sr-users February 2005

sr-users@lists.kamailio.org

215 participants
467 discussions

by szj

Hi, Daniel: Thanks for your kind help! but I check it. Right, use_domain is 0. Do you have any further advance? Thanks again. Best Regards. Sun Zongjun

19 years, 9 months

[Serusers] SER stateful

by Vitor Brasileiro

Hi list, I'm in a doubt about stateful vs. stateless ser operation and accounting issues. Using stateless mode, accounting informations became different? There's other important thing about it? What kind of method should I use? Thanks, Vitor Brasileiro.

19 years, 9 months

[Serusers] rtpproxy mess

by Christian de Balorre

Hello, I am struggling with rtpproxy and I would like some help, because the documentation is rather scarce to be honest. I am running ser on an OpenBSD 3.3 box, with one internal interface (192.168.x.x) and a public interface (62.x.x.x). I am not sure wether I have to force rtpproxy to listen to a specific interface or not. I launch rtpproxy rather basically with ./rtpproxy -s /var/run/rtpproxy.sock Then I launch ser, it spawns some process and then run into race condition (see ps -aux below) Here is log file : Feb 2 15:34:28 cdg8-ser1 ser: WARNING: could not read from /dev/random (5) Feb 2 15:34:28 cdg8-ser1 ./ser[20970]: ERROR: send_rtpp_command: can't connect to RTP proxy Feb 2 15:34:28 cdg8-ser1 ./ser[32404]: ERROR: send_rtpp_command: can't connect to RTP proxy Feb 2 15:34:28 cdg8-ser1 ./ser[32404]: WARNING: rtpp_test: can't get version of the RTP proxy Feb 2 15:34:28 cdg8-ser1 ./ser[32404]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily Feb 2 15:34:28 cdg8-ser1 ./ser[20970]: WARNING: rtpp_test: can't get version of the RTP proxy Feb 2 15:34:28 cdg8-ser1 ./ser[20970]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily Feb 2 15:34:28 cdg8-ser1 ./ser[147]: ERROR: send_rtpp_command: can't connect to RTP proxy ... And the ps USER PID %CPU %MEM VSZ RSS TT STAT STARTED TIME COMMAND ser 16586 99.0 0.5 1744 2672 ?? R 3:34PM 6:15.11 ./ser -P /var/run/ser.pid root 16613 0.0 0.1 752 272 ?? Ss 3:33PM 0:00.01 ./rtpproxy -s /var/run/rtpproxy.sock #./rtpproxy -v 20040107 # ./ser -V version: 0.8.14 (i386/openbsd) Nathelper config : #-------------- nathelper parameters modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") Could it be that ser running as ser user cannot read the rtpproxy socket (running as root) ? Thanks Christian

19 years, 9 months

[Serusers] Problems Compiling SEMS on Solaris 10.

by Sergio Armando Gutierrez Betancur

Good afternoon Currently, we are triyng to compile SEMS on a Sun FIRE 220R with Solaris 10 build 72, but we are having a kind of conflict with the definition of a structure in the file AmApi.cpp We are using gcc 3.3.2, and gnu make. The error is the following. -------------- Begin compiling output --------- # make make[1]: Entering directory `/export/home/operador/answer_machine' ...... Some output ommited: g++ -c -o AmApi.o AmApi.cpp -Wall -Wno-reorder -D_DEBUG -g -DTHREAD_SAVE -D_REENTRANT -D_THREAD_SAFE In file included from /usr/include/netinet/in.h:62, from AmRtpPacketTracer.h:7, from AmRtpStream.h:34, from AmSession.h:31, from AmApi.h:32, from AmApi.cpp:28: /usr/include/sys/stream.h:67: error: `struct queue' redeclared as different kind of symbol /usr/local/include/c++/3.3.2/bits/stl_queue.h:71: error: previous declaration of `template<class _Tp, class _Sequence> class std::queue' make[1]: *** [AmApi.o] Error 1 make[1]: Leaving directory `/export/home/operador/answer_machine' make: [all] Error 2 (ignored) -------------- End compiling output --------- Does somebody have any hint about this? Thanks in advance for your help. Sergio Armando Gutiérrez Betancur Equipo GSC Área Operación Nuevos Servicios Subgerencia DSS Telco. Gerencia Telco Empresas Publicas de Medellín E.S.P. * (574)4150540 * sgutierb(a)eeppm.com -----Mensaje original----- De: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] En nombre de serusers-request(a)lists.iptel.org Enviado el: Viernes, 04 de Febrero de 2005 04:20 p.m. Para: Sergio Armando Gutierrez Betancur Asunto: Welcome to the "Serusers" mailing list Welcome to the serusers(a)lists.iptel.org mailing list! To post to this list, send your email to: serusers(a)lists.iptel.org General information about the mailing list is at: http://lists.iptel.org/mailman/listinfo/serusers If you ever want to unsubscribe or change your options (eg, switch to or from digest mode, change your password, etc.), visit your subscription page at: http://lists.iptel.org/mailman/options/serusers/sgutierb%40eeppm.com You can also make such adjustments via email by sending a message to: serusers-request(a)lists.iptel.org with the word `help' in the subject or body (don't include the quotes), and you will get back a message with instructions. You must know your password to change your options (including changing the password, itself) or to unsubscribe. It is: sagutier Normally, Mailman will remind you of your iptel.org mailing list passwords once every month, although you can disable this if you prefer. This reminder will also include instructions on how to unsubscribe or change your account options. There is also a button on your options page that will email your current password to you. "Este mensaje y/o sus anexos son para uso exclusivo de su destinatario intencional y puede contener información legalmente protegida por ser privilegiada o confidencial. Si usted no es el destinatario intencional del mensaje por favor infórmenos de inmediato y elimine el mensaje y sus anexos de su computador y sistema de comunicaciones. Igualmente, le comunicamos que cualquier retención, revisión no autorizada, distribución, divulgación, reenvío, copia, impresión, reproducción, o uso indebido de este mensaje y/o sus anexos, está estrictamente prohibida y sancionada legalmente. Agradecemos su atención. Empresas Públicas de Medellin E.S.P."

19 years, 9 months

[Serusers] Radius Authentication Problem

by M.V. Jaga Mohan

Hi List, I have downloaded freeradius 0.9.1 and SER-0.8.14 source and radiusclient library version 0.4.2. compiled all of them and my radius is working well as I have tested with Radtest. I have compiled SER with radius module and I have got auth_radius.so also. But when I am trying to run ser I am getting error called : init_mod() : Error while initializing module. Kindly help me in this regard. Its very urgent. Below I have provided the debug log. 0(3672) WARNING: could not rev. resolve 192.168.5.1 Listening on 127.0.0.1 [127.0.0.1]:5060 192.168.5.1 [192.168.5.1]:5060 Aliases: localhost:5060 localhost.localdomain:5060 WARNING: no fork mode and more than one listen address found (will use only the the first one) 0(3672) DEBUG: init_mod: sl_module stateless - initializing 0(3672) DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(3672) DEBUG: MD5 calculated: b27e1a1d33761e85846fc98f5f3a7e58 0(3672) DEBUG: init_mod: tm 0(3672) TM - initializing... 0(3672) Call-ID initialization: '7fbeafda' 0(3672) DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(3672) DEBUG: register_fifo_cmd: new command (t_uac_cancel) registered 0(3672) DEBUG: register_fifo_cmd: new command (t_hash) registered 0(3672) DEBUG: lock_initialize: lock initialization started 0(3672) DEBUG: register_fifo_cmd: new command (t_stats) registered 0(3672) DEBUG: MD5 calculated: 533cb9e91f4b999cf76861cbb9ed54ed 0(3672) DEBUG: MD5 calculated: a6a1c5f60faecf035a1ae5b6e96e979a 0(3672) DEBUG: init_mod: rr 0(3672) rr - initializing 0(3672) DEBUG: init_mod: maxfwd_module Maxfwd module- initializing 0(3672) DEBUG: init_mod: usrloc 0(3672) usrloc - initializing 0(3672) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(3672) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(3672) DEBUG: init_mod: registrar 0(3672) registrar - initializing 0(3672) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(3672) find_export: found <ul_bind_usrloc> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_register_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_get_all_ucontacts> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_insert_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_delete_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_get_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_lock_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_unlock_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_release_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_insert_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_delete_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_get_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) find_export: found <ul_update_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(3672) DEBUG: init_mod: auth 0(3672) auth module - initializing 0(3672) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(3672) DEBUG: init_mod: auth_radius 0(3672) auth_radius - Initializing 0(3672) auth_radius: Error opening configuration file 0(3672) init_mod(): Error while initializing module auth_radius ERROR: error while initializing modules 0(3672) DEBUG: tm_shutdown : start 0(3672) DEBUG: tm_shutdown : empting DELETE list 0(3672) DEBUG: tm_shutdown : empting hash table 0(3672) DEBUG: tm_shutdown: releasing timers 0(3672) DEBUG: tm_shutdown : removing semaphores 0(3672) DEBUG: tm_shutdown : done 0(3672) shm_mem_destroy 0(3672) destroying the shared memory lock and My ser.conf file is like : # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) #Uncomment these lines to enter debugging mode debug=9 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("auth_radius", "service_type", 15); # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("192.168.5.1")) { www_challenge("192.168.5.1","0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } =====

19 years, 9 months

[Serusers] How to make precall announcements?

by Hauke Zuehl

Hello all :) I have a problem which I do think about for some days now. The situation is: I have one server running SER and another one with Asterisk. Some users (which are authenticated by SER) want to have an announcement before they do a call. An example: A user wants to know how much he has to pay for a call. So he piks up the phone, logs into SER and dials a number. Now a voice (spoken via Asterisk) should tell him: "Hey, your call willl cost 2 Cents per minute". Then the call should be given back to SER by Asterisk and SER routes the call. Another user does not want to hear any information so all calls will be routed directly. I thought about a script that looks into a database which user has booked this neat feature. That works! On Asterisk I have created an extension called "precall_<user>" (<user> is - of course - user's id). Now I want SER to rewirte <user> to precall_<user>, jump to Asterisk, afterwards SER snips precall_ and does the call (oh dear...so much "call"-ing *g*). I have following part of ser.cfg: if(method==INVITE) { # # some other stuff # ..... rewriteuser("precall_test_hauke"); rewritehost("asterisk.mydomain.de"); log(1,"MYLOG: Forward to precall\n"); forward(uri:host, uri:port); strip(8); log(1,"MYLOG: Do call\n"); forward(uri:host, uri:port); }; Unfortunately SER seems to try to do a new INVITE while forwarding!? But great, I see in Asterisk the incoming call to precall_test_hauke :) But then the call hangs somewhere :( Would be great if anyone has a hint or a link with an example for me :) Thanks and kind regards, Hauke This is done in ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

19 years, 9 months

[Serusers] Dynamic Redirect Issue about SER!

by George Lee

SER gurus: I am an newbie to SER usuage. At present a dynamic redirect issue happens to me. But it is very pity for SER not to support this case!! Problem Description: ==================== When configuring SER as only redirect server, config file regarding redirect routing section is partly writen as follows: #-----------------------request routing logic------------------------- #main routing logic if (method=="INVITE") { #rewrite current URI, which is always part of destination ser rewriteuri("sip:80000000@192.168.0.191:5060"); #redirect now sl_send_reply("301", "Redirect"); break; } #--------------------------------------------------------------------- where, redirecting destination URI is staticly configed into SER, but dynamic destination URI is what our IP-PBX product(RTCCP) want to hope. The following call flow diagram helps to understand above mentioned scenario. GrandStream HandyTone486 SER Our B2BUA IP PBX(RTCCP) IP: 192.168.0.253 IP: 192.168.0.252 IP: 192.168.0.191 +++++++++++++++++++++++++++++++++++++ First Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F1(INVITE)---------->| |<-----------------F2(302)------------>| |-----------------F3(ACK)------------->| |------------------------------------F4(INVITE)------------------------>| ( The subsequent call flow is omitted ) +++++++++++++++++++++++++++++++++++++ Second Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F5(INVITE)---------->| |<-----------------F6(302)------------>| |-----------------F7(ACK)------------->| |------------------------------------F8(INVITE)------------------------>| ( The subsequent call flow is omitted ) F1: INVITE sip:8000000@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=c526ecce44ec6d75 To: <sip:8000000@192.168.0.252;user=phone> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: d6169571fe4f59a2(a)192.168.0.233 CSeq: 50105 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 F2: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=c526ecce44ec6d75 To: <sip:8000000@192.168.0.252;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f9ed Call-ID: d6169571fe4f59a2(a)192.168.0.233 CSeq: 50105 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6155 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:8000000@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1" F4: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKc6905874d1906c58 From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=1be4a5d7d169d8f1 To: <sip:80000000@192.168.0.191:5060> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: 0d325b0e778321ee(a)192.168.0.233 CSeq: 17809 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 F5: INVITE sip:80000002@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=466d9b06b51ca7b4 To: <sip:80000002@192.168.0.252;user=phone> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: c755c6b0908a8ce9(a)192.168.0.233 CSeq: 25248 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 F6: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=466d9b06b51ca7b4 To: <sip:80000002@192.168.0.252;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7906 Call-ID: c755c6b0908a8ce9(a)192.168.0.233 CSeq: 25248 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6151 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:80000002@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1" F8: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bK475fff37270b762d From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=58a90f4229e70a80 To: <sip:80000000@192.168.0.191:5060> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: 2a79d9ac84e2f72e(a)192.168.0.233 CSeq: 62121 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 In fact, I think SER should retrieve user portion of header "to" URI of Invite message(for above example it should be 80000000 for first call,whereas 80000002 for second call) to construct userinfo of destination redirect uri for header contact of 3xx response(but hostinfo maybe get from SER config file). Meanwhile I study into sl_send_reply source code, but no progress to issue. So any suggestions or solutions are appreciated!! Thanks in advance!! George Lee(ShenZhen, CHINA)

19 years, 9 months

[Serusers] Can't find the latest registered users continued

by szj

Hi, All: I forget to remember one thing, My proxy domain name is sip.voiptest.com, registrar is voiptest.com. All of them are translated by my DNS. so registered URI is: sip:861111@voiptest.com while INVITE URI is: 861111(a)sip.voiptest.com. Does that have influence on the Usrloc and registrar modules. Usrloc only uses the username as a index to lookup. Thanks a lots. Best Regards. Sun Zongjun

19 years, 9 months

[Serusers] Success Report inside LAN

by info＠beeplove.com

Hello, I have succeessfuly installed SER and was able to register from Windows Messenger 5.1 in my LAN. I have two computers, one for SER and another for MySQL. Next, I will try to connect to SER from outside my LAN. All computers in my LAN have private IPs. I use NAT in my router. Please suggest me, what documents/readme I would read. Thanks, -- Mohammad Khan sip:beeplove@iptel.org

19 years, 9 months

[Serusers] new MediaProxy release 1.2.1

by Adrian Georgescu

Hello, this is a bug fixing release for the media relay part, the obvious improvement is the relaying of video streams, SER module remains unchanged. Changes from version 1.2.0 to 1.2.1 ------------------------------------------------- - Fixed payload=101 (Telephone event) as being wrongly interpreted as codec - Fixed memory leak caused by cyclic references between objects - Fixed memory leak caused by exception tracing - Fixed typo in variable name for counting idle time - Fixed issue with jerky video To download: http://mediaproxy.ag-projects.com/mediaproxy-1.2.1.tar.gz To upgrade, unpack over your existing installation and restart mediaproxy. Regards, Adrian Georgescu

19 years, 9 months

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2006

2005

2004

2003

2002

sr-users February 2005