sr-users October 2003

sr-users@lists.kamailio.org

105 participants
230 discussions

[Serusers] SEMS
by Alessio Focardi 31 Oct '03

31 Oct '03

Hi, I need some documentation for SEMS, havent found none by now. I have it running and working but I would like to explore more possibilities, for instance: Message playback via SIP client or/and Storing the voice message in a folder then sending an email with a http link to it instead of using attach method of/and Compressing the message using MP3/Ogg and so on ... Also: can SEMS be used to provide voice mail messages insted of result codes for failed calls ? Something like "the user is busy", "no such user". I tested addaline.com service, that runs on SER, and they do this. Any help will be appreciated! TNX -- Best regards, Alessio mailto:alessiof@interconnessioni.it

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[Serusers] Re: One Way Audio call
by Raymond May 31 Oct '03

31 Oct '03

Hi, I have just installed SER on my Linux Debian server. It seems to work just great for users on the Internet calling each other ( 2 way Audio is successful ). However if someone calls me ( I am behind my SER server ) they can hear me but I can't hear them ( 1 way Audio ONLY ). Based on all the docs I've read it seems to have something to do with NAT and RTP ports. My Phone connected to LAN --- SER Server connected to my LAN on one interface and to DSL Provider on the other ---- DSL Service Provider. I am also doing NAT/PAT on SER server. Below is my Topology and ser.cfg. Appreciate any feedback I get. My Phone --- SER --- Internet --- Users that register with my SER server. SER.CFG File # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- 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=7 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/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.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") # -- 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 (len_gt( 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 (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "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(); }; } __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

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RE: [Serusers] Outbound proxy
by Klaus Darilion 31 Oct '03

31 Oct '03

Hi Peter! Not every SIP proxy can deal with clients behind a firewall/NAT box. Therefore it is useful to us an outbound proxy which is known to be capable of NATed clients. This proxy will send keep alive messages to your client to keep the pinhole in your firewall open and rewrites the IP addresses in the SIP messages of the client behind the FW (or NAT box). yes. ser can be an outbound proxy. every proxy can be a outbound proxy. An outbound proxy is not a special proxy type. Outbound proxy just means that the SIP UA client always sends the SIP messages to this proxy instead of contacting the other participant directly. regards, klaus > -----Original Message----- > From: Peter Hudec [mailto:phudec@postel.sk] > Sent: Saturday, November 01, 2003 12:21 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] Outbound proxy > > > hello, > > this is maybe a bit silly question, but. > > Outbound proxy is needed, when UA is behind FW. > What is the function of this proxy (how can this help the UA to make > calls) and what kind of software can do this. > > Can SER be also outbound proxy, or I have to use something else ? > > hudecof > > -- > mail: [phudec(a)postel.sk] www: [http://www.postel.sk] > cellular: [+421 02 50203166] icq: [99518783] > gpg: [http://hudecof.net/data/hudecof.gpg] > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] Fwd: SIP Tutorial at IEEE CCNC
by Jiri Kuthan 31 Oct '03

31 Oct '03

FYI -- there will be a SIP tutorial in the upcoming CCNC conference, hold by Charles Baker. -jiri >I will be presenting a half-day tutorial on SIP at the IEEE Consumer >Communications and Networking Conference in Las Vegas on Thursday, January >8. This is a new conference which is positioned to overlap the Consumer >Electronics Show. See < www.CCNC2004.org > [...] > Dr. H. Charles Baker, P.E., > Professional Web Site: http://engr.smu.edu/~hbaker

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[Serusers] t_uac_dlg() bug ?
by Jim Burwell 31 Oct '03

31 Oct '03

Hi, I was trying to get click-to-dial working from SERweb and it was failing for unclear reasons. I then experimented with the "ctd.sh" script, which does the same thing as the SERweb PHP code. I found that it would fail during the 'dummy invite' stage, returning an odd error if I used a URI which used a SIP domain/realm, but would work fine if I used the IP address of the SIP proxy as the domain. i.e.: ./ctd.sh sip:joebob@mydomain.com sip:billybob@mydomain.com <- fails ./ctd.sh sip:joebob@1.2.3.4 sip:billybob@mydomain.com <- succeeds The reply from SER would be "500 fifo_uac: no mem for hf block" . The log file would have these errors: ERROR: mk_proxy: could not resolve hostname: "mydomain.com" ERROR: uri2proxy: bad host name in URI <sip:joebob@mydomain.com> ERROR: uri2sock: Can't create a dst proxy ERROR: get_hf_block: send_sock failed ERROR: fifo_uac: no mem for hf block So it appears that SER didn't understand that the URI's domain was itself, even though there's a host alias line in the ser.cfg file listing "mydomain.com" as an alias for this host. It's as if it thinks it needed to relay the transaction to another SIP router, and look up an IP to do so. If "mydomain.com" were placed in /etc/hosts as an alias for the system SER was running on, or if a DNS SRV entry for the SIP domain were created pointing to the SER system, voila, the "sip:user@domain" form would work just fine. Is this a bug ? I think SER isn't checking SER host aliases against the SIP domain if the SIP request comes in via the fifo t_uac_dlg() command. I'm using the SER release 0.8.11-r1, aka the "stable CVS" release. TIA, Jim -- +---------------------------------------------------------------------------+ | Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC | +---------------------------------------------------------------------------+ | "I never let my schooling get in the way of my education." - Mark Twain | | "UNIX was never designed to keep people from doing stupid things, because | | that policy would also keep them from doing clever things." - Doug Gwyn | | "Cool is only three letters away from Fool" - Mike Muir, Suicyco | | "..Government in its best state is but a necessary evil; in its worst | | state an intolerable one.." - Thomas Paine, "Common Sense" (1776) | +---------------------------------------------------------------------------+ | Email: jimb(a)jsbc.cc ICQ UIN: 1695089 | +---------------------------------------------------------------------------+ | Reply problems ? Turn off the "sign" function in email prog. Blame MS. | +---------------------------------------------------------------------------+

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[Serusers] SerWeb
by jason 31 Oct '03

31 Oct '03

I have Ser installed and running on debian, I have mysql installed with the ser module. I can see the table the module created. I am getting this error message and not sure what it is, When trying the link..127.0.0.1/html/admin/index.php *Warning*: Unable to include '/prolog.html' - request execution failed in */var/www/html/page.php* on line *46* iptel.org User Management *Warning*: Unable to include '/separator.html' - request execution failed in */var/www/html/page.php* on line *48* *Warning*: Unable to include '/epilog.html' - request execution failed in */var/www/html/page.php* on line *82 Any ideas? *

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[Serusers] Warning - Noisy Feedback when trying to register from a messenger to SER
by Madhuri garigipati 31 Oct '03

31 Oct '03

Hello All, I am trying to register a user from a Messenger Application. I am able to register that user successfully and I am recieving a SIP/2.0 200 OK message. And I am able to see the registered user using the serctl utility. But I am getting the following warning message: Warning: 392 12.213.181.254:5060 "Noisy feedback tells: pid=2489 req_src_ip=67.161.15.238 req_src_port=2681 in_uri=sip:rajesh@12.213.181.254 out_uri=sip:rajesh@12.213.181.254 via_cnt==1" Can some one explain the reason for this warning and how to remove it? Find the complete send and recieve message trace below: ---- Sending ------- REGISTER sip:rajesh@12.213.181.254 SIP/2.0 Via: SIP/2.0/UDP 67.161.15.238 From: madhu <sip:madhu@67.161.15.238> To: rajesh <sip:rajesh@12.213.181.254> Call-ID: 522193516(a)67.161.15.238 Expires:10000 m: madhu <sip:madhu@67.161.15.238> CSeq: 1 REGISTER ---- Receiving ----- SIP/2.0 200 OK Via: SIP/2.0/UDP 67.161.15.238 From: madhu <sip:madhu@67.161.15.238> To: rajesh <sip:rajesh@12.213.181.254>;tag=b27e1a1d33761e85846fc98f5f3a7e58.d4f5 Call-ID: 522193516(a)67.161.15.238 CSeq: 1 REGISTER Contact: <sip:madhu@67.161.15.238>;q=0.00;expires=10000 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0 Warning: 392 12.213.181.254:5060 "Noisy feedback tells: pid=2489 req_src_ip=67.161.15.238 req_src_port=2681 in_uri=sip:rajesh@12.213.181.254 out_uri=sip:rajesh@12.213.181.254 via_cnt==1" Thanks, Madhuri. --------------------------------- Do you Yahoo!? Exclusive Video Premiere - Britney Spears

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Re[3]: [Serusers] Ser and Voicemail
by Alessio Focardi 31 Oct '03

31 Oct '03

Hello Jiri, Wow ... voicemail is working now: TNX ! I can ear "something" with my Xlite client but voice is pretty distorted, I suppose that I have to convert the default announcement to another sample rate/frequency/codec. Unfortunately nothing is recorded, here is a trascript of voicemail session, any help will be appreciated ... 200 OK 58383:45571714 00000BD46BBF2446 Contact: <sip:300001@interconnessioni.it:5090> Content-Type: application/sdp . v=0 o=username 0 0 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 30410 RTP/AVP 3 a=rtpmap:3 gsm/8000 . > (3163) DEBUG: openReplyFifo (AmRequest.cpp:292): openReplyFifo start (3163) DEBUG: openReplyFifo (AmRequest.cpp:313): creating reply fifo filehandle (3163) DEBUG: openReplyFifo (AmRequest.cpp:334): opening reply fifo (3163) DEBUG: write_to_fifo (AmRequest.cpp:635): Write to fifo: completed (3163) DEBUG: getReturnCode (AmRequest.cpp:380): response from Ser: 200 fifo_t_reply succeeded (3163) DEBUG: getReturnCode (AmRequest.cpp:401): r_uri= <> (3163) DEBUG: getReturnCode (AmRequest.cpp:404): cmd.next_hop= <> (3163) DEBUG: getReturnCode (AmRequest.cpp:405): cmd.route= <> (3163) DEBUG: run (AmSession.cpp:115): Sending Rtp data to 212.141.166.135/8000 (3163) DEBUG: onSessionStart (AnswerMachine.cpp:65): starting sending announce...(/sems/audio/default_en.wav) (3163) DEBUG: trying to read WAV file (3163) DEBUG: tag = <RIFF> (3163) DEBUG: file size = <129710> (3163) DEBUG: tag = <WAVE> (3163) DEBUG: tag = <fmt > (3163) DEBUG: chunk_size = <18> (3163) DEBUG: fmt = <01> (3163) DEBUG: channels = <1> (3163) DEBUG: rate = <8000> (3163) DEBUG: bits/sample = <16> (3163) DEBUG: tag = <fact> (3163) DEBUG: chunk size = <4> (3163) DEBUG: tag = <data> (3163) DEBUG: chunk size = <129660> (3163) DEBUG: play (AmRtpStream.cpp:244): AmRtpStream::play: using payload 3 (3163) ERROR: pcm16_2_gsm: number of blocks should be integral (block size = 320) (3163) DEBUG: onSessionStart (AnswerMachine.cpp:67): ... announce reaches its end. (3163) DEBUG: onSessionStart (AnswerMachine.cpp:73): starting recording message...(/tmp/4FC9AB89-F383-412E-930B-E1694B82AAFB(a)212.141.166.135.wav) (3163) DEBUG: record (AmRtpStream.cpp:267): AmRtpStream::record: using payload 3 ... (3163) DEBUG: onSessionStart (AnswerMachine.cpp:76): ... finished recording message (3163) DEBUG: onSessionStart (AnswerMachine.cpp:82): recorded data size: 0 (3163) DEBUG: sadSession (AmSession.cpp:274): searching for session to destroy (hash=4FC9AB89-F383-412E-930B-E1694B82AAFB(a)212.141.166.135569566049,sess_key=00000BD46BBF2446) (3163) DEBUG: destroySession (AmSession.cpp:313): sess_it == sessions.end() (3028) DEBUG: getReturnCode (AmRequest.cpp:380): response from Ser: 200 fifo_t_reply succeeded (3028) DEBUG: getReturnCode (AmRequest.cpp:401): r_uri= <> (3028) DEBUG: getReturnCode (AmRequest.cpp:404): cmd.next_hop= <> (3028) DEBUG: getReturnCode (AmRequest.cpp:405): cmd.route= <> (3030) DEBUG: run (AmSession.cpp:213): Session cleaner starting its work (3030) DEBUG: run (AmSession.cpp:227): session 80e6740 has been destroyed' (3030) DEBUG: run (AmSession.cpp:244): Session cleaner finished -- Best regards, Alessio mailto:alessiof@interconnessioni.it

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[Serusers] Pdt module troubles with accounting
by Todd Wright 30 Oct '03

30 Oct '03

Hi, Im currently trying to set SER 0.8.11 up and am having a problem with the pdt module which translates addresses based on a numeric prefix. When the accounting module tries to write its record to the database (mysql), the INSERT query is truncated just after the address that has been translated by the pdt module. This causes mysql to fail to insert the record and give an error about query syntax. Other accounting records for untranslated addresses are inserted properly with no error. Is this a known issue? I suspect the translated address is being null terminated by the pdt module but Im not enough of a code hacker to tell for sure. -- _--_|\ --------- Todd Wright -- wylie(a)geekasylum.org -------- / \ ICQ: 9589981 YIM: mvs38j \_.--._* <--- http://www.geekasylum.org/~wylie/ v Mobile: +61-403-796-001 Ph: +61-2-9699-1746 ----------------------------------------------------------------

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[Serusers] Question about an adaptation with SER
by Ivan Dario Salazar 30 Oct '03

30 Oct '03

Hi... Thanks a lot for your help I have a question... I have tens of telephony lines in a university that all they work with PSTN... so i wish i could do something with SER to manage the phone calls that they do, and this way save money...is it possible, what i need(Hardware, Software). Thankyou so much.. IVAN DARIO _________________________________________________________ http://www.latinmail.com. Gratuito, latino y en espa�ol.

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sr-users October 2003