); > } > else { > ds_select_dst("0", "4"); > forward(); > } > } > } > > > And in my dispatcher.list I have: > > 0 sip:212.xxx.xxx.153:5060 > 4 sip:212.xxx.xxx.153:5060 > > > > With this configuration, a snom phone, a X-lite phone and a Cisco > phone seams to work .... > But with the Thomson ST2030, it doesn't work. > In fact, the Thomson REGISTER on asterisk, the Asterisk send a OPTION > message to the FW IP adress and the right port where the thomson is > and then, after a while, the FW close the connection. The thomson > phone is so "UNREACHABLE" on asterisk status and nobody can call it. > I'm not sure that my configuration is ok in ser but I think that is a > problem with the Thomson2030. Do you had the same kind of problem or > do you understand the problem here ? > > > Thanks for your help, > > Thomas Deillon > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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RE: [Users] SER as proxy in front of Asterisk
by Thomas Deillon 03 Jan '07

03 Jan '07

Hi, I have made more tries and now, all my phones don't work ... The firewall opens a connection to REGISTER phone but after a while the OPTION message from the Asterisk will be DROP by the FW ... You can find a diagram of the exchange here: http://deillon.eu/tmp/exchange_sip.png A list of all exchange: http://deillon.eu/tmp/ip_sip.txt And all sip message: http://deillon.eu/tmp/message.txt Thanks a lot for your help, Thomas ________________________________ De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Thomas Deillon Envoyé : mardi, 2. janvier 2007 11:44 À : users(a)openser.org Objet : [Users] SER as proxy in front of Asterisk Hi all, I wish you first a happy new year !! One again, I ask you some help. Thanks a lot for your patience and your answers that really helped me. So, I want to put a SER in front of one Asterisk for the moment (more after). All Servers have public IP address and so, I don't care about the RTP/SDP messages. All phones are behind NAT somewhere in Switerland :) I have the configuration below in openser: modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") #modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 2) modparam("usrloc", "db_mode", 0) #modparam("rr", "enable_full_lr", 1) route{ xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # if (search("User-Agent:.*Thomson.*")) { # }; if ((src_ip==212.xxx.xxx.152) || (src_ip==212.xxx.xxx.153)) { if(method=="OPTIONS") { }; avp_pushto("$ru","$tu"); forward(); }else { fix_nated_contact(); force_rport(); if(method=="REGISTER"){ ds_select_dst("4", "0"); t_relay(); } else { ds_select_dst("0", "4"); forward(); } } } And in my dispatcher.list I have: 0 sip:212.xxx.xxx.153:5060 4 sip:212.xxx.xxx.153:5060 With this configuration, a snom phone, a X-lite phone and a Cisco phone seams to work .... But with the Thomson ST2030, it doesn't work. In fact, the Thomson REGISTER on asterisk, the Asterisk send a OPTION message to the FW IP adress and the right port where the thomson is and then, after a while, the FW close the connection. The thomson phone is so "UNREACHABLE" on asterisk status and nobody can call it. I'm not sure that my configuration is ok in ser but I think that is a problem with the Thomson2030. Do you had the same kind of problem or do you understand the problem here ? Thanks for your help, Thomas Deillon

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[Users] 1.1.0 Compiling for Solaris 10
by Martin Klisch 03 Jan '07

03 Jan '07

Hi, i can not compile openser 1.1.0 for solaris 10. any idea why? gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4.3"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -c cfg.tab.c -o cfg.tab.o /usr/ccs/bin/yaccpar:5: warning: ignoring #pragma ident cfg.y: In function `yyparse': cfg.y:859: error: `IPTOS_MINCOST' undeclared (first use in this function) cfg.y:859: error: (Each undeclared identifier is reported only once cfg.y:859: error: for each function it appears in.) /usr/ccs/bin/yaccpar:164: warning: label `yynewstate' defined but not used /usr/ccs/bin/yaccpar:374: warning: label `yyerrlab' defined but not used gmake: *** [cfg.tab.o] Error 1

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