sr-users July 2004

sr-users@lists.kamailio.org

178 participants
449 discussions

[Serusers] ser 0.8.14 released
by Andrei Pelinescu-Onciul 01 Aug '04

01 Aug '04

The new ser stable version, ser 0.8.14 has just been released. This version is based on the latest rel_0_8_12 branch code and not on the cvs head. We decided to name it 0.8.14 to avoid confusion (0.8.13 was already taken by the unstable cvs branch). The new release can be downloaded from: ftp://ftp.berlios.de/pub/ser/0.8.14 (source code, pre-compiled binaries for various arhitectures, packages for various operating systems and linux distributions; for more informations see ftp://ftp.berlios.de/pub/ser/0.8.14/README ). If you use cvs, just add -rrel_0_8_14 to your cvs checkout command. Please read the attached NEWS file for more information. Andrei

5 7

[Serusers] CPL and billing
by Arne Scheffer 01 Aug '04

01 Aug '04

Gents, I've been going throught the draft-ietf-iptel-cpl-09.txt to try and find information on doing the following: I want to offer my clients the option to do call forwarding and other smart things with cpl. Now I can not find information on how I can change the source name of the caller. Just to explain: - I am a user and out of the office - I set a call forward to my mobile - some calls me and my cpl script is run - it tells the proxy to route the call to my mobile Who will now be billed for the call ? I can not find a way to change the calling-id. In this case I need to be billed for the outgoing call to the PSTN (mobile). Am I correct to state this is not possible with CPL ? regards, Arne.

3 2

[Serusers] Problem in SEMS voice recording..
by Koyama Zehi 01 Aug '04

01 Aug '04

Dear All, thanks for all of you to provide me your valuable advice to setup voice mail scenario... Now I am able to listen greeting message on my sip user agent ..but i am getting two problem .. 1) After greeting message answer machine has asked to record my message when i have recored the message and the answer machine has created the sound file its size is only 44 bytes and their is nothing to related my recorded message.. 2) The Ser is not forwarding the recorded message on callee's mail id..does it require some special setting on SER or user client... i am also inserting the part of log of answer_machine generated at the time voice recording... Thanks koyama 3657) DEBUG: execute (AmServer.cpp:284): everything is OK ! (13657) DEBUG: onInvite (AnswerMachine.cpp:137): email address for user 'koyama': zehi_koyama(a)yahoo.com (13657) DEBUG: startSession (AmSession.cpp:436): Starting session... (hash=A5243DB9-305C-43D8-9069-064621390DF1(a)192.168.2.202377134671,sess_key=00003559795FCA9D) (13657) DEBUG: parse_sdp_attribute (AmSdp.cpp:596): unknown attribute definition 'fmtp:101 0-11,16' (13657) DEBUG: run (AmSession.cpp:126): dir=0; msg_flags=0 (13657) DEBUG: setLocalPort (AmRtpStream.cpp:129): setting random local port (13657) DEBUG: setLocalPort (AmRtpStream.cpp:158): local rtp port set to 1074 (13657) DEBUG: sendToFIFO (AmRequest.cpp:194): msg=<:vm_reply:000035590D146ECF 200 OK 36435:321248925 00003559795FCA9D Contact: <sip:koyama@210.251.64.168> Content-Type: application/sdp . v=0 o=username 0 0 IN IP4 210.251.64.168 s=session c=IN IP4 210.251.64.168 t=0 0 m=audio 1074 RTP/AVP 3 a=rtpmap:3 / . > (13657) DEBUG: openReplyFifo (AmRequest.cpp:362): flags=800 (13657) DEBUG: openReplyFifo (AmRequest.cpp:364): flags=0 (13657) DEBUG: write_to_fifo (AmRequest.cpp:708): Write to fifo: completed (13657) DEBUG: getReturnCode (AmRequest.cpp:400): Waiting for Ser to respond to the FIFO request... (13657) DEBUG: getReturnCode (AmRequest.cpp:421): response from Ser: 200 fifo_t_reply succeeded (13657) DEBUG: getReturnCode (AmRequest.cpp:442): n_cmd.from_uri= <> (13657) DEBUG: getReturnCode (AmRequest.cpp:445): n_cmd.next_hop= <> (13657) DEBUG: getReturnCode (AmRequest.cpp:448): n_cmd.route= <> (13657) DEBUG: sendToFIFO (AmRequest.cpp:216): sendToFifo returns ret=getReturnCode=0 (13657) DEBUG: setRAddr (AmRtpStream.cpp:318): RTP remote address set to 202.224.193.183:16386 (13657) DEBUG: run (AmSession.cpp:141): Sending Rtp data to 202.224.193.183/16386 (13657) DEBUG: onSessionStart (AnswerMachine.cpp:153): starting sending announce...(/usr/local/lib/sems/audio/default_en.wav) (13657) DEBUG: trying to read WAV file (13657) DEBUG: tag = <RIFF> (13657) DEBUG: file size = <129710> (13657) DEBUG: tag = <WAVE> (13657) DEBUG: tag = <fmt > (13657) DEBUG: chunk_size = <18> (13657) DEBUG: fmt = <01> (13657) DEBUG: channels = <1> (13657) DEBUG: rate = <8000> (13657) DEBUG: bits/sample = <16> (13657) DEBUG: tag = <fact> (13657) DEBUG: chunk size = <4> (13657) DEBUG: tag = <data> (13657) DEBUG: chunk size = <129660> (13657) DEBUG: streamGet (AmAudio.cpp:483): 0-stuffing packet: adding 260 bytes (packet size=320) (13657) DEBUG: convert (AmAudio.cpp:208): end of stream (13657) DEBUG: trying to read WAV file (13657) DEBUG: tag = <RIFF> (13657) DEBUG: file size = <1638> (13657) DEBUG: tag = <WAVE> (13657) DEBUG: tag = <fmt > (13657) DEBUG: chunk_size = <16> (13657) DEBUG: fmt = <01> (13657) DEBUG: channels = <1> (13657) DEBUG: rate = <8000> (13657) DEBUG: bits/sample = <16> (13657) DEBUG: tag = <data> (13657) DEBUG: chunk size = <1602> (13657) DEBUG: streamGet (AmAudio.cpp:483): 0-stuffing packet: adding 318 bytes (packet size=320) (13657) DEBUG: convert (AmAudio.cpp:208): end of stream (13657) DEBUG: onSessionStart (AnswerMachine.cpp:156): ... announce reaches its end. (13657) DEBUG: onSessionStart (AnswerMachine.cpp:162): starting recording message...(/tmp/A5243DB9-305C-43D8-9069-064621390DF1(a)192.168.2.202.wav) (13657) DEBUG: run (AmServer.cpp:167): version= <0.2> (13657) DEBUG: run (AmServer.cpp:177): fct_name= <sip_request.bye> (13657) DEBUG: execute (AmServer.cpp:236): cmd.method= <BYE> (13657) DEBUG: execute (AmServer.cpp:237): cmd.user= <koyama> (13657) DEBUG: execute (AmServer.cpp:238): cmd.email= <> (13657) DEBUG: execute (AmServer.cpp:239): cmd.domain= <> (13657) DEBUG: execute (AmServer.cpp:240): cmd.dstip= <210.251.64.168> (13657) DEBUG: execute (AmServer.cpp:241): cmd.port= <> (13657) DEBUG: execute (AmServer.cpp:242): cmd.r_uri= <sip:koyama@210.251.64.168> (13657) DEBUG: execute (AmServer.cpp:243): cmd.from_uri= <sip:202.224.193.183:5060> (13657) DEBUG: execute (AmServer.cpp:244): cmd.from= <<sip:202.224.193.183>> (13657) DEBUG: execute (AmServer.cpp:245): cmd.to= <<sip:koyama@210.251.64.168>;tag=00003559795FCA9D> (13657) DEBUG: execute (AmServer.cpp:246): cmd.callid= <A5243DB9-305C-43D8-9069-064621390DF1(a)192.168.2.202> (13657) DEBUG: execute (AmServer.cpp:247): cmd.from_tag= <377134671> (13657) DEBUG: execute (AmServer.cpp:248): cmd.to_tag= <00003559795FCA9D> (13657) DEBUG: execute (AmServer.cpp:251): cseq_str= <2> (13657) DEBUG: execute (AmServer.cpp:256): cseq= <2>(2) (13657) DEBUG: execute (AmServer.cpp:258): cmd.key= <36432:51628236> (13657) DEBUG: execute (AmServer.cpp:259): cmd.route= <> (13657) DEBUG: execute (AmServer.cpp:260): cmd.next_hop= <> (13657) DEBUG: execute (AmServer.cpp:266): hdrs: `' (13657) DEBUG: execute (AmServer.cpp:273): body: `' (13657) DEBUG: execute (AmServer.cpp:284): everything is OK ! (13657) DEBUG: sadSession (AmSession.cpp:310): searching for session to destroy (hash=A5243DB9-305C-43D8-9069-064621390DF1(a)192.168.2.202377134671,sess_key=00003559795FCA9D) (13657) DEBUG: findSession (AmSession.cpp:374): session found with key (13657) DEBUG: destroySession (AmSession.cpp:332): sess_it != a_sessions.end() (13657) DEBUG: destroySession (AmSession.cpp:335): it != sess_it->second.end() (13657) DEBUG: destroySession (AmSession.cpp:342): session found: stopping session (13657) DEBUG: stopAndQueue (AmSession.cpp:290): session cleaner trying to stop 80cecf8 (13657) DEBUG: postEvent (AmEventQueue.cpp:47): AmEventQueue: trying to post event (13657) DEBUG: postEvent (AmEventQueue.cpp:55): AmEventQueue: event posted (13657) DEBUG: destroySession (AmSession.cpp:344): session stopped and destroyed (#sessions=0) (13657) DEBUG: sendToFIFO (AmRequest.cpp:194): msg=<:vm_reply:000035591E83AFBB 200 OK 36432:51628236 00003559795FCA9D Contact: <sip:koyama@210.251.64.168> . . > (13657) DEBUG: openReplyFifo (AmRequest.cpp:362): flags=800 (13657) DEBUG: openReplyFifo (AmRequest.cpp:364): flags=0 (13657) DEBUG: write_to_fifo (AmRequest.cpp:708): Write to fifo: completed (13657) DEBUG: getReturnCode (AmRequest.cpp:400): Waiting for Ser to respond to the FIFO request... (13657) DEBUG: onSessionStart (AnswerMachine.cpp:165): ... finished recording message (13657) DEBUG: onSessionStart (AnswerMachine.cpp:171): recorded data size: 0 (13657) DEBUG: sadSession (AmSession.cpp:310): searching for session to destroy (hash=A5243DB9-305C-43D8-9069-064621390DF1(a)192.168.2.202377134671,sess_key=00003559795FCA9D) (13657) DEBUG: destroySession (AmSession.cpp:351): sess_it == sessions.end() (13657) DEBUG: getReturnCode (AmRequest.cpp:421): response from Ser: 200 fifo_t_reply succeeded (13657) DEBUG: getReturnCode (AmRequest.cpp:442): n_cmd.from_uri= <> (13657) DEBUG: getReturnCode (AmRequest.cpp:445): n_cmd.next_hop= <> (13657) DEBUG: getReturnCode (AmRequest.cpp:448): n_cmd.route= <> (13657) DEBUG: sendToFIFO (AmRequest.cpp:216): sendToFifo returns ret=getReturnCode=0 (13657) DEBUG: run (AmSession.cpp:243): Session cleaner starting its work (13657) DEBUG: run (AmSession.cpp:257): session 80cecf8 has been destroyed' (13657) DEBUG: run (AmSession.cpp:280): Session cleaner finished --------------------------------- Do you Yahoo!? Y! Messenger - Communicate in real time. Download now.

2 1

[Serusers] To Send a mail...
by murat 01 Aug '04

01 Aug '04

Hello, My 'ser' and 'mysql' are running without any problem !. Last situation of the "ser.cfg" is: if (uri=~"mustafa.avrupayazilim.com") { if (method=="REGISTER") { if (!www_authorize("mustafa.avrupayazilim.com", "subscriber")) { www_challenge("mustafa.avrupayazilim.com", "0"); break; }; My proble is: I tried to send an email with "serctl mail " command. I have an user name: "murat", at the database and with e-mail addres: murat_memet(a)yahoo.com. I finished the letter with "Ctrl+c ", and see a messege " message send. Bu t I didnt' see any maseege my malil box at Yahoo. Dou you have any suggesions?. One more question: I can not send any mail to . murat(a)iptel.org..... Thanks for your help

2 1

[Serusers] sems, ser routing
by Wulliamoz David 31 Jul '04

31 Jul '04

Hi, I'm trying to set up a capi gw with sems and ser. How to write the routing statement from sip phone to pstn if ser and sems are on the same host ? Do they communicate via ser_fifo and am_fifo? thanks, david

2 1

[Serusers] ser_moni.php is not showing any values
by Sanjiv Ranjan 31 Jul '04

31 Jul '04

Hi All, My ser_moni.php is not showing any values.Although, I have started cron_job from command prompt. Any help. Thanks. Sanjiv

2 1

[Serusers] connect_db():too many connections
by Blanca Mayayo 30 Jul '04

30 Jul '04

Hi, when I try to configure ser for the administration of two different domains (with two instancesin two different ports), I obtain the following error: connect_db():Too many connections. db_init():Error while trying ot connect database then signal 15 is received Could you tell me if that would be a problem of the mysql configuration, and if true, how can I solve it? Regards, Blanca ______________________________________________________ Consigue tu propio Hosting y Dominio al mejor precio: http://webcenter.lycos.es/

5 4

[Serusers] pstn AS5350 Radius_Acc
by Rafael J. Risco G.V 30 Jul '04

30 Jul '04

I have an accounting problem in this simple escenario: PSTNCaller --> Cisco AS5350 --> SER --> IPPhone I can see complete radius Accounting log (Start and Stop generated by ser) when the callee IPPhone hung up first, but when the PSTNCaller hung up fisrt just the "start" is generated in radius and the "stop" log is missing, in this case when I watch messages using ngrep in port 5060 I´ve realized that GW sends BYE twice to ser and IPPhone also sends OK twice to Ser... any idea? regards Rafael PS: relevant as5350 config: dial-peer voice 2001 voip description PSTN-->SIPSER destination-pattern 6604000 session protocol sipv2 session target sip-server codec g723r63 ! ! sip-ua sip-server ipv4:200.110.X.Y ! --------- ser.cfg: # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ /* Uncomment these lines to enter debugging mode debug=20 fork=yes 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 ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) modparam("auth_radius","radius_config","/usr/local/etc/radiusclient/radiusclient.conf") modparam("auth_radius","service_type",15) modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) # ------------------------- 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 (method=="INVITE") { log(1, "INVITE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { log (1, "BYE or CANCEL\n"); setflag(1); }; if (uri==myself) { if (method=="REGISTER") { log(1, "REGISTER: Authenticating user\n"); if (!radius_www_authorize("")) { log(1, "REGISTER: challenging user\n"); www_challenge("", "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(); break; }; } ---------------------- --------------------------------- Do You Yahoo!? Todo lo que quieres saber de Estados Unidos, América Latina y el resto del Mundo. Visíta Yahoo! Noticias.

1 0

[Serusers] Conference ID for SIP calls and exec_dset problem. TID != Conferenceid
by David Stanaway 30 Jul '04

30 Jul '04

Hi, I am trying to prototype a ser module by first using a perl script in exec_dset, and I would like to record the calls conference ID. I had thought that the TID was the conferenceid (As seen on cisco vsa accounting) but they do not match. Here is an example: I get an INVITE request from a Cisco ATA using ser as an outbound proxy. The exec_dset script sees this in the environment: SIP_TID = 6d2796abca7d6e1d65249aa229bc2f53 The cisco 5350 I redirect to shows this in its cisco-vsa accounting: h323-conf-id = "h323-conf-id=8457403F E38341D8 872CFC52 183D0ADC" What is happening here? Is the cisco gateway generating the conferencid for the SIP call? Should it be using the SIP TID? -- David Stanaway <david(a)dialmex.net>

2 1

[Serusers] help me!! sems problem!!
by 祥仔 30 Jul '04

30 Jul '04

I had install ser+voicemail(sems), but I don't know how to edit my ser.cfg file. could someone post the ser.cfg file.And tell me how to know voicemail is work?How to test by myself ua? Is any ua support to use voicemail by sems? help me!! thanks.

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sr-users July 2004