sr-users May 2008

sr-users@lists.kamailio.org

121 participants
202 discussions

[OpenSER-Users] auth_radius:radius_authorize_sterman: rc_auth failed

by Simon J Xuereb

Hi, I am trying to configure openser + freeradius for authentication & accounting. I have my freeradius configured with openldap and it's working successfully. tested with radclient -f digest 127.0.0.1:1812 auth testing123 results are successfull. However I cannot get it to work with openser. I see nothing happening in my "/usr/sbin/radius -X" output coming from openser when i try to register. My enviroment was tested on Fedora 8 and Fedora 9 both with rpm's and same results. auth_radius:radius_authorize_sterman: rc_auth failed no output is seen in my /usr/sbin/radius -X Below please find my openser.cfg & radiusclient.conf & servers ######## radiusclient.conf # General settings # specify which authentication comes first respectively which # authentication is used. possible values are: "radius" and "local". # if you specify "radius,local" then the RADIUS server is asked # first then the local one. if only one keyword is specified only # this server is asked. auth_order radius,local # maximum login tries a user has login_tries 4 # timeout for all login tries # if this time is exceeded the user is kicked out login_timeout 60 # name of the nologin file which when it exists disables logins. # it may be extended by the ttyname which will result in # a terminal specific lock (e.g. /etc/nologin.ttyS2 will disable # logins on /dev/ttyS2) nologin /etc/nologin # name of the issue file. it's only display when no username is passed # on the radlogin command line issue /etc/radiusclient-ng/issue # RADIUS settings # RADIUS server to use for authentication requests. this config # item can appear more then one time. if multiple servers are # defined they are tried in a round robin fashion if one # server is not answering. # optionally you can specify a the port number on which is remote # RADIUS listens separated by a colon from the hostname. if # no port is specified /etc/services is consulted of the radius # service. if this fails also a compiled in default is used. authserver 127.0.0.1 authserver 10.0.0.10 authserver localhost # RADIUS server to use for accouting requests. All that I # said for authserver applies, too. # acctserver 127.0.0.1:1813 # file holding shared secrets used for the communication # between the RADIUS client and server servers /etc/radiusclient-ng/servers # dictionary of allowed attributes and values # just like in the normal RADIUS distributions #dictionary /usr/share/radiusclient-ng/dictionary dictionary /etc/radiusclient-ng/dictionary.openser # program to call for a RADIUS authenticated login login_radius /usr/sbin/login.radius # file which holds sequence number for communication with the # RADIUS server seqfile /var/run/radius.seq # file which specifies mapping between ttyname and NAS-Port attribute mapfile /etc/radiusclient-ng/port-id-map # default authentication realm to append to all usernames if no # realm was explicitly specified by the user # the radiusd directly form Livingston doesnt use any realms, so leave # it blank then default_realm # time to wait for a reply from the RADIUS server radius_timeout 10 # resend request this many times before trying the next server radius_retries 3 # local address from which radius packets have to be sent bindaddr * # LOCAL settings # program to execute for local login # it must support the -f flag for preauthenticated login login_local /bin/login ######################################################## #/etc/radiusclient-ng/servers #Server Name or Client/Server pair Key #---------------- --------------- #portmaster.elemental.net hardlyasecret #portmaster2.elemental.net donttellanyone localhost testing123 127.0.0.1 testing123 10.0.0.10 testing123 ####################################################### ##########openser.cfg # -- rr params -- debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 port=5060 # ------------------ module loading ---------------------------------- mpath="/usr/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 "auth.so" loadmodule "auth_radius.so" loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc","db_url","mysql://openser:openser@localhost/openser") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("auth_radius","radius_config","/etc/radiusclient-ng/radiusclient.conf") # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # authenticate registers if (method=="REGISTER") { if (!radius_www_authorize("")) { www_challenge("", "0"); exit; }; save("location"); exit; }; route(1); } # generic forward route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } # Thanks for your help SJX

16 years, 6 months

Re: [OpenSER-Users] problem - 484 address incomplete for the correct TO NUMBER

by Kaeman Chris

Like i already took this as a benefit of doubt. They used G711u for a California calls and it was successful and used G711u NSE to the NewYork and the call failed with 484. so i asked them to use the G711u for NewYork and not G711u NSE still the call failed with G711u (excluding NSE). I enquired with the provider already that they support G711u but not G711u NSE. but still with G711u they failed and that too with 484. thats really wierd and i am stuck up in the middle. KC LetMeKnow <sunkara.raviprakash.feb14(a)gmail.com> wrote: Hi Chris, This Problem is happened to me, when i used EyeBeam---> OpenSER--> Asterisk as PSTN Gateway, Asterisk is seems to send 484, this is due to the imcompartable of the Codecs in Eyebean and Asterisk gateway........................................... Check out.... logs from these customer signaling ip-addr - 23.21.76.23 customer media ip-addr - 231.43.25.1 supplier media ip-addr - 43.25.32.85 supplier signaling ip-addr - 13.45.213.245 my signaling ip-addr - 123.123.1.234 supplier have producing this t SIP headers and found no issues. Also, when i use the same called number and make the calls through the same provider it works from my end with 200 OK. its really interesting and seems to be some issues with the headers. I am using openser and following are the logs. sorry - its too long, but i need someones help to figure out whats going on really. Also, following is the definition for the ips and telephone numbers. FROM number - 17592034561 TO number - 19846572302 customer signaling ip-addr - 23.21.76.23 customer media ip-addr - 231.43.25.1 supplier media ip-addr - 43.25.32.85 supplier signaling ip-addr - 13.45.213.245 my signaling ip-addr - 123.123.1.234 --------------------------------------------------------------------- logs as below; Y#HE& _ d E pl@ 7 ! @ V)ÇèNÃ Ä Ä |MàINVITE sip:+17592034561@123.123.1.234 SIP/2.0 Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#H¯' ¨ ¨ 0H% E èÌ@ @ £ ÇèNÃ@ V) Ä Ä ¯ SIP/2.0 100 Trying To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H ; ® ® 0H% E èÍ@ @ ¢ÿÇèNÃ@ V) Ä Ä ¯ SIP/2.0 100 Giving a try To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H; M M 0H% E =èÎ@ @ G;ÇèNÃ@®N Ä Ä ) ÑINVITE sip:17592034561@13.45.213.245 SIP/2.0 Record-Route: <sip:123.123.1.234;lr=on;ftag=5d411c09;osp-o=t5198095097179410177_s23.21.76.23_a1210276292_c1> Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 68 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#HÖº Ê Ê _ d E º** 2 Ws@®NÇèNÃ Ä Ä ¦ SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#HY Ö Ö _ d E Æ+ 2 V@®NÇèNÃ Ä Ä ² SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#H v v 0H% E fè×@ @ K ÇèNÃ@®N Ä Ä R úACK sip:17592034561@13.45.213.245 SIP/2.0 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Call-ID: 5d1bb15abd63917a To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a CSeq: 2 ACK User-Agent: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 --------------------------------- Download prohibited? No problem. CHAT from any browser, without download. _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead 91-9999882776 --------------------------------- Check out the all-new face of Yahoo! India. Click here. --------------------------------- Check out the all-new face of Yahoo! India. Click here.

16 years, 6 months

[OpenSER-Users] Implementing international destination allow/deny system with PDT module (patch included for 1.3.X)

by Iñaki Baz Castillo

Hi, I've implemented a system to control international destinations (allow/deny) using "PDT" module (based on prefix). In OpenSer 1.4 there will be a new module "userblacklist" which can handle this issue in a more ellegant way, but for OpenSer 1.3.X we can use "pdt" module (with a little patch to pdt.c) to implement it. I've submitted the patch and the how-to: https://sourceforge.net/tracker/index.php?func=detail&aid=1965211&group_id=… Hope it could be useful. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

16 years, 6 months

[OpenSER-Users] Res: sermyadmin almost works

by Flavio Goncalves

Please, check the updated instructions to install at: www.sermyadmin.org Flavio ----- Mensagem original ---- De: whudson05 <whudson05(a)gmail.com> Para: users(a)lists.openser.org Enviadas: Quarta-feira, 14 de Maio de 2008 11:29:26 Assunto: Re: [OpenSER-Users] sermyadmin almost works I'm also at the same stage. I've looked at the mySQL queries and they appear to be correct. Have you added the user admin to subscriber? Lucas Catranis wrote: > > Anyone Get this setup correctly? I've been at it for a while and after > getting it to insert new users I cannot login as an admin. I'm not sure > where the problem is but Any help would be appreciated. > -- View this message in context: http://www.nabble.com/sermyadmin-almost-works-tp17219053p17232461.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

16 years, 6 months

[OpenSER-Users] problem - 484 address incomplete for the correct TO NUMBER

by Kaeman Chris

Hello, I am having the following problem. please help me to figure out the issue. I have my client calling me and getting 484 Address incomplete. I checked all the SIP headers and found no issues. Also, when i use the same called number and make the calls through the same provider it works from my end with 200 OK. its really interesting and seems to be some issues with the headers. I am using openser and following are the logs. sorry - its too long, but i need someones help to figure out whats going on really. Also, following is the definition for the ips and telephone numbers. FROM number - 17592034561 TO number - 19846572302 customer signaling ip-addr - 23.21.76.23 customer media ip-addr - 231.43.25.1 supplier media ip-addr - 43.25.32.85 supplier signaling ip-addr - 13.45.213.245 my signaling ip-addr - 123.123.1.234 --------------------------------------------------------------------- logs as below; Y#HE& _d E pl@ 7!@V)ÇèNÃÄÄ|MàINVITE sip:+17592034561@123.123.1.234 SIP/2.0 Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#H¯' ¨ ¨ 0H% EèÌ@ @£ÇèNÃ@V)ÄÄ¯SIP/2.0 100 Trying To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H; ® ® 0H% EèÍ@ @¢ÿÇèNÃ@V)ÄÄ¯ SIP/2.0 100 Giving a try To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H; M M 0H% E=èÎ@ @G;ÇèNÃ@®NÄÄ) ÑINVITE sip:17592034561@13.45.213.245 SIP/2.0 Record-Route: <sip:123.123.1.234;lr=on;ftag=5d411c09;osp-o=t5198095097179410177_s23.21.76.23_a1210276292_c1> Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 68 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#HÖº Ê Ê _d E º** 2Ws@®NÇèNÃÄÄ¦ SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#HY Ö Ö _d E Æ+ 2V@®NÇèNÃÄÄ² SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#H v v 0H% Efè×@ @K ÇèNÃ@®NÄÄRúACK sip:17592034561@13.45.213.245 SIP/2.0 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Call-ID: 5d1bb15abd63917a To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a CSeq: 2 ACK User-Agent: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 --------------------------------- Download prohibited? No problem. CHAT from any browser, without download.

16 years, 6 months

[OpenSER-Users] OpenSER 1.3.2 is released

by Bogdan-Andrei Iancu

Hi, I'm happy to announce that OpenSER 1.3.2 is out - after almost 6 months from the first 1.3 release, 1.3.2 is available and it includes valuable experiences and feedback from operating 1.3.x in live platform. This means more stability, more reliability. This is a minor release, so it includes only fixes. That means that configuration and DB structure was not changes and it is fully compatible with all the other releases from 1.3 branch. A comprehensive compilation of fixes (thanks to Henning Westerholt) is available via the Changelog - see http://www.openser.org/pub/openser/1.3.2/ChangeLog For more about how to download this version, see http://www.openser.org/mos/view/News/NewsItem/OpenSER-1.3.2-is-released/ If people want to contribute with packages generated on different OS, please let me know - I will be glad to upload them on the repository. Regards, Bogdan

16 years, 6 months

[OpenSER-Users] [RFC] - siptrace change

by Bogdan-Andrei Iancu

Hi, I would like to get your opinion/comments on some siptrace issue. In 1.3, a new capability was added in siptrace - to trace the ACK resulted from sending stateless replies. The problem I discovered with this is that it cannot be controlled by selecting the messages or transactions to be be traced. Usually you use siptrace to trace only certain transactions / messages - tracing all traffic is usually not a realistic option. So, with the stateless ACK, the siptrace module cannot apply the selection from script and trace them all, with no filtering at all. So, if you want to trace a traffic for a single user, all ACK going through the platform will be traced. This reduces the usability of the module and you get a lot of garbage tracing. My suggestion will be to disable stateless ACK tracing until a way to control/filter it is found. I'm asking this considering the upcoming 1.3.2 release from tomorrow. Regards, Bogdan

16 years, 6 months

Re: [OpenSER-Users] problem in audio when using media proxy

by Krunal Patel

Hi, Actually I am new to openser. Here i am sending trace from openser. So please check out SDP. *************************************************************************************************************************************** Message Body: INVITE sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com>SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK7F77064E8D8FD3FD41779B0F67EF5C05 From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>> Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28079 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 291 v=0 o=5555555555 3961416264 3961416447 IN IP4 122.xxx.xxx.35 s=X-Lite c=IN IP4 122.xxx.xxx.35 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=291 0(14396) found end of header 0(14396) find_first_route: No Route headers found 0(14396) loose_route: There is no Route HF 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48297, isACK=0 0(14396) DEBUG: RFC3261 transaction matching failed 0(14396) DEBUG: t_lookup_request: no transaction found 0(14396) DBG: trans=0xb60e0960, callback type 1, id 0 entered 0(14396) ##########################method=INVITE################################## 0(14396) ##############################client_nat_test(7)############################## 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) DEBUG:check_content_type: type <application/sdp> found valid 0(14396) parse_headers: flags=80 0(14396) parse_headers: flags=10000 0(14396) pre_auth(): Credentials with given realm not found 0(14396) build_auth_hf(): 'Proxy-Authenticate: Digest realm=" voip.domain.com", nonce="482c8213b23f69ac2b268e593620cb2367dc643b" ' 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) WARNING: script writer didn't release transaction 0(14396) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(14396) DEBUG: add_to_tail_of_timer[2]: 0xb60e09a8 (10704) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <ACK> 0(14396) uri: <sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> > 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK7F77064E8D8FD3FD41779B0F67EF5C05>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) parse_headers: flags=8 0(14396) DEBUG: add_param: tag=b802e73ab92fec121a4f880263d541a3.3a34 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [83]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <INVITE> 0(14396) uri: <sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> > 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) In request log route 0(14396) 0(14396) ############################## NEW REQUEST RECEIVED ############################## 0(14396) parse_headers: flags=10 0(14396) DEBUG: add_param: tag=2027736222 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={5555555555}, ruri={ sip:5555555555@voip.domain.com:5061} 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:28:55 2008]: INVITE sip:5555555555@voip.domain.com:5061 -> sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> : 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:28:55 2008]: Request: INVITE sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> 0(14396) From: sip:5555555555@voip.domain.com:5061;tag=2027736222 0(14396) parse_headers: flags=8 0(14396) DEBUG:parse_to:end of header reached, state=10 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [41]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> > ] 0(14396) To: sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> ;tag=<null> 0(14396) SIP message-id: 201 0(14396) parse_headers: flags=40 0(14396) Call-id: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 0(14396) parse_headers: flags=20 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) CSeq: 28080 0(14396) Contact: <sip:5555555555@122.xxx.xxx.35:5061> 0(14396) Message Body: INVITE sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com>SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>> Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE Proxy-Authorization: Digest username="5555555555",realm="voip.domain.com ",nonce="482c8213b23f69ac2b268e593620cb2367dc643b",response="7a7d3cf46a5b7cf90cc623664ca52e4d",uri=" sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 291 v=0 o=5555555555 3961416264 3961416447 IN IP4 122.xxx.xxx.35 s=X-Lite c=IN IP4 122.xxx.xxx.35 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=291 0(14396) found end of header 0(14396) find_first_route: No Route headers found 0(14396) loose_route: There is no Route HF 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48303, isACK=0 0(14396) DEBUG: RFC3261 transaction matching failed 0(14396) DEBUG: t_lookup_request: no transaction found 0(14396) DBG: trans=0xb60df1c8, callback type 1, id 0 entered 0(14396) ##########################method=INVITE################################## 0(14396) ##############################client_nat_test(7)############################## 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) DEBUG:check_content_type: type <application/sdp> found valid 0(14396) parse_headers: flags=80 0(14396) check_nonce(): comparing [482c8213b23f69ac2b268e593620cb2367dc643b] and [482c8213b23f69ac2b268e593620cb2367dc643b] 0(14396) DEBUG:auth_radius:radius_authorize_sterman: Success 0(14396) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 0(14396) ERROR:avp_radius:avp_load_user: rc_auth failed 0(14396) DEBUG:usrloc:db_load_urecord: aor not found in DB 0(14396) lookup(): '999999999(a)voip.domain.com' Not found in usrloc 0(14396) grep_sock_info - checking if host==us: 23==23 && [voip.domain.com] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) alias_db_lookup: no alias found for R-URI 0(14396) does_uri_exit(): User in request uri does not exist 0(14396) parse_headers: flags=78 0(14396) ------[sip:5555555555@voip.domain.com:5061] ---checkeding---- 0(14396) radius_is_user_in(): Success 0(14396) -------- media route 5 ---- 0(14396) ########################## isbflagset(6) isbflagset(7)#################################### 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=8000000 0(14396) DEBUG:lcr:load_gws: add matched_gws[0]=[0,0] 0(14396) load_gws(): DEBUG: Added gw_uri_avp <sip:|0@213.XXX.XXX.28 :5060;transport=udp> 0(14396) -----------------lcr is loaded into the ------ 0(14396) load_gws(): DEBUG: Added ruri_user_avp <999999999> 0(14396) ------------- Gateway is loaded --------- 0(14396) DEBUG: mk_proxy: doing DNS lookup... 0(14396) parse_headers: flags=2000 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) clen_builder: content-length: 313 (313) 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) DBG:check_against_rule_list: using list dns 0(14396) DEBUG:tm:set_timer: relative timeout is 500000 0(14396) DEBUG: add_to_tail_of_timer[4]: 0xb60df314 (10701100000) 0(14396) DEBUG:tm:set_timer: relative timeout is 20 0(14396) DEBUG: add_to_tail_of_timer[0]: 0xb60df330 (10720) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <100> 0(14396) reason: <Trying> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=0, new_code=100 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=-1 0(14396) DEBUG:tm:set_timer: relative timeout is 50 0(14396) DEBUG: add_to_tail_of_timer[1]: 0xb60df330 (10750) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <183> 0(14396) reason: <Session Progress> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=0, uac[0]=100 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=0, new_code=183 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(14396) old size: 884, new size: 822 0(14396) build_res_from_sip_res: copied size: orig:318, new: 256, rest: 566 msg= SIP/2.0 183 Session Progress Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Content-Length: 209 Content-Type: application/sdp Contact: sip:999999999@213.XXX.XXX.28:5060 User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05 v=0 o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28 s=SIP Call c=IN IP4 213.XXX.XXX.28 t=0 0 m=audio 19462 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 0(14396) WARNING:vqm_resize: resize(0) called 0(14396) DEBUG:tm:relay_reply: sent buf=0x819a510: SIP/2.0 1..., shmem=0xb60e8218: SIP/2.0 1 0(14396) DEBUG:tm:set_timer: relative timeout is 50 0(14396) DEBUG: add_to_tail_of_timer[1]: 0xb60df330 (10750) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 1(14397) DEBUG: timer routine:4,tl=0xb60df314 next=(nil), timeout=10701100000 1(14397) DEBUG: timer routine:2,tl=0xb60e09a8 next=(nil), timeout=10704 1(14397) DEBUG: wait_handler : removing 0xb60e0960 from table 1(14397) DEBUG: delete transaction 0xb60e0960 1(14397) DEBUG: wait_handler : done 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <200> 0(14396) reason: <OK> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=183, uac[0]=183 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=183, new_code=200 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(14396) old size: 891, new size: 829 0(14396) build_res_from_sip_res: copied size: orig:304, new: 242, rest: 587 msg= SIP/2.0 200 OK Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Content-Length: 209 Content-Type: application/sdp Supported: replaces Contact: sip:999999999@213.XXX.XXX.28:5060 User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05 v=0 o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28 s=SIP Call c=IN IP4 213.XXX.XXX.28 t=0 0 m=audio 19462 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 0(14396) DEBUG: add_to_tail_of_timer[2]: 0xb60df210 (10710) 0(14396) DEBUG:tm:relay_reply: sent buf=0x819a510: SIP/2.0 2..., shmem=0xb60e05e0: SIP/2.0 2 0(14396) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <200> 0(14396) reason: <OK> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=200, uac[0]=200 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=200, new_code=200 0(14396) DEBUG:tm:t_should_relay_response: 200 OK for INVITE after final sent 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(14396) old size: 891, new size: 829 0(14396) build_res_from_sip_res: copied size: orig:304, new: 242, rest: 587 msg= SIP/2.0 200 OK Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Content-Length: 209 Content-Type: application/sdp Supported: replaces Contact: sip:999999999@213.XXX.XXX.28:5060 User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05 v=0 o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28 s=SIP Call c=IN IP4 213.XXX.XXX.28 t=0 0 m=audio 19462 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 0(14396) DEBUG:tm:relay_reply: sent buf=0x819a510: SIP/2.0 2..., shmem=0xb60e05e0: SIP/2.0 2 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <ACK> 0(14396) uri: <sip:999999999@213.XXX.XXX.28:5060> 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK384293560A7C9508E28212EC0D72CDFA>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(14396) In request log route 0(14396) 0(14396) ############################## NEW REQUEST RECEIVED ############################## 0(14396) parse_headers: flags=10 0(14396) DEBUG: add_param: tag=2027736222 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={5555555555}, ruri={ sip:5555555555@voip.domain.com:5061} 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: ACK sip:5555555555@voip.domain.com:5061 -> sip:999999999@213.XXX.XXX.28:5060: 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: Request: ACK sip:999999999@213.XXX.XXX.28:5060 0(14396) From: sip:5555555555@voip.domain.com:5061;tag=2027736222 0(14396) parse_headers: flags=8 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) To: sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> ;tag=5430ebc2c5d6575 0(14396) SIP message-id: 206 0(14396) parse_headers: flags=40 0(14396) Call-id: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 0(14396) parse_headers: flags=20 0(14396) get_hdr_field: cseq <CSeq>: <28080> <ACK> 0(14396) CSeq: 28080 0(14396) Contact: <sip:5555555555@122.xxx.xxx.35:5061> 0(14396) Message Body: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 70 Content-Length: 0 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) is_preloaded: No 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:check_self: host != me 0(14396) grep_sock_info - checking if host==us: 23==23 && [voip.domain.com] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) after_loose: Topmost route URI: 'sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on' is me 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=0 0(14396) found end of header 0(14396) find_next_route: No next Route HF found 0(14396) after_loose: No next URI found 0(14396) #################IN Loose_route############################ 0(14396) DEBUG: has_totag: totag found 0(14396) ############################has_totag() method==INVITE method==ACK################################ 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) ###########################setbflag6################################# 0(14396) ###############################force_rport############################# 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) #############################use_media_proxy############################### 0(14396) parse_headers: flags=78 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48303, isACK=1 0(14396) parse_headers: flags=38 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_lookup_request: e2e proxy ACK found 0(14396) DEBUG:tm:t_newtran: building branch for end2end ACK 0(14396) DEBUG:tm:t_relay: forwarding ACK 0(14396) DEBUG: mk_proxy: doing DNS lookup... 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) DBG:check_against_rule_list: using list dns 0(14396) DEBUG:forward_request: sending: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Via: SIP/2.0/UDP 195.XXX.XXX.51;branch=z9hG4bKfacb.bd19b862.2 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport=5061;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 69 Content-Length: 0 . 0(14396) DEBUG:forward_request: orig. len=519, new_len=593, proto=1 0(14396) -------------- cheked t reply --- 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list 0xb60e04c8 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <ACK> 0(14396) uri: <sip:999999999@213.XXX.XXX.28:5060> 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK384293560A7C9508E28212EC0D72CDFA>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(14396) In request log route 0(14396) 0(14396) ############################## NEW REQUEST RECEIVED ############################## 0(14396) parse_headers: flags=10 0(14396) DEBUG: add_param: tag=2027736222 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={5555555555}, ruri={ sip:5555555555@voip.domain.com:5061} 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: ACK sip:5555555555@voip.domain.com:5061 -> sip:999999999@213.XXX.XXX.28:5060: 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: Request: ACK sip:999999999@213.XXX.XXX.28:5060 0(14396) From: sip:5555555555@voip.domain.com:5061;tag=2027736222 0(14396) parse_headers: flags=8 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) To: sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> ;tag=5430ebc2c5d6575 0(14396) SIP message-id: 207 0(14396) parse_headers: flags=40 0(14396) Call-id: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 0(14396) parse_headers: flags=20 0(14396) get_hdr_field: cseq <CSeq>: <28080> <ACK> 0(14396) CSeq: 28080 0(14396) Contact: <sip:5555555555@122.xxx.xxx.35:5061> 0(14396) Message Body: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 70 Content-Length: 0 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) is_preloaded: No 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:check_self: host != me 0(14396) grep_sock_info - checking if host==us: 23==23 && [voip.domain.com] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) after_loose: Topmost route URI: 'sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on' is me 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=0 0(14396) found end of header 0(14396) find_next_route: No next Route HF found 0(14396) after_loose: No next URI found 0(14396) #################IN Loose_route############################ 0(14396) DEBUG: has_totag: totag found 0(14396) ############################has_totag() method==INVITE method==ACK################################ 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) ###########################setbflag6################################# 0(14396) ###############################force_rport############################# 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) #############################use_media_proxy############################### 0(14396) parse_headers: flags=78 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48303, isACK=1 0(14396) parse_headers: flags=38 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_lookup_request: e2e proxy ACK found 0(14396) DEBUG:tm:t_newtran: building branch for end2end ACK 0(14396) DEBUG:tm:t_relay: forwarding ACK 0(14396) DEBUG: mk_proxy: doing DNS lookup... 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) DBG:check_against_rule_list: using list dns 0(14396) DEBUG:forward_request: sending: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Via: SIP/2.0/UDP 195.XXX.XXX.51;branch=z9hG4bKfacb.bd19b862.2 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport=5061;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 69 Content-Length: 0 . 0(14396) DEBUG:forward_request: orig. len=519, new_len=593, proto=1 0(14396) -------------- cheked t reply --- 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list 0xb60e04c8 0(14396) receive_msg: cleaning up 1(14397) DEBUG: timer routine:2,tl=0xb60df210 next=(nil), timeout=10710 1(14397) DEBUG: wait_handler : removing 0xb60df1c8 from table 1(14397) DEBUG: delete transaction 0xb60df1c8 1(14397) DEBUG: wait_handler : done *************************************************************************************************************************************** -- Thanks & regards, Krunal Patel On Thu, May 15, 2008 at 12:03 AM, LetMeKnow < sunkara.raviprakash.feb14(a)gmail.com> wrote: > hey krunal, > i hopes ur fine and also all members of Ecosmob are fine, > > i have read ur mail, > yours problem is mention properly in mailing list. > > In ur openser script file some problem in there and u not handling the nat > in ur script. > make sure u have use both modules nathelper and mediaproxy, > > Before doing on this checkout is openser system is located inside the nat > or not. > > while test use ngrep tool on openser system, see whether u have got the > properly signaling happening not. > > check out after 200 ok of invite msg > > > On Wed, May 14, 2008 at 5:40 PM, Krunal Patel <krunal.patel(a)ecosmob.com> > wrote: > >> Hi, >> >> We are using openser with media proxy and radiator. We're having audio >> problems when we send calls to mediatrix gateway for termination. >> Internal calls work fine. We hear audio both side and we can see the >> session and bytes transfer properly in media proxy monitoring webpage. >> When we send calls to mediatrix gateway for termination, calls connect >> properly but we don't hear audio. We see inactive session in mediaproxy >> status page with no bytes transferring. >> One doubt I have is it is not entering reply route when it receives >> replies like 183, 180, etc. >> Has anyone encountered similar issue? How can we fix it? >> Thanks in advance! >> >> -- >> regards, >> Krunal Patel >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> > > > -- > Thanks &Regards > Ravi Prakash Sunkara > VoIP Development Tech Lead > 91-9999882776 >

16 years, 6 months

Re: [OpenSER-Users] [OpenSER-Devel] [RFC] - siptrace change

by Bogdan-Andrei Iancu

Hi Norman, yup, that is exactly my problem - I use it only for doing siptrace per user and on platform with high traffic, the stateless ACK tracing produces a lot of residual data in DB. Regards, Bogdan Norman Brandinger wrote: > This change would be good. We have reduced our use of siptrace for > this very reason. > > Norm > > Bogdan-Andrei Iancu wrote: >> Hi, >> >> I would like to get your opinion/comments on some siptrace issue. >> >> In 1.3, a new capability was added in siptrace - to trace the ACK >> resulted from sending stateless replies. >> >> The problem I discovered with this is that it cannot be controlled by >> selecting the messages or transactions to be be traced. Usually you >> use siptrace to trace only certain transactions / messages - tracing >> all traffic is usually not a realistic option. >> >> So, with the stateless ACK, the siptrace module cannot apply the >> selection from script and trace them all, with no filtering at all. >> So, if you want to trace a traffic for a single user, all ACK going >> through the platform will be traced. >> >> This reduces the usability of the module and you get a lot of garbage >> tracing. >> >> My suggestion will be to disable stateless ACK tracing until a way to >> control/filter it is found. I'm asking this considering the upcoming >> 1.3.2 release from tomorrow. >> >> Regards, >> Bogdan >> >> _______________________________________________ >> Devel mailing list >> Devel(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/devel >> >> >> > >

16 years, 6 months

[OpenSER-Users] parallel forking

by Allan Chao ( 趙廷睿 )

Dear All: I have found there are some devices can't process two 180 ringing after use OPENSER to do parallel forking, so, i have a idea that is used OPENSER to delete To tag in the To header before relay it , and it works well with eyebeam. But i can't find any spec to support my idea because it is against "early dialog" prograss and my question is how does UE(eyebeam or some client) to create dialog if there is no any To tag in 180 ringing but 200 OK has in the latter ? i think the UA will generate local ring tone but no any dialog after receives 180 ringing with no To tag? is it right ? thanks, allan DISCLAIMER: Sample Disclaimer added in a VBScript.

16 years, 6 months

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sr-users May 2008