October 2003 - sr-users

[Serusers] does not work voicemail
by Sergio Gerardo Festa 28 Oct '03

28 Oct '03

hello I have problems to make to operate the voicemail. when it is accomplished the call not lets me to record the message and is lost the ans_machine. Español hola tengo problemas para hacer funcionar el voicemail. cuando se realiza la llamada no me deja grabar el mensaje y se pierde el ans_machine. estos son mis archivos de configuracion. Log messages Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: ACC: call missed: from=Save1<sip:102@mydomain.com>;tag=28778-a725, i-uri=sip:100@mydomain.com, method=INVITE, o-uri=sip:100@262.61.161.129:5060;user=phone, code=488 Not Acceptable Here Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: failure_route[1]:jump to route[4]:vm Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: submit_query(): You have an error in your SQL syntax near ''' at line 1 Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: submit_query(): Error while submitting query Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: ERROR: vm: db_query() failed. Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: ERROR: vm: vm_get_user_info failed Oct 28 17:14:09 chitara /usr/sbin/ser[21057]: route[3]:vm:voicemail failed Oct 28 17:14:09 chitara /usr/sbin/ser[21047]: route[]:eof Oct 28 17:14:10 chitara /usr/sbin/ser[21073]: route[]:eof ---------------------------------------------------------------------------- ------------------------------ SER.CFG # # $Id: ser.cfg,v 1.20 2003/05/31 21:12:19 jiri Exp $ # # config script with voicemail, PSTN dial-out functionality # # ----------- global configuration parameters ------------------------ debug=1 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) # Uncomment these lines to enter debugging mode /* debug=8 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" sip_warning=no listen=262.69.20.92 alias=sip.boxip.com.ar # # ------------------ 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/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" loadmodule "/usr/lib/ser/modules/vm.so" loadmodule "/usr/lib/ser/modules/pa.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/textops.so" # loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/group.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") # # modparam("tm", "fr_inv_timer", 30) #INVITE timeout modparam("tm", "fr_timer", 15) #negative INVITE reply or no #final reply for a request for ACK # modparam("voicemail", "db_url", "sql://ser:heslo@localhost/ser") # modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # modparam("acc", "log_level", 1) modparam("acc", "log_flag", 2) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_fmt", "fimos") # # ------------------------- request routing logic ------------------- # # main routing logic # alias=262.69.20.92 #sip server IP address alias=mydomain.com #sip server FQDN # route{ # log(1,"entering main route"); setflag(2); #set flag for accounting # 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") { # digest authentication # log(1,"request for registration"); # if (!www_authorize("mydomain.com", "subscriber")) { # www_challenge("mydomain.com", "0"); # break; # }; save("location"); break; }; /* ********** Dial out to PSTN logic ************* */ #forward 411[information] and 911[emergency] requests to gateway if(uri=~"^sip:(9|2)11@(boxip\.com.ar|200\.69\.20\.92)"){ log(1,"411/911 expression match"); route(2); break; }; #forward numerical 7 digit requests to gateway if(uri=~"^sip:[0-9]{7}@(mydomain\.com|262\.69\.20\.92)"){ log(1,"7 digit expression match"); route(2); break; }; # strip 650 and forward to GW if user dials 650 before phone no. if(uri=~"^sip:650[0-9]{7}@mydomain\.com|262\.69\.20\.92)"){ strip(3); log(1,"650 area code dialed, 650 stripped"); route(2); break; }; #forward numerical 10 digit requests to gateway, append a 1 first if(uri=~"^sip:[0-9]{10}@(mydomain\.com|262\.69\.20\.92)"){ prefix("1"); log(1,"10 digit expression match, prefix 1"); route(2); break; }; #forward numerical 11 digit requests that start with a 1 to GW if(uri=~"^sip:1[0-9]{10}@(mydomain\.com|262\.69\.20\.92)"){ log(1,"10 digit exp match w/leading 1"); route(2); break; }; #forward international N digit requests to gateway if(uri=~"^sip:011[0-9]+@(mydomain\.com|262\.69\.20\.92)"){ log(1,"international expression match"); route(2); break; }; /* ********** VOICEMAIL logic ************* */ if (uri=~"^sip:voicemail\+@"){ log(1,"sip:voicemail uri match"); route(3); break; }; /* ****** Find Aliases and Locations of users ********* */ #lookup "aliases" before looking up "location" lookup("aliases"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { route(3); break; # this section needs help. I need to differentiate between users that # exits buy aren't online (send these requests to voicemail) # requests for users that aren't in my subscriber database # (send these requests a 404 reply) sl_send_reply("404", "User Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; log(1,"route[]:eof"); } route[2]{ log(1,"route[2]:SIP-to-PSTN call routed"); rewritehostport("262.69.20.92:5060"); if(!t_relay()){ sl_reply_error(); }; } route[3]{ log(1,"route[3]:vm:1"); if (method=="INVITE" || method=="BYE" || method=="REFER"){ log(1,"route[3]:vm:2"); if(t_newtran()){ t_reply("100","Trying -- just a second"); if(method=="INVITE" || method=="REFER"){ log(1,"route[3]:method==INVITE || REFER"); if(uri =~ "conference" ){ if(!vm("/tmp/am_fifo","conference")){ log(1,"route[3]:vm:conference failed"); t_reply("500","could not contact conference server"); }; } else if (uri =~"echo"){ if(!vm("/tmp/am_fifo","echo")){ log(1,"route[3]:vm:echo failed"); t_reply("500","could not contact echo"); }; } else{ if(!vm("/tmp/am_fifo","voicemail")){ log(1,"route[3]:vm:voicemail failed"); t_reply("500", "voicemail error"); }; }; break; }; if(method=="BYE"){ log(1,"route[3]:vm:method==BYE"); if(!vm("/tmp/am_fifo","bye")){ log(1,"route[3]:vm:bye failed"); t_reply("500" , "could not contact the media server"); }; break; }; } else{ log(1,"route[3]:vm:new transaction failed"); sl_send_reply("500", "new transaction failed"); break; }; }; } route[4]{ # this should be voicemail logic that is specific to a failure_route # i.e. line busy, or timeout after a certain period with no answer if(method=="INVITE" || method=="REFER"){ if(!vm("/tmp/am_fifo","voicemail")){ log(1,"route[3]:vm:voicemail failed"); t_reply("500", "voicemail error"); }; }else if(method=="BYE"){ log(1,"route[3]:vm:method==BYE"); if(!vm("/tmp/am_fifo","bye")){ log(1,"route[3]:vm:bye failed"); t_reply("500" , "could not contact the media server"); }; }; } failure_route[1]{ log(1,"failure_route[1]:jump to route[4]:vm"); # append_branch("sip:info at mydomain.com"); route(4); } ---------------------------------------------------------------------------- ------------------ SEMS.CFG # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for announce_path=/usr/lib/sems/audio/ # optional parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # optional parameter: max_record=<seconds> # # - maximum record time max_record=30 # optional parameter: smtp_server=<hostname> # # - sets address of smtp server smtp_server=262.61.161.129 # optional parameter: smtp_port=<port> # # - sets port of smtp server smtp_port=25 ################################## # module specific parameters # ################################## # add more module configurations here (inline or external): # # config.mymodule=<filename> # or # config.mymodule=inline # ... # config.mymodule=end ---------------------------------------------------------------------------- --------------------------- Thanks Gracias Saludos A su Disposición. Sergio Gerardo Festa

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[Serusers] exec module question
by Andres 28 Oct '03

28 Oct '03

Hi, I am testing the exec module by outputting some variables to a flat file using: SIP_HF_<hf_name> So for example I put something like exec_msg("echo `date` - From: $SIP_HF_FROM To: $SIP_HF_TO Contact: $SIP_HF_CONTACT Src_IP: $SIP_SRCIP User-Agent: $SIP_HF_USER-AGENT >> /var/log/ser/ser_register.log"); but the User-Agent header field does not get passed even though a sniffer trace shows it in the SIP REGISTER request. Can somebody tell me why not? Thanks, Andres

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[Serusers] Support for offline messages in an IM conference
by Madhuri garigipati 28 Oct '03

28 Oct '03

Hello All, I was going through the msilo package in SER modules. It says - "It stores received messages for an offline user and sends them when the user becomes online." I am creating a J2ME IM client for mobile devices using SIP at the backend. And the problem with mobile devices is connection loss. So I am planning to provide a feature to my IM client where in if more than 2 users are in a conference chat room, and one user gets disconnected for some reason, I want to store the messages during the user's absense and send it back to the user when they reconnect. Is this possible with SIP and SER? Does SER module msilo or any other module support offline messages for a conference chat? Thanks, Madhuri. --------------------------------- Do you Yahoo!? Exclusive Video Premiere - Britney Spears

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[Serusers] Messenger5.0 481 Error continued
by Rork, Joseph (J.P.) 28 Oct '03

28 Oct '03

I've followed the instructions: Attention: instant messaging model has changed from 4.7 to 5.0 dramatically and in a non-standard way. 5.0 takes session establishement using INVITE and drops out-of-dialog MESSAGEs on the floor, whereas sending "481 Call Leg/Transaction...". To permit them, apply the following registry setting: HighSecurityMode HKLM\Software\Policies\Microsoft\Messenger\Client\{83D4679F-B6D7-11D2-BF36-0 0C04FB90A03}\_Default\EnableSIPHighSecurityMode DWORD= 0 -- Low Security 1 -- High Security 2(Default, same as not set) --- Medium security You need set it to 0 in order to make peer-to-peer IM. Since WM4.7 client is not registered in the LCS server, I guess that WM5 treats it as peer-to-peer call. And I'm still getting the following error: <<messenger5dump>> Any ideas? --joe

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Re: [Serusers] SER Contact rewrite for Softswitch
by Jiri Kuthan 28 Oct '03

28 Oct '03

If you need to achieve security, why don't you let the proxy server authenticate callers using digest identity and the gateway trust the proxy by ip address? That's the most common practice today, we use it with our Cisco gateway too. -jiri At 04:27 PM 10/23/2003, Harry Behrens wrote: >Hello, > >we have the following setup in our "Unified Messaging lab": > >SUAs <-> LAN <-> SER <-> Softswitch <-> PSTN > >The SER is provisioned as a SIP Server (essentially a SUA) from the Softswich point of view. > >We have provisioned a number block of 20 numbers (from a 30-channel ISDN PRI) for direct inward dialling (DID) to the SER. >E.g. 12345620-12345639 are routed to the SER. > >On the SER we have a number of users <user>@dai-labor.de, where the first 20 users have an alias >of 123456[23][0-9] >Each user also has his number added to his permanent contacts to enable (on the cheap) a "call by PSTN if not online" poliy. > >The aim obviously being to have a 1-1 mapping between URI and allocated telephone number (we are still in the proof-of-concept phase and will move to DNS and ENUM based architecture at a later stage). > >Scenario 1: OK > PSTN user Up calls 12345621 - which is mapped to harry(a)dai-labor.de. > If harry(a)dai-labor.de is registered, his SIP UA rings and the call is established. > >Scenario 2: problem > harry(a)dai-labor.de wants to use his UA to place a call through the Softswitch. > The Softswitch "authenticates" (if you can call this authentication ;-( by looking at the > Contact header and comparing it to its database of numbers assigned to provisioned SIP Servers. > I therefore need to be able to specify a rewrite of the Contact header to > replace harry@<Contact IP:port> with 12345621@<Contact IP:port> > when the callee is of the form <number>@<whatever> > >I am aware that this would be strictly speaking not proxy but rather B2BUA behaviour, but would still be >interested to know whether anyone has a solution to this problem or whether we need to hack one ourselves. > >Many thanks in advance, > > Harry > >Dr. Harry Behrens >Projektleitung BIB3R >DAI Labor - Technische Universität Berlin >Sekretariat GOR 1-1, Franklinstrasse 28/29, 10587 Berlin > >Fon: +49 30 314 23383 >Fax: +49 30 314 21799 > >Email: harry.behrens(a)dai-labor.de > >http://www.dai-labor.de > > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] noch Voicemail
by Chen Youcheng 28 Oct '03

28 Oct '03

Hallo, laut Beschreibung von http://iptel.org/ser/doc/modules/txt/vm.txt muss Mann viele Parameters einstellen. Aber das Beispiel von http://www.iptel.org/ser/doc/prerelease/x1294.html ist anders. Nur ein Parameter : modparam("voicemail","db_url","modules/vm/db"). Kann jemand einmal erkl�ren? dayong1973 __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

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[Serusers] append_rpid_hf() and search()
by Alexander Mayrhofer 28 Oct '03

28 Oct '03

Hi, i'm adding remote party id information to certain SIP messages (to those destined to the PSTN gateway). After doing that, i try to search() the "Remote-Party-ID" header field for a specific string, and it seems to me that search() only works on the initial (received) SIP message rather than on the "current" modified message. Is that assumption correct? (i hope it's not :) If yes, is there another way to search SER's rpid information for a specific string? config snippet: if (!is_rpid_user_e164()) { append_rpid_hf(); xlog("L_INFO", "rpid added"); if (search("Remote-Party-ID:.*@1005")) { xlog("L_INFO", "rpid search succeeded"); }; ... thanx axelm

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[Serusers] SIP Instant Messaging UA
by dhiraj.2.bhuyan＠bt.com 28 Oct '03

28 Oct '03

Hello List, Can anyone recommend some SIP Instant Messaging UA (linux and windows) that follows the specification word by word? I tried MSN Messenger 4.6 and 5 - both appears to have some problems (version 5 tries to send a MESSAGE with the INVITE method and gets back a BYE in response!!!!!!!!!!!!!). Thanks in advance, Dhiraj Bhuyan Security Research Engineer BT Exact Tel: +44 1473 643932 Mob: +44 7962 012145 Email: dhiraj.2.bhuyan(a)bt.com

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RE: [Serusers] DHCP and SER
by Jiri Kuthan 28 Oct '03

28 Oct '03

At 03:13 AM 10/28/2003, Iizuka, Hirotaka, VF-JP wrote: >I'm sorry to reply back to you, I know you are busy. >How does SER query DNS? using DNS protocol. It takes DNS names from SIP messages and resolves them to IP addresses using DNS. >(Query with SIP message form?) >Where do I set the DNS's address in SER? What DNS address? If you mean the address at which SER is reachable, see Dan Austin's howto http://iptel.org/ser/doc/ser-howto/ser-Howto.html#AEN132 -jiri

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RE: [Serusers] DHCP and SER
by Iizuka, Hirotaka, VF-JP 28 Oct '03

28 Oct '03

I'm sorry to reply back to you, I know you are busy. How does SER query DNS?(Query with SIP message form?) Where do I set the DNS's address in SER? -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Monday, October 27, 2003 4:20 PM To: Iizuka, Hirotaka, VF-JP Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] DHCP and SER Of course. Jan. On 27-10 10:04, Iizuka, Hirotaka, VF-JP wrote: > Hi, > > Does SER send query to the DNS to get SIP clients' address? Thank you. > > Hiro > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] RTP Proxy and SER
by Stephen Miles 27 Oct '03

27 Oct '03

Thanks for that, I'll get the source from the CVS and give it a go. -----Original Message----- From: Andrei Pelinescu-Onciul [mailto:pelinescu-onciul@fokus.fraunhofer.de] Sent: Tuesday, 28 October 2003 10:05 a.m. To: Stephen Miles Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] RTP Proxy and SER On Oct 28, 2003 at 09:54, Stephen Miles <smiles(a)callplus.co.nz> wrote: > Hi Andrei, > > I have the nathelper module being loaded but am still getting an error. > Here is my log, can you shed any light on this for me? Yes, force_rtp_proxy exists only in the nathelper module from ser unstable (cvs HEAD). Sorry I didn't notice the part about using ser debian packages (which implies you use ser stable). You could try getting ser unstable sources and compiling it, however right now we are doing a lot of changes and the current cvs might be broken. It is probably safer to use a version from 2 weeks ago. Andrei

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[Serusers] How does SER's Stateful Proxy works?
by Dharani Narayanan 27 Oct '03

27 Oct '03

Hello I went through few modules of SER regarding the session maintenance. When i walked through the Transaction Module, I came to know that SER provides Stateful Proxy. I have few questions regarding this: 1. How does SER maintains session? i.e., Does it maintain any state table similar to file/socket descriptor? 2. In the case of "dialog" interaction between clients are identified by callID and routed through the Record-Route. Does Record-Route traces the session? If so where it is maintained? Will this become a overhead in header while routing all the details? Does the client feel efficient in performance? 3. How does SER knows whether the session is disconnected or logged off? And under what situation does the SER stores the offline messages? 4. Can you provide me the total architecture of how the stateful proxy is designed and works? Can you please update me for the above mentioned questions. Thanks Dharani. __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

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[Serusers] ser does not respond to messages with NATed address
by Yang Xiang 27 Oct '03

27 Oct '03

Hello, I have a NATed network behind a cisco PIX firewall as follows: client --- SER 1-----PIX (NAT) --------SER 2 The client has a private address 192.168...., and is NATed to a public IP. Because PIX has a bulit-in SIP-application-gateway, the IP of the client in message body will be transformed to his public IP by firewall.. However, what most strange is SER2 doesn't respond to messages from SER1. -------------- #################### U 153.96.51.2:5060 -> 153.96.52.2:5060 INVITE sip:6091@153.96.52.2:5060 SIP/2.0..Record-Route: <sip:6091574@153.96.51.2;ftag=781a5806-7443-4c67-b8a6-4ae6f9b64bc8;lr=on>..Max-Forwards: 9 . .Record-Route: <sip:6091@192.4.3.2;ftag=781a5806-7443-4c67-b8a6-4ae6f9b64bc8;lr=on>..Via: SIP/2.0/UDP 153.96.51.2;branch=z9hG4bK75cf.a543.0..Via: SI P/2.0/UDP 192.4.3.2;branch=0..Via: SIP/2.0/UDP 153.96.51.77:9779..From: "xia" <sip:xia@nd.fhg.de>;tag=781a5806-7443-4c67-b8a6-4ae6f9b64bc8..To: <s ip:6091@nd.fhg.de>..Call-ID: 9947a08e-d8c5-47ef-90a3-cdd10f3b143c@153.96.51.77..CSeq: 1 INVITE..Contact: <sip:153.96.51.77:9779>..User-Agent: W indows RTC/1.0..Content-Type: application/sdp..Content-Length: 457....v=0..o=yang-mobil 0 0 IN IP4 153.96.51.77..s=session..c=IN IP4 153.96.51.77..b=CT :1000..t=0 0..m=audio 61052 RTP/AVP 97 111 112 6 0 8 4 5 3 101..a=rtpmap:97 red/8000..a=rtpmap:111 SIREN/16000..a=fmtp:111 bitrate=16000..a=rtpmap:112 G7 221/16000..a=fmtp:112 bitrate=24000..a=rtpmap:6 DVI4/16000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:5 DVI4/8000..a=rtp map:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16.. ---------------------------------------------- (originally the client has an IP 192.168.44.33 and it has been transformed to 153.96.51.77 by firewall as seen above) SER2 gets all messages repeatedly from SER1 but does nothing as if it hadn't seen any messages. (no reply sent, no log written) If I change the client to another network with not NATed adress, it works fine! SER1's config: --------------------------- ... if (uri=~"^sip:[0-9]*@.*"){ # PSTN call after the authorization. Authorization has been done in bolck route[2] and [3] if (uri=~"^sip:00[0-9]+@.*"){ sl_send_reply("699", "Sorry, international call not allowed"); break; }; log(2, "Forward to ser2 \n"); rewritehostport("153.96.52.2:5060"); forward("153.96.51.2"); break; }; ... ----------------------- And SER2's config: # ----------- global configuration parameters ------------------------ debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) 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/ser/lib/ser/modules/sl.so" loadmodule "/usr/local/ser/lib/ser/modules/tm.so" loadmodule "/usr/local/ser/lib/ser/modules/rr.so" loadmodule "/usr/local/ser/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser/lib/ser/modules/nathelper.so" loadmodule "/usr/local/ser/lib/ser/modules/textops.so" modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ sl_send_reply("100","msg received, trying"); ### (not any reply was sent as msg arrived!) if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","too many hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "msg too big"); break; }; record_route(); loose_route(); if (search("(c|C)ontact:.*192\.168.*")) { fix_nated_contact(); if (method=="INVITE") { fix_nated_sdp("2"); }; }; if (!src_ip==10.20.0.2) { log(3, "Forward to inn. ser\n"); forward("10.20.0.2"); break; }; # if the request is for other domain use UsrLoc if (!t_relay()) { sl_reply_error(); }; } --------------------------------------------------- Any idea? Thanks Yang

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[Serusers] voicemail installation
by Chen Youcheng 27 Oct '03

27 Oct '03

Hallo, ich habe iptel ser instaliert. Es funktioniert gut. ich moechte jetzt Voicemail integrieren. unter http://www.iptel.org/ser/doc/prerelease/x1294.html ist die Beschreibung: First, you need to compile Ser with voicemail support. Thatfor, you must edit Ser's Makefile.defs file and uncomment the line with '-DVOICE_MAIL' and '-D_TOTAG'. Es gibt es keine '-DVOICE_MAIL' and '-D_TOTAG'. Kanst du mir mal helfen? dayong1973 __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

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[Serusers] Does SER support load balancing on the server-side?
by Madhuri garigipati 27 Oct '03

27 Oct '03

Hello, I have a question on load balancing. Does SER support load balancing on the server-side? If yes, how does it provide load balancing? What is the mechanism used? Thanks, Madhuri. __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

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Re: [Serusers] multiple users registering problem
by Jiri Kuthan 27 Oct '03

27 Oct '03

There may be lot of reasons why the old contact disappears. A typical reason is that the old contact expires after its expiration period. If you don't think that is the case, send us your message dumps. regards, -jiri At 03:01 AM 10/22/2003, Annie Sasidar wrote: >Hi, > I am using Windows messenger 4.7 for testing ser. I have a linux box as a >stateless proxy. Only a single user gets registered. When i register a new >user, the old record seems to be deleted. When i do a serctl ul show i get >only the last user registered. What should i do? >Any tips will be greatly appreciated. > >Thanks in advance, >Annie > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers]SIP Gateway
by Director General: NEFACOMP 27 Oct '03

27 Oct '03

Hi, I have received it but I couldn't reply because I am not in town now. I will check tomorrow to see if I can help. Thanks Emery ----- Original Message ----- From: "Sesha B" <sesha(a)iic.com> To: "Director General: NEFACOMP" <dg(a)nefacomp.net> Sent: Monday, October 27, 2003 17:05 Subject: RE: [Serusers]SIP Gateway > I'm sorry to trouble you again and again, but did you get the mail I sent > you earlier about my setup? Please let me know if you found anything wrong > with the setup or the config? Thank you very much. > > Sesha > > -----Original Message----- > From: Director General: NEFACOMP [mailto:dg@nefacomp.net] > Sent: Friday, October 24, 2003 7:06 AM > To: sesha(a)iic.com; serusers > Subject: Re: [Serusers]SIP Gateway > > > You asked: > >> Does the mediatrix be assigned a number too? > Can you explain? > Normally each port on the Mediatrix is like a UA (like Xten, ... [I might be > wrong!!]) > That is why, the account you tell it to use should be created in SER first. > > NGREP can be found at http://ngrep.sourceforge.net/ > After installing it you may use [>>>>>ngrep port 5060] to watch the traffic > on port 5060. > > When you get the dial tone, what do you do with it? Normally you should be > able to dial a number to call. > For example if the Xten software is running on your PC and registered to use > 500 as main alias, after getting the dial tone (with the phone connected to > the Mediatrix) you may dial 500 without hanging up and your UA will start > ringing on the PC. > > Can you tell me in more details how your devices are interconnected? > > > Thanks > Emery > ----- Original Message ----- > From: "Sesha B" <sesha(a)iic.com> > To: "Director General: NEFACOMP" <dg(a)nefacomp.net> > Sent: Thursday, October 23, 2003 16:58 > Subject: RE: [Serusers]SIP Gateway > > > > The analog gateway ID is in readonly mode in my MIB and it is set to 5 by > > default. I have disabled the redirection and I get the dialtone after two > > rings. And with the original configuration I have, I get the engage tone > > after two rings. I have one question! Does the mediatrix be assigned a > > number too? Also, can you please let me know how do use ngrep to watch the > > registration headers on the SER registrar? Thank you. > > > > Sesha > > > > -----Original Message----- > > From: Director General: NEFACOMP [mailto:dg@nefacomp.net] > > Sent: Thursday, October 23, 2003 10:01 AM > > To: sesha(a)iic.com > > Subject: Re: [Serusers]SIP Gateway > > > > > > I also have another question about your device: > > analogGatewayID (.1.3.6.1.4.1.4935.1.1.1.4.1.2.0) = 5 > > > > The above line says that your gateway ID is 5. It seems it is different > from > > mine. Mine is 2. > > Possibly I won't be of much help to you. But we can always try. > > > > Can you remove the redirection and see if it works? > > gatewayPort1RedirectEnable (.1.3.6.1.4.1.4935.1.1.1.4.1.14.10.2.0) = 0 > > ////// Just disable the redirection to see if that helps > > gatewayPort1RedirectToAddress (.1.3.6.1.4.1.4935.1.1.1.4.1.14.10.3.0) > = > > x.x.x.x > > gatewayPort1NetToPstnSourceFilter > > (.1.3.6.1.4.1.4935.1.1.1.4.1.14.10.4.0) = 24.105.144.220 > > > > For the remaining parts, I don't see any difference with my configurations > > (except that you are using country codes while me I have disabled them) > > > > > > Thanks > > Emery > > ----- Original Message ----- > > From: "Sesha B" <sesha(a)iic.com> > > To: "Director General: NEFACOMP" <dg(a)nefacomp.net> > > Sent: Thursday, October 23, 2003 14:32 > > Subject: RE: [Serusers]SIP Gateway > > > > > > > I'm attaching my getwalk from the branch voipgateway, which includes > > > everything. The phone I was mentioning earlier is the SIP phone into > which > > > I'm trying to dial from the regular phones on the PBX. I've the > mediatrix > > on > > > 4 ports on the PBX. Thank you. > > > > > > -----Original Message----- > > > From: Director General: NEFACOMP [mailto:dg@nefacomp.net] > > > Sent: Thursday, October 23, 2003 4:40 AM > > > To: sesha(a)iic.com; serusers > > > Subject: Re: [Serusers]SIP Gateway > > > > > > > > > What phone? > > > Can you send me the GetWalk generated by your Unit? > > > > > > > > > Thanks > > > Emery > > > ----- Original Message ----- > > > From: "Sesha B" <sesha(a)iic.com> > > > To: "Director General: NEFACOMP" <dg(a)nefacomp.net>; <serusers(a)lists.iptel.org> > > > Sent: Thursday, October 23, 2003 00:34 > > > Subject: RE: [Serusers]SIP Gateway > > > > > > > > > > I have these parameters already set. I also restarted mediatrix after > > > > > setting these MIB variables but I can see only the SIP phone > registering > > > > with the SER in the ser diagnostics but I dont see the SIP gateway's > > > address > > > > at all !!! > > > > > > > > -----Original Message----- > > > > From: Director General: NEFACOMP [mailto:dg@nefacomp.net] > > > > Sent: Wednesday, October 22, 2003 10:47 AM > > > > To: sesha(a)iic.com; serusers(a)lists.iptel.org > > > > Subject: Re: [Serusers]SIP Gateway > > > > > > > > > > > > I will send instructions for the first port and you will be able to do > > the > > > > same for the other ports. > > > > > > > > In the provisioning MIB, find the key: > > > > sipMib -> sipUAGroup > > > > Start by configuring the SER address: > > > > sipMib -> sipUAGroup -> sipUAServerGroup -> sipUAServerStaticGroup > > > > and configure the parameters therein. Such as Registrar Host, Port > > ... > > > > > > > > Second, configure the port 1 as follows: > > > > sipMib -> sipUAGroup -> sipUA1Group > > > > configure the main alias entry which should be the account created > > on > > > > the SER server for that port > > > > you may even configure other things if needed such as other > aliases, > > > > .... > > > > For other ports, just replace # in sipUA#Group by the correct port > > number > > > > (2, 3 or 4) > > > > > > > > Also, you may need to enable some options in the provisioningMib key. > > > > > > > > > > > > Thanks > > > > Emery > > > > ----- Original Message ----- > > > > From: "Director General: NEFACOMP" <dg(a)nefacomp.net> > > > > To: <sesha(a)iic.com>; <serusers(a)lists.iptel.org> > > > > Sent: Wednesday, October 22, 2003 15:53 > > > > Subject: Re: [Serusers]SIP Gateway > > > > > > > > > > > > > I was asking the firmware version (the software installed on the > > 1204). > > > > Not > > > > > the Unit Manager. > > > > > > > > > > I am going to send you instructions on how to set up accounts in the > > > unit. > > > > > These are accounts the unit uses in registering with the SER server. > > > > > Note that you will need to configure the same accounts within SER. > > > > > > > > > > > > > > > Thanks > > > > > Emery > > > > > ----- Original Message ----- > > > > > From: "Sesha B" <sesha(a)iic.com> > > > > > To: "Director General: NEFACOMP" <dg(a)nefacomp.net>; > > <serusers(a)lists.iptel.org> > > > > > Sent: Wednesday, October 22, 2003 14:55 > > > > > Subject: RE: [Serusers]SIP Gateway > > > > > > > > > > > > > > > > Hi, > > > > > > I'm using Unit Manager Express version 1.1.1.5 to configure > > Mediatrix > > > > > 1204. > > > > > > I have configured it to use SIP. I am confused about setting up > > > accounts > > > > > on > > > > > > Mediatrix. Is there a way to set up accounts on mediatrix? I'm > > > testing > > > > it > > > > > > with one SIP phone. > > > > > > > > > > > > -----Original Message----- > > > > > > From: Director General: NEFACOMP [mailto:dg@nefacomp.net] > > > > > > Sent: Wednesday, October 22, 2003 4:55 AM > > > > > > To: sesha(a)iic.com; serusers(a)lists.iptel.org > > > > > > Subject: Re: [Serusers]SIP Gateway > > > > > > > > > > > > > > > > > > I used Mediatrix 1204. > > > > > > Have set up all the accounts that Mediatrix should use in > > registering > > > to > > > > > > SER? > > > > > > What software version are your Mediatrix using? You sometimes need > > to > > > > > > upgrade your firmaware. > > > > > > Also, the same Mediatrix can be H.323 or SIP. Are you sure that > > yours > > > is > > > > > > using SIP? > > > > > > > > > > > > > > > > > > Thanks > > > > > > Emery > > > > > > ----- Original Message ----- > > > > > > From: "Sesha B" <sesha(a)iic.com> > > > > > > To: <serusers(a)lists.iptel.org> > > > > > > Sent: Monday, October 20, 2003 23:02 > > > > > > Subject: RE: [Serusers]SIP Gateway > > > > > > > > > > > > > > > > > > > Hi, > > > > > > > > > > > > > > I'm using a SIP gateway (Mediatrix 1204) on 4 ports on PBX. I am > > not > > > > > able > > > > > > to > > > > > > > register the gateway on the SIP server. Is there a way I check > the > > > > logs > > > > > on > > > > > > > the SER to see if it is really registering or not? And can > someone > > > > > please > > > > > > > suggest why is the SIP Gateway not registering with the SER? > Thank > > > you > > > > > > very > > > > > > > much. > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Serusers mailing list > > > > > > > serusers(a)lists.iptel.org > > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > Serusers mailing list > > > > > serusers(a)lists.iptel.org > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >

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[Serusers] Click to dial
by Teppo Lunnas 27 Oct '03

27 Oct '03

Hello, Do you happen to know if Messenger 5.0 supports click-to-dial feature or is there any other free softphone that supports? br Teppo

2 1

[Serusers] SER and inter-domain routing
by Alexandru Deac 27 Oct '03

27 Oct '03

Hello, This might be a trivial issue, but I can't think of a solution to it. I have a SER installed on a linux machine acting as proxy for our VoIP network I'd like to configure so that it takes PSTN calls and forwards them to the proper GW for termination. However this is a 3rd paty GW over which we have no control. I've setup the ser to use rewritehostport("host", "port"); in order to send the call to the PSTN-GW but the call fails every time with a 404 Not found This i reported back by the PSTN GW. I belive this is due to the fact that the <To> field contains our realm but the remote GW cannot process it. Any idea how I can get ser to work with this? Has any one setup something similar? Any help would be appreciated, Alex.

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[Serusers] NAT & RTPPRoxy
by Howard March 27 Oct '03

27 Oct '03

Hi Gary, Did you get solution on how to change IP:port in the REGISTER packet to the extrenal NAT mappimg??? I'm also using RTC API 1.2. You help is apprecieated. Thankyou, BTW, anybody else has a solution for this?? Howard --------------------------------- Do you Yahoo!? Exclusive Video Premiere - Britney Spears

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[Serusers] error with mysql
by schweizer laurent 27 Oct '03

27 Oct '03

Hello, I have updated mysql to 4.0.X (client & server ) but now i have this error : Call to undefined function: mysql_pconnect() in /var/www/html/phplib/db_mysql.inc on line 73 Thanks

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[Serusers] [Fwd: nathelper with force_rtp_proxy and a working mysql]
by Jac Barben 27 Oct '03

27 Oct '03

-------- Original Message -------- Subject: nathelper with force_rtp_proxy and a working mysql Date: Mon, 27 Oct 2003 00:45:38 -0700 From: Jac Barben <jbarben(a)claritytel.com> To: Jan Janak <jan(a)iptel.org> Jan: Thanks for your last reply... I hope you didn't spend to much time on my last message... I since found that issue on my own. Jan, I've been playing with the published 0.8.11 per your suggestion and found that force_rtp_proxy is not part of the release. I've been monkeying all weekend with the latest developments from the CVS... but have been unable to get it all to work together. Right now I'm stuck on an apparent database issue: Oct 27 00:22:19 sip2 /usr/local/sbin/ser[31731]: INFO: signal 15 received Oct 27 00:23:08 sip2 /usr/local/sbin/ser[31738]: mod_init(): Can't bind database module Oct 27 00:23:08 sip2 /usr/local/sbin/ser[31738]: init_mod(): Error while initializing module usrloc Oct 27 00:34:26 sip2 /usr/local/sbin/ser[31772]: mod_init(): Can't bind database module Oct 27 00:34:26 sip2 /usr/local/sbin/ser[31772]: init_mod(): Error while initializing module usrloc "/var/log/messages" 88L, 7929C Any suggestions on a version of the code I could try with all elements working? I saw on the web some reference to a developer who'd used a "diff" file against the nathelper.c for the released version of 0.8.11... but I was unable to find it. I'd really like to try it all. What should I do next? J # # $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" alias="claritycolorado.com" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("nathelper", "natping_interval", 10) # ------------------------- 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; }; # Enable some NAT traversal help if (search("@192.*")) { setflag(1); force_rport(); fix_nated_contact(); }; # 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 (method == "INVITE") { if (isflagset(1)) { fix_nated_sdp("3"); force_rtp_proxy(); }; t_on_reply("1"); }; # 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") { if (!www_authorize("claritycolorado.com", "subscriber")) { www_challenge("claritycolorado.com", "0"); break; }; save("location"); break; }; # These would be destinations dialed from the # ATA devices bound for the Vega if (uri=~"^sip:303" || uri=~"^sip:720" || uri=~"^sip:213417" || uri=~"^sip:86") { t_relay_to_udp("64.140.77.59", "5060"); break; }; if (uri=~"^sip:1") { t_relay_to_udp("64.140.77.59", "5060"); break; }; # Force routes to Cisco # These would be numbers dialed from the Vega # or another ATA device if (uri=~"^sip:011") { if (uri=~"^sip:011") { strip(3); }; prefix("2826"); t_relay_to_udp("209.170.94.124", "5060"); break; }; if (uri=~"^sip:9011") { strip(4); prefix("2826"); t_relay_to_udp("209.170.94.124", "5060"); break; }; if (uri=~"^sip:91") { strip(1); prefix("2826"); t_relay_to_udp("209.170.94.124", "5060"); break; }; if (uri=~"^sip:81") { strip(1); t_relay_to_udp("198.79.3.246", "5060"); break; }; # These would be our IPBX numbers dialed from the Vega # or another ATA device if (uri=~"^sip:4729") { strip(4); }; if (uri=~"^sip:4728") { strip(4); }; # 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(); }; } onreply_route[1] { if (status=~"2[0-9][0-9]" && search("@192.*")) { fix_nated_contact(); fix_nated_sdp("3"); force_rtp_proxy(); }; }

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[Serusers] DHCP and SER
by Iizuka, Hirotaka, VF-JP 27 Oct '03

27 Oct '03

Hi, Does SER send query to the DNS to get SIP clients' address? Thank you. Hiro

2 1

[Serusers] A Question
by Alexandru Deac 27 Oct '03

27 Oct '03

Hi, I've been testing ser for the past few days and I ran into a little problem with search_append() and replace(). I have a string let's say: some_text If I do search_append("some_", "more_"); I get some_more_text which is great. However if I try to do a replace("some_more_text", "no_text'); it fails. Is this a known issue? Are there any workarounds? I have tried to verify the problem with the current CVS but I couldn't get it to start because all the modules that depend on a db fail to initialize. Thank you, Alex.

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Re: [Serusers] dbtext module
by Daniel-Constantin Mierla 24 Oct '03

24 Oct '03

Hello, comments inline. On 10/24/2003 2:43 PM, Annie Sasidar wrote: > > >Hi Daniel, >Thanks for the pointers. In your reply you have mentioned the following >.... >The user location information is stored in memory and duplicated to >database for a possible hot restart of SER. Did you restart the SER >after you registered the users and before trying to call? > >I am confused ... Should i restart SER after registering each user ? > Of course NOT!!! I just asked to see if the database was really involved into user location processing. Let me explain again, when a user registers all the contact information is stored into memory from where ser takes it for any further processing. Also, it is stored into table location of the database __only if you choose__ so in your config file -- by setting db_mode parameter of usrloc module to the value 1 or 2. This duplication, to database, helps when you, for some reasons -- let's say you have changed something in the config file, need to restart ser. In that moment, ser reloads all information about registered users prior the restart moment from database, so those users don't need to re-register. Again, you don't need to restart ser unless you modified the config file. Why I asked if you had restarted ser ... because in this case the users contact information would have been loaded from database and if dbtext wasn't working properly then some data might be damaged and some users were not able to be contacted. If you still have some doubts, let me know. Best regards, }Daniel > So far >i have not restarted SER after regisration. After registering user's i have >only tried to invite a user for a conversation. Please let me know your >thoughts. >Thanks, >Annie > > >

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sr-users October 2003