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37547 discussions

[Users] Redirect the same INVITE to the SIP SERVER from Balancer?
by Alejandro Sanchez 30 Apr '07

30 Apr '07

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/

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[Serusers] How to define avp localy on he ser.cfg
by tzieleniewski 30 Apr '07

30 Apr '07

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

2 3

[Serusers] t_relay_to_udp() method with ip address passed as avp? - ser 2_0_0
by tzieleniewski 30 Apr '07

30 Apr '07

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

2 1

[Serusers] Ser2.0 wrong behaviour of the sl_reply() function?
by tzieleniewski 30 Apr '07

30 Apr '07

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

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[Serusers] load_attrs("$tu","@ruri") not working - SER 2_0_0
by tzieleniewski 30 Apr '07

30 Apr '07

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

2 1

[Users] who removes top most route header? (openser 1.2.0 + wesip)
by Jose Antonio Garvayo 30 Apr '07

30 Apr '07

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

2 3

[Serusers] table in database ser
by Tuan Nguyen 30 Apr '07

30 Apr '07

Hi, I have created ser database for run ser. but I got follows error: 0(8054) init_tcp: using epoll_lt as the io watch method (auto detected) 0(8054) Maxfwd module- initializing 0(8054) submit_query: Table 'ser.global_attrs' doesn't exist 0(8054) db_query: Error while submitting query 0(8054) gflags:load_attrs: Error while quering database 0(8054) init_mod(): Error while initializing module gflags ERROR: error while initializing modules Can someone tell me where talk about which table i have to create in database ser. I can't find it. Many thanks for response! Tuan.

1 0

[Users] WeSIP redundancy/failover
by Andreas Granig 30 Apr '07

30 Apr '07

Hi, I was wondering if there's any mechanism in WeSIP for redundancy and failover. Assuming that I've one application server holding SIP dialog states, how could I add high availability to it? There's some sort of session replication in Tomcat [1], so is this somehow used by WeSIP? Are there any other best practices to redundantly set up WeSIP? Andreas [1] http://www.onjava.com/pub/a/onjava/2004/11/24/replication1.html This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

2 1

[Users] Forward Sessions to another SIP server
by konstantinos nikou 30 Apr '07

30 Apr '07

Dear all, I have the following issue: I would like my calls coming to openSER with URI 40XXXXXXX(a)abc.com to be forwarded to another SIP server with IP e.g. 10.10.0.1 with URI +140XXXXXXX(a)domain.com. Could someone tell me how i could do it? Thank you very much. Best regards, K.N. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] Problem with avp_db_load in OpenSER 1.2
by Daryl Sanders 29 Apr '07

29 Apr '07

I cannot seem to get my usr_preferences to load using the following load statement. avp_db_load("$avp(s:caller_username)", "*"); I have verified with xlog and avp_print() that $avp(s:caller_username) does contain a valid value before calling avp_db_load, however nothing is returned. It will work if I use this: avp_db_load("$fu/username", "*"); I really need to make it work via $avp(s:caller_username) if possible because my call forwarding and custom call routing features change this value often. Thanks, - Daryl

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