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37462 discussions

[Users] release 1.1.0 status
by Daniel-Constantin Mierla 07 Jul '06

07 Jul '06

Hello, just to give you an update about the status of the new release. It should be out in terms of days. Between developers, it was scheduled to happen at the beginning of this week, but the feedback we got in the last week made us to postpone it a bit. New issues were reported during mentioned period and we were hunting them, willing to have the release free of known bugs. Last week issues: - possible memory leak in TLS (lot of testing and debugging) - memory leak in tm (fixed) - crash in register when using path extension (fixed) - crash in pike (fixed) - crash in usrloc when user updated contact after migrating from non-NAT to NAT environment (fixed) - investigate "detached timer" warning (low priority) - is_avp_set() fails when to check AVPs with index (patch made, testing in progress) Cheers, Daniel

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[Users] remove_hf("Contact") after fix_nated_contact()
by Walter Schober 07 Jul '06

07 Jul '06

Hello! Just curious: I do remove_hf("Contact") right after fix_nated_contact() (for whatever reason). Then the remove_hf() removes anything but the new SIP URI from the message, leaving e.g. sip:user@received_ip:port in front of the headerfield following the Contact. e.g. Before: Some-Header: bla Contact: Name <sip:user@privateip:port> Authorization: bla Then: Some-Header: bla sip:user@public_ip:portAuthorization: bla How to remove that Contact: after having it nat_helped and then saved? br Walter

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[Users] Suggestion Required
by Abdul Lateef khan 07 Jul '06

07 Jul '06

Hi friends, I am new in OpenSER, I want to use it only for SIP Proxying with freeradius. I made plan to install openser with freeradius on Virtual Server to get only 100 cuncurent calls. 1- Is it possible to install on Virtual Server? 2- Which Codecs or used, because i want to calculate the bandhwidth according to the codec? 3- What RAM should be used to handel 100 cuncurent calls? 4- It can accept h323-credit-time from radius to control max credit call time? I will appriciate for your kind of suggestion. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

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[Serusers] Exec_msg ERROR
by Michel Zenone 07 Jul '06

07 Jul '06

Hi! Since I have added an exec_msg in my ser.cfg I have an error in my logs: Jul 7 14:59:31 localhost /usr/local/ser_directory/sbin/ser[25318]: ERROR: exec_msg: cmd echo hello: [$SIP_TID] from $SIP_SRCIP to $SIP_RURI >> /root/memos/ser.txt; failed. exit_status=-1, errno=32: Broken pipe My 'ser.cfg' is : # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 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 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/ser_directory/lib/ser/modules/mysql.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/sl.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/tm.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/rr.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/textops.so" # Modules allowing digest authentication loadmodule "/usr/local/ser_directory/lib/ser/modules/auth.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/auth_db.so" # Modules allowing to check groups and identity: loadmodule "/usr/local/ser_directory/lib/ser/modules/group.so" loadmodule "/usr/local/ser_directory/lib/ser/modules/uri_db.so" # Module allowing shell command execution loadmodule "/usr/local/ser_directory/lib/ser/modules/exec.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) #Added on July4 modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_timer", 10 ) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { log(1,"truc"); append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri=~"192.168.0.120") { append_hf("P-hint: outbound\r\n"); route(2); break; }; if (uri=~"192.168.0.120") { if (method=="REGISTER") { if (!www_authorize("192.168.0.120", "subscriber")) { www_challenge("192.168.0.120", "0"); break; }; save("location"); break; }; lookup("aliases"); if (method == "INVITE") { log(1,"invite"); if ((uri=~"^sip:0[0-9]*@192.168.0.120")&&(!is_user_in("From","int"))) { sl_send_reply("403", "You can't go out!!"); break; }; if (!((does_uri_exist())|(uri=~"^sip:0[0-9]*@192.168.0.120"))) { sl_send_reply("403", "Who is it??"); break; }; }; if (!uri=~"192.168.0.120") { append_hf("P-hint: outbound alias\r\n"); route(2); break; }; if (!lookup("location")) { log(1,"lookup"); forward(192.168.0.120,5070); break; }; }; log(1,"hello!!"); exec_msg("echo hello: [$SIP_TID] from $SIP_SRCIP to $SIP_RURI >> /root/memos/ser.txt;"); append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { log(1,"route1"); t_on_failure("1"); t_relay(); } route[2] { log(1,"route2"); if (!t_relay()) { sl_reply_error(); }; } failure_route[1] { log(1,"failureroute"); rewriteport("5070"); append_branch(); t_relay_to_udp("192.168.0.120","5070"); } I don't know what to do...Can somebody help me? Thanks in advance! Michel

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[Serusers] Remote sip and rtp analyzing
by İlker Aktuna (Koç.net) 07 Jul '06

07 Jul '06

Hi, I am looking for a solution to analyze sip and rtp packets on a Ser + Mediaproxy server. SIPAlanyzerV6 seems to receive remote capturing, but there is no information on its manuals for remote capture. What is the best solution to analyze sip and rtp packets remotely from a SER server ? I also would like to have some statistical output, if possible. Thanks for your comments. ilker <http://387555.sigclick.mailinfo.com/sigclick/070A0404/03044400/0604014D/041…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Users] Dispatcher Module Question
by Dominik Bay 07 Jul '06

07 Jul '06

Hi all, how can i remove/add and reload the dispatcher module when it is in service? I've to servers with openser + dm which are dispatching the traffic to two different OpenSERs which handle the registrations etc. If one of this two servers dies, i want to have this one remove from the dispatch Servers. How can i do that? TIA -- Mit freundlichen Grüßen / Kind regards Dominik Bay Cablesurf Technik

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Re: [Users] Radius
by Cseke Tamas 07 Jul '06

07 Jul '06

Hi, I ve added /etc/openser/dictionary.radius to /etc/radiusclient-ng/dictionary and to BSDradius dictionary file too Largo wrote: > Hi > > Try to use the file provided in the folder "/usr/local/etc/openser" > called "dictionary.radius" as a dictionary. > > Bye > > Cseke Tamas schrieb: > >> Hello, >> >> We 'd like to use openser with radius authentication, accounting >> >> We have installed openser from debian package on an ubuntu system: >> deb http://www.openser.org/debian sarge main >> >> But radius authentication failed, and we need avp_radius module too, >> wich is not included in this package i guess >> >> error message in the log: >> ERROR:auth_radius:radius_authorize_sterman: rc_auth failed >> >> on http://openser.org/docs/openser-radius-1.0.x.html i ve 'read that >> openser should be compiled (with some macros defined in makefile) to >> work with >> radius. >> Is there any possibility to install some deb packages instead of >> compilation? >> Which package should we install? >> >> Thanks, >> Tamas >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Serusers] How to do REGEXP operation of mysql with db-api?
by wu yaqian 07 Jul '06

07 Jul '06

Hello,erery one, I want to use mysql's regexp operator in ser,but with db-api ,I can't do this.Because the patten is stored in the mysql table.The db-api can only do key op value.I want to do the value REGEXP key operation, Any one has advices to me? Thanks very much.

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Re: [Serusers] IP address spoofing
by Greger V. Teigre 07 Jul '06

07 Jul '06

>Hi Greger. SER and Asterisk can be configured to use TCP for SIP/SDP >messages? > Try this before forwarding the INVITE to Asterisk: if(!uri_param("transport")) { add_uri_param("transport=tcp"); } >The reason I said that the caller wouldn't receive audio is because the >callee's RTP stream would be directed to SER, not the caller. >-- Nick How come? I don't see rtpproxy or mediaproxy in your setup. So, unless you force an RTP proxy in your SER, I would expect Asterisk and caller to communicate directly on RTP. g-) Nick Hoffman wrote: >> Nick Hoffman wrote: >> >>> Hi guys. Say you have this setup, with an account for the caller on >>> both Asterisk and SER: >>> Caller -> SER -> Asterisk -> VoIP Provider -> Callee >>> >>> If the caller were to spoof SER's IP address and place a call directly >>> to Asterisk (thus circumventing SER), what would happen? >>> >>> If the call was in fact setup, obviously the caller would not receive >>> any audio from the callee. However, would the call be setup? When >>> Asterisk responds to the caller's request and sends SIP packets back >>> (to SER), would SER say "I don't know anything about this call! >>> Asterisk, kill this call please."? >>> >>> Thanks for your input! >>> -- Nick >>> e: nick.hoffman(a)altcall.com >>> p: +61 7 5591 3588 >>> f: +61 7 5591 6588 >>> > > > On Wed July 5 2006 17:58, "Greger V. Teigre" <greger(a)teigre.com> wrote: > >> Depends on the config and what type of message the caller managed to >> make asterisk create... You could (and probably should) put asterisk on >> a private routable network (i.e. NATed behind a firewall). The best >> would be to put ser and asterisk on the same network and only allow >> outside world to contact ser and let ser contact asterisk using the >> private address of asterisk. Alternatively you could use tcp to >> asterisk and stop udp traffic. >> Why the caller wouldn't receive audio, I don't understand... >> g-) >> > > > Hi Greger. SER and Asterisk can be configured to use TCP for SIP/SDP > messages? > > The reason I said that the caller wouldn't receive audio is because the > callee's RTP stream would be directed to SER, not the caller. > -- Nick > e: nick.hoffman(a)altcall.com > p: +61 7 5591 3588 > f: +61 7 5591 6588 > > If you receive this email by mistake, please notify us and do not make any > use of the email. We do not waive any privilege, confidentiality or > copyright associated with it. > > >

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[Users] rtpproxy - fax
by Hakan YASTI 07 Jul '06

07 Jul '06

Dear All, I use openser 1.0.1 and use rtp proxy for some cases other than nat option. At voice calls I do not have any problem. But When I try to send fax,it is unsuccesful. I investigated the logs and saw that, after the reinvites ( to take the session from g729 to G711u bypass mode ), the last OK message's SDP that is sending from openser to user agent is not correct.Etheral logs are attached. Is it a bug of rtpproxy ? Any suggestion. Best Regards, Hakan.

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