sr-users August 2006

sr-users@lists.kamailio.org

184 participants
360 discussions

[Serusers] Simple Request/Response Forwarding

by Sebastian Gabris

Hello everybody, i'm new to SER and try to write my first config file. Since my english is not that good and i dont have much time, i want to minimize my reading of SER tutorials. This is my VoIP - test enviroment: UA(private/local IP) <-> (private/local IP)SER(public IP) <-> (public IP) VoIP - Provider I want SER to simply relay/forward all incoming and outgoing calls. I dont want that SER should have a SIP -domain, no authentication or registration. RTP - Streams should also be forwarded over the Proxy. No my questions: What is the simplest way to implement this? As far as i know i need the rtpproxy module. Do i need the nathelper or any other important modules? I also figured out to use the mhomed option. Has someone a sample config file for such a scenario? Any Documentation in German for the config file? Thanks in advance and any Help & Tipps would be very appreciated. _____________________________________________________________________ Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! http://smartsurfer.web.de/?mc=100071&distributionid=000000000071

18 years, 3 months

[Users] OpenSER v1.1.0 notls in Debian unstable official repository

by Daniel-Constantin Mierla

Hello, thanks to the work of Julien BLACHE and pkg-voip-maintainers group from Debian, OpenSER v1.1.0 is available via official Debian APT repositories with the unstable distribution. http://packages.debian.org/unstable/net/openser People liking to use default repositories can remove the openser.org site from sources.list. Beware that the version included in Debian does not include the TLS support -- integration of GPL and OpenSSL license to satisfy Debian policies being in progress. If you want to use the TLS version, stay with openser.org's repository for a while. Cheers, Daniel

18 years, 3 months

[Users] Rewriting Contact HF port

by Justin Pearce

I am wondering if there is a good way to rewrite the port in the contact header in OpenSER 1.0.1 before it is stored in the location table?(I'm using the MySQL database backend). Thus far, I am using "subst(/^(Contact.+):[0-9]+(.+)$/\1:5060\2/ig);" just before I do "save("location");", but that does not seem to be working. Suggestions? Justin Pearce Information Technology/Programming Price Video Productions JustinP(a)PriceVideo.com 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com

18 years, 3 months

FW: [Serusers] using SER for SIP conversion from H323

by Richard C. Thompson

I know this may sound like heresy but a Quintum call relay does a great job doing this. We use a number of them with our SER and asterisk servers. Rick _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Torres, Javier Sent: Wednesday, August 02, 2006 12:19 PM To: serusers(a)lists.iptel.org Subject: [Serusers] using SER for SIP conversion from H323 Hello, I was wondering if anyone was using this in conjunction with Cisco Call manager to get out to a voip network. Call manager is horrible at SIP so I was hoping to have a h323 connection from Call manager to SER, the SER box would take that connection and forward it with SIP out to a voip peering fabric we are testing out. I know asterisk can be used in this fashion but had some voice quality issues and figured this piece of software might be better suited. Just hoping someone has done this and can confirm and maybe point me in the right direction for a how-to? Thanks, Javier

18 years, 3 months

Re: [Serusers] registered UA's

by Jiri Kuthan

Well, I think there is nothing else better than troubleshooting and looking at what it is message by message. I agree NAT issues may be worth looking at. Other problem we experience frequently is that we drive registration period low, indicate that using Expires header-field, and client is silly not to honor it. -jiri At 17:14 03/08/2006, Weiter Leiter wrote: >My number one suspect would be the nat. >WL. > >On 8/3/06, Richard C. Thompson <<mailto:rthompson@vir2com.com>rthompson(a)vir2com.com> wrote: > >Hi All > > > >I hope someone can help me figure this out. We've had a ser server running for quite some time now on a Dell 2850 with dual cpu's. The load is very low due to the use of multiple mediaproxy servers to handle the RTP streams. We have about 2300 UID in the mysql database but at any given time only about 1300 show as registered in the location table and we have customers complaining about not receiving calls. When tech support has a customer power cycle their ATA, it registers just fine and they start receiving calls. What could cause this? From the beginning, I noticed this was happening but didn't give it much thought as the number of UIDs was low. But now it is a growing problem. SER version is 0.9.0 > > > >Any hints or help would be appreciated. > > > >Rick > > > > > > > >_______________________________________________ >Serusers mailing list ><mailto:Serusers@lists.iptel.org>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 -- Jiri Kuthan http://iptel.org/~jiri/

18 years, 3 months

[Serusers] SIP MESSAGE method

by sip

At some point in the not too distant past, I used to bounce SIP MESSAGEs back and forth from UA to UA on our server. Now, it no longer seems to work (get an error 513 -- message too big error... which is kind of a default for all unhandled SIP methods), and I'm not entirely sure how to go about fixing it. Can someone explain how a MESSAGE should be handled and maybe give me a hint as to why it used to work but does no longer (I know that's pretty vague, but just some ideas of what I might look at to point me in the right direction would be good)? Meanwhile I'll fiddle, but I don't seem to be getting anywhere. N.

18 years, 3 months

[Serusers] Q:ABOUT:DELAY OF BEGIN OF THE INVITE

by Alejandro Sanchez

MY scenario is the next. I have 2 video telephones that are conected to the PSTN and they has access trougth a dial up conection to the IP network and the problem is that the 2 video telephones shuld be register before one to other geretate the INVITE then i need make that when the first REGISTER arrive the sip server generate a delay for wait for the REGISTER of the other video telephone and both VT state register for the INVITE message.. Someone knows how can i do this? Thank's Alex. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

18 years, 3 months

[Users] t_newtran in BYE

by Kapil Dhawan

With evey Bye i am receiving t_newtran ERROR: t_newtran: transaction already in process DEBUG:sl:sl_reply_error: error text is I'm terribly sorry, server error occurred (1/SL) # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=8 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode 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/openser_fifo" fifo_db_url="mysql://root:root@localhost/openser" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # !! Accounting loadmodule "/usr/local/lib/openser/modules/acc.so" # !! Nathelper loadmodule "/usr/local/lib/openser/modules/nathelper.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) # -- acc params -- # set the reporting log level modparam("acc", "log_level", 3) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels", 1) modparam("acc", "db_url", "mysql://root:root@localhost/openser") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # !! Nathelper # -- nathelper params -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for presence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # labeled all transaction for accounting # setflag(1); # setflag(2); # 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 if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(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") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { rewriteport("5065"); route(1); # sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; t_on_failure("2"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); return; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } failure_route[2] { t_on_reply("1"); append_branch(); t_relay("udp:x.y.z.w:5065"); exit; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } _________________________________________________________________ Shah Rukh fan? Know all about the Baadshah of Bollywood. On MSN Search http://server1.msn.co.in/profile/shahrukh.asp

18 years, 3 months

[Users] how to get the core file...

by Jayesh Nambiar

Hi all, My openser 1.1.0 crashed suddenly while running. In the syslog I saw, child 28187 terminated due to signal 11, core generated, signal 15 received signal 15 received and so on. I did'nt get to know what exactly was the problem. Can somebody help me where to locate the core file and how to understand where was the problem?? pls help me. w/regards, jayesh --------------------------------- Heres a new way to find what you're looking for - Yahoo! Answers

18 years, 3 months

Re: [Users] OpenSER Administrator Release

by Mike Williams

Ram, No, unfortunately it is not advanced enough yet to create configuration files. It basically manages database data, like creating users, groups, group members, accounting, and so on. It does no actual configuring of OpenSER. Mike On Friday 04 August 2006 10:42, you wrote: > Hi > > Looks good Job, still need to install and check my site > before i take intiative > > is this GUI Like AMP of Asterisk > > can create all configfiles ? > > Ram > > On 8/4/06, Mike Williams <mwilliams(a)etc1.net> wrote: > > All, > > > > I've just finished my first release of OpenSER Administrator, a web > > interface > > for managing OpenSER. It's GPL'd and written in Ruby on Ralis, and can be > > downloaded from openseradmin.sourceforge.net > > > > Please let me know what you think, and let me know of any features you > > want, > > bugs you find, etc. > > > > Thanks, > > > > Mike Williams > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users

18 years, 3 months

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sr-users August 2006