sr-users April 2007

sr-users@lists.kamailio.org

177 participants
272 discussions

[Users] BYE message without Route header field

by rubel Ahmed

Hi, I am using Openser-1.2.0 with dialog module. When I make a call then full session is saved in dialog correctly and when bye received then the session will be free. but when the route header feild is not available in BYE message then the session is not removed at receive BYE, Because This BYE message is not caught bye dlg_onroute function. Now can I destroy this session in dlg_onreq function? Thanks BDRubel

17 years, 6 months

[Serusers] iptel.org - Technical board - Suggestion - REQUEST FOR FEEDBACK

by Greger V. Teigre

I believe most people by now have had time to respond to the thread where the idea and discussion from Prague around a technical board was started (April 12). Based on the feedback received, I summarize and offer the following make-it-simple suggestion: We create an iptel.org technical board (TB: 1. The board is responsible for resolving issues that is not possible to be resolve through a community consensus process, and it is allowed to take action where it finds necessary for the benefit of the iptel.org community. 2. The board is common for all iptel.org projects. 3. The board shall consist of five (5) members and the term shall be two years. 4. A board member can stay for several terms. 5. The five members shall be selected with one representative from SEMS, two from the SER developer (cvs write access) community, and two representatives for the broader iptel.org community. 6. All seruser and serdev subscribers nominate candidates, as well as vote. 7. Electable members must be contributors to iptel.org in some way and should not be controversial among subscribers. 8. Each subscriber can give his/her vote to one or more candidates, up to the number of board members (5). Process forward: People should respond with concerns and suggestions to this post. If there is consensus for changes, I will work as the editor and write up a new suggestion. Once the community can agree to the set of rules governing the board, we will proceed to nomination of candidates and then voting. g-)

17 years, 6 months

[Users] When is RTP Proxy needed?

by lists＠infoway.net

Hi all, We are interested in developing yet-another peer-to-peer VoIP service. Something with a simple Softphone/IM-like client that allows users to talk/chat with one another. However, and pardon our technical ignorance, we don't have 100% clear if/when we need to use something like rtpproxy or mediaproxy. Basically, we want to do all the setup of the calls and keep track of CDRs. However, we're hope that the client's actual media traffic will flow directly among the users and not have to come through our servers. For practical purposes, most users will be in domestic US with some sort of xDSL service and behind NAT. Eventually, we would like to offer voicemail services and even PSTN termination (for a fee) to these users. Can anyone shed some light as to when, if really needed, will something like rtpproxy kick in? Thanks

17 years, 6 months

[Users] User Realm on Location table

by Edson

Hi... I have to manage several domains on a same SIP server, but there is a policy that prohibit/block inter-domain calls. Consider A(a)a.a, B(a)a.a and B(a)b.b. Now suppose that A(a)a.a wants to call B(a)a.a. But B(a)a.a is not connected and B(a)b.b is. After a 'lookup("location")', the RURI returned is B(a)b.b and that is not the user wanted. Location table has a column named 'Domain', but it's allways empty. How to populate this column? How to use it (make 'lookup' consider it)? And how to prevent A(a)a.a calling B(a)a.a to reach B(a)b.b? Edson

17 years, 6 months

[Users] Redirect the same INVITE to the SIP SERVER from Balancer?

by Alejandro Sanchez

Hello. My enviroment is the next: -Balancer OS. Red Hat 4 up. 4 Openser 1.1.1 (balance with the module dispatcher) -Sip Server OS. Red Hat 4 up. 4 Openser 1.1.1 The actual flow is this. VTA B C VTB ---register----> ---register----> <---200 OK------ <----200 OK----- ---invite------> -----invite----> <-404 NotFound-- --404 NotFound-- <----------register--------- ----register----> <----200 OK------ ------------ 200 OK--------> ----invite-----> <-404 NotFound-- VTA= videotelephone A B= balancer (openser 1.1.1 with dispatcher module) C= SipServer (openser 1.1.1) VTB= video telephone B How you see when the VTA sends again the same invite, the balancer(B) answer with the 404 not found but without ask again to the SipServer(C), and my problem is that VTB is late and i have to try find it in the second invite, then in the second invite i need ask again to the SipServer(C) if the VTB is register. VTA y VTB access to the IP network making a dial-up connection. Why the balancer dosen't send the second invite to SipServer? How do i get this flow? Thank's for the Help __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/

17 years, 6 months

[Serusers] How to define avp localy on he ser.cfg

by tzieleniewski

Hi, Is it possible to define avp in ser.cfg without loading it from the database?? What is the syntax? Thanks in advance! Tomasz

17 years, 6 months

[Serusers] t_relay_to_udp() method with ip address passed as avp? - ser 2_0_0

by tzieleniewski

Hi, Is it possible to pass the ip address in the function t_relay_to_udp(..) in the m module using avp. For instance: ... t_relay_to_udp("$relay_to_udp","5060"); ... When I try to do it this way I get the following error: Apr 30 14:30:03 sen /home/ser/ser/ser-2.0/ser[7552]: Maxfwd module- initializing Apr 30 14:30:03 sen ser[7552]: AVPops - initializing Apr 30 14:30:03 sen ser[7552]: ERROR: mk_proxy: could not resolve hostname: "$relay_to_udp" Apr 30 14:30:03 sen ser[7552]: ERROR: fixup_hostport2proxy: bad host name in URI <$relay_to_udp> Apr 30 14:30:03 sen ser[7552]: ERROR: fix_expr : fix_actions error Cheers Tomasz

17 years, 6 months

[Serusers] Ser2.0 wrong behaviour of the sl_reply() function?

by tzieleniewski

Hi, I am using SER rel 2_0_0 from CVS. I encountered the following situation. My scenario was the following: UA -> Application Serwer -> SER -> UA 1. UA sent an INVITE message to the application server. 2. Application Server changed the Request URI and forwarded statelessly the INVITE message to SER. 3. Everything would be ok but SER replied to the Application server instead of the client. Application server changed only the Request URI and nothing else. Shouldn't SER sent the response according to the VIA header value?? I kindly ask for Your help. Below is the ngrep: 192.168.0.112:5060 - this is application server 192.168.0.112:5061 - this is calling UA U 2007/04/25 16:17:05.652934 192.168.0.112:5060 -> 192.168.0.165:5060 INVITE sip:sen@voip.rd.touk.pl SIP/2.0. Via: SIP/2.0/UDP 192.168.0.112:5061;rport=5061;branch=z9hG4bKhnrxbtgp;received=127.0.0.1. Max-Forwards: 70. To: <sip:sen@voip.rd.touk.pl>. From: <sip:pz@voip.rd.touk.pl>;tag=jlegd. Call-ID: pfnqwekjwtuxbfv(a)192.168.0.112. CSeq: 87 INVITE. Contact: <sip:pz@192.168.0.112:5061>. Content-Type: application/sdp. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. Supported: 100rel. User-Agent: Twinkle/0.9. Content-Length: 304. . v=0. o=pz 86411392 1733597180 IN IP4 192.168.0.112. s=-. c=IN IP4 192.168.0.112. t=0 0. m=audio 8000 RTP/AVP 98 97 8 0 3 101. a=rtpmap:98 speex/16000. a=rtpmap:97 speex/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. # U 2007/04/25 16:17:05.699923 192.168.0.165:5060 -> 192.168.0.112:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.112:5061;rport=5060;branch=z9hG4bKhnrxbtgp;received=127.0.0.1;received=192.168.0.112. To: <sip:sen@voip.rd.touk.pl>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6a45. From: <sip:pz@voip.rd.touk.pl>;tag=jlegd. Call-ID: pfnqwekjwtuxbfv(a)192.168.0.112. CSeq: 87 INVITE. Proxy-Authenticate: Digest realm="voip.rd.touk.pl", nonce="462f640d527dfc442d53379e1f0167c8a12a583d", qop="auth". Server: Sip EXpress router (2.0.0-rc1 (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.0.165:5060 "Noisy feedback tells: pid=20441 req_src_ip=192.168.0.112 req_src_port=5060 in_uri=sip:sen@voip.rd.touk.pl out_uri=sip:sen@voip.rd.touk.pl via_cnt==1". . Cheers Tomasz

17 years, 6 months

[Serusers] load_attrs("$tu","@ruri") not working - SER 2_0_0

by tzieleniewski

Hi, I was trying to load user attributes form the database (which where previously defined and set through serweb) and after I tried to load them with load_attrs("$tu","@ruri") no attributes appeared for the user identified by ruri. When I loaded attributes using load_attrs("$tr","@ruri") from the uri_attrs table then everything was ok and attributes were loaded. Please point me what do I missed so the attributes are not loaded? Below is my ser.cfg code and database contents: if (lookup_user("$tu.uid", "@ruri")) { xlog("L_INFO", " route[INBOUND]: load_attrs($tu,@ruri) <%rm> <%ru> \n"); load_attrs("$tu", "@ruri"); dump_attrs(); xlog("L_INFO", " route[INBOUND]: load_attrs($tr,@ruri) <%rm> <%ru> \n"); load_attrs("$tr", "@ruri"); dump_attrs(); .... } database: ser@sen:~/ser/ser-2.0/serctl$ ./ser_db show user_attrs | grep testser 32 2 testser(a)voip.rd.touk.pl testser first_name 32 2 testser(a)voip.rd.touk.pl last_name 33 2 testser(a)voip.rd.touk.pl tzl(a)touk.pl email 32 2 testser(a)voip.rd.touk.pl phone 32 2 testser(a)voip.rd.touk.pl Europe/Warsaw timezone 33 2 testser(a)voip.rd.touk.pl 2007-02-26 13:50:18 datetime_created 32 0 testser(a)voip.rd.touk.pl 0 sw_allow_find 32 0 testser(a)voip.rd.touk.pl 0 sw_send_missed 33 2 testser(a)voip.rd.touk.pl sip:touk2@voip.touk.pl forward_blind 33 2 testser fd for and output in the log file: Apr 30 11:11:40 sen ser[29958]: route[INBOUND]: load_attrs($tu,@ruri) <INVITE> <sip:testser@voip.rd.touk.pl> Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:540: class=GLOBAL Apr 30 11:11:40 sen ser[29958]: AVP["lang"]="en" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:550: track=FROM class=DOMAIN Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:560: track=TO class=DOMAIN Apr 30 11:11:40 sen ser[29958]: AVP["did"]="voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: AVP["digest_realm"]="voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:570: track=FROM class=USER Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:580: track=TO class=USER Apr 30 11:11:40 sen ser[29958]: AVP["uid"]="testser(a)voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:590: track=FROM class=URI Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:600: track=TO class=URI Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: route[INBOUND]: load_attrs($tr,@ruri) <INVITE> <sip:testser@voip.rd.touk.pl> Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:540: class=GLOBAL Apr 30 11:11:40 sen ser[29958]: AVP["lang"]="en" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:550: track=FROM class=DOMAIN Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:560: track=TO class=DOMAIN Apr 30 11:11:40 sen ser[29958]: AVP["did"]="voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: AVP["digest_realm"]="voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:570: track=FROM class=USER Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:580: track=TO class=USER Apr 30 11:11:40 sen ser[29958]: AVP["uid"]="testser(a)voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:590: track=FROM class=URI Apr 30 11:11:40 sen ser[29958]: INFO: No AVP present Apr 30 11:11:40 sen ser[29958]: INFO: avp.c:600: track=TO class=URI Apr 30 11:11:40 sen ser[29958]: AVP["forward_blindd"]="sip:hellboy@voip.rd.touk.pl" Apr 30 11:11:40 sen ser[29958]: AVP["forward_voicemaill"]="sip:hellboy@voip.rd.touk.pl" Cheers Tomasz

17 years, 6 months

[Users] who removes top most route header? (openser 1.2.0 + wesip)

by Jose Antonio Garvayo

Hi, I have testing openser 1.2.0 with seas module and wesip. If I send an INVITE message with a Route header, the SIP servlet receives the INVITE with the full Route Header, and when I proxy it the Route is the same, but I expected that the top most Route header has been removed. Who is in charge to do it? openser or wesip? I first thought it might be openser through loose_route(), but it seems not to be the case. The openser cfg file is similar to the example given in wesip tutorial: route{ 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; }; xlog("L_INFO", "$rm: ci=$ci ft=$ft tt=$tt cseq=$cs ruri:$ru received from: $si:$sp \n"); if (!method=="REGISTER") record_route(); if (loose_route()) { xlog("L_INFO", "RR enforced\n"); append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { xlog("L_INFO", "uri == myself"); if (method=="REGISTER") { save("location"); exit; }; lookup("aliases"); if (!uri==myself) { xlog("L_INFO", "outbound alias\n"); append_hf("P-hint: outbound alias\r\n"); route(1); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; xlog("L_INFO", "usrloc applied\n"); append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if(!as_relay_t("app_server_one")){ t_reply("500","Application Server error"); } } route[2] { xlog ("L_INFO", "$rm: cseq:$cs $ru: RELAYING (PROXY) to $du (flags: $mF) ...\n"); if (!t_relay()) { sl_reply_error(); }; exit; } This the log from wesip: 14:43:37 27abr2007 DEBUG SeasMessageEvent [SeasRouter]- New message received, from 10.161.14.10:6025 to 10.161.14.10:7000 over UDP 14:43:37 27abr2007 DEBUG SeasTransactionEvent [SeasRouter]- hash_index = 60870,label=1842040723 flags=0 14:43:37 27abr2007 DEBUG ExpressMessageChannel [ExpressMChannel[76]]- peerAddress = 10.161.14.10/6025 14:43:37 27abr2007 DEBUG ExpressMessageChannel [ExpressMChannel[76]]- About to process INVITE sip:user2@mot.net SIP/2.0 14:43:37 27abr2007 DEBUG SipConnector [ExpressMChannel[76]]- createProcessor: Reusing existing processor 14:43:37 27abr2007 INFO SipProcessor [ExpressMChannel[76]]- An incoming message is being assigned 14:43:37 27abr2007 DEBUG ExpressMessageChannel [ExpressMChannel[76]]- Done processing INVITE sip:user2@mot.net SIP/2.0 14:43:37 27abr2007 DEBUG SipProcessor [SipProcessor[3]]- <<<<<<<<< Request Received <<<<<<<<< INVITE sip:user2@mot.net SIP/2.0 Via: SIP/2.0/UDP 10.161.14.10:6025;branch=z9hG4bK1-13972359389@10.161.14.101 Contact: <sip:10.161.14.10:6025;transport=UDP> Max-Forwards: 14 From: <sip:user1@mot.net>;tag=6661 To: <sip:user2@mot.net> Supported: 100rel Route: <sip:10.161.14.10:7000;lr=on>,<sip:10.161.14.10:6025;lr=on> Call-ID: 1-13972359389(a)10.161.14.10 CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,UPDATE,MESSAGE,PRACK,REFER Content-Type: application/sdp Content-Length: 337 14:43:37 27abr2007 DEBUG SeasHashTable [SipProcessor[3]]- Inserting hashIdx:60870 label:1842040723 into hashTable 14:43:37 27abr2007 DEBUG SipProcessor [SipProcessor[3]]- Transaction [ z9hG4bK1-13972359389(a)10.161.14.101] is new Transaction?? --> true 14:43:37 27abr2007 DEBUG EngineSipMapper [SipProcessor[3]]- Request from 10.161.14.10 mapped to: ConvergedEngine[Standalone].StandardHttpHost[localhost] 14:43:37 27abr2007 DEBUG ConvergedEngineValve [SipProcessor[3]]- Invoking Host:localhost 14:43:37 27abr2007 DEBUG ConvergedHostValve [SipProcessor[3]]- SipSession not found for requestINVITE, trying match against Rules 14:43:37 27abr2007 DEBUG ConvergedHost [SipProcessor[3]]- Mapped request to:/sampleservlet 14:43:37 27abr2007 DEBUG ConvergedContextValve [SipProcessor[3]]- new session created :1-13972359389@10.161.14.10 14:43:37 27abr2007 INFO ExpressMessageChannel [SipProcessor[3]]- Creating Express message channel 10.161.14.10/7000 14:43:37 27abr2007 DEBUG ExpressMessageChannel [SipProcessor[3]]- setting spiral=false 14:43:37 27abr2007 DEBUG SipRequest [SipProcessor[3]]- >>>>>>>>> Sending Request >>>>>>>>> INVITE sip:user2@mot.net SIP/2.0 Max-Forwards: 14 From: <sip:user1@mot.net>;tag=83FBA2EBC5A9EE4615D194DECA1EE955 To: <sip:user2@mot.net> Supported: 100rel Route: <sip:10.161.14.10:7000;lr=on>,<sip:10.161.14.10:6025;lr=on> CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,UPDATE,MESSAGE,PRACK,REFER Content-Type: application/sdp Call-ID: 1-13972359389(a)10.161.14.10 Contact: <sip:10.161.14.10:7000;transport=udp> P-AS-Hint: B2BUA Via: SIP/2.0/UDP 10.161.14.10:7000;branch=z9hG4bK2137041384 Content-Length: 337 Thanks in advance

17 years, 6 months

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sr-users April 2007