sr-users January 2009

sr-users@lists.kamailio.org

85 participants
110 discussions

by Alex R.S.M

Hi, What might be the cause that t_check_trans() function fails and ACK message is not being routed? This is a simple SIP to SIP call. if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } _________________________________________________________________ Windows Live™ Hotmail®: Chat. Store. Share. Do more with mail. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_hm_justgotbetter_explor…

15 years, 3 months

[Kamailio-Users] Run Server in background with maximum debug to log file

by Ali Jawad

Hi I am running the server with debug=9 and fork=yes and log standard output = yes. when I do that I can see a lot of debug info and sip packets in the standard output however the log file does show only some notices and xlog entries. So when I close putty and get back later or a few days later I have to kill and restart openser to see debug info as the log file is not showing maximum debug levels unlike the standard output. Please advice, the goal I want to reach is to see maximum loggin in a file so I can check it out without having to restart the server. Thanks

15 years, 3 months

[Serusers] [somewhat OT] seeking ideas/input for my thesis

by sp4rc

Hello VoIP guys Sorry for being somewhat off-topic. At the moment I am studying informatics in the seventh semester and I need to start thinking about my thesis. As I am very interested in VoIP technologies I thought about picking this as my main topic. So far I have only little experience in this area. I have been fiddling around with siproxd and pfSense and have red the one or the other packet dump containing SIP and RTP traffic, had a look into codecs, STUN, etc... but very cursorily, and that's the reason why I am quite unsure on which track to go. I think I am quite familiar with many network protocols and devices... so here comes the question of the questions: What would be a great project for my thesis to work on in the VoIP field? What are topics that still need special development? The time frame should be around 300 hours but don't take this value too seriously... Any ideas are welcome! /sp4rc

15 years, 3 months

[Kamailio-Users] SNMPStats per request statistics, is that possible?

by Noel R. Morais

Hi Guys, Is possible to have per request statistics from the SNMPStats module? What I mean is if I can know how many INVITES, REGISTERS, SUBSCRIBERS, NOTIFIES and so on were processed by kamailio. is that possible? Cheers, Noel

15 years, 3 months

[Kamailio-Users] new feature: run route blocks on timer

by Daniel-Constantin Mierla

Hello, a new module was committed in trunk, to be released as 1.5.0 - the plan at: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x The rtimer module allows to execute route block on timer. It can create new processes that will do the timer execution of the routes or use existing timer. There can be multiple timers and each timer can execute multiple route blocks.See readme at: http://kamailio.org/docs/modules/devel/rtimer.html Such features is useful if you want to do periodic jobs in your service 1) clean a database were you store temporary data - e.g., delete entries older than 2 hours modparam("rtimer", "timer", "name=ta;interval=10;mode=1;") modparam("rtimer", "exec", "timer=ta;route=8") route[8] { sql_query("ca", "delete from tmp where timeval<$TS-3600","ra"); } 2) save statistics in database so you can build graph, etc... http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:devel#statistics route[8] { sql_query("ca", "insert into kamailio_stats values ($stat(...), ...)","ra"); } 3) Along with another pretty new feature - sending sip requests from config file: http://lists.kamailio.org/pipermail/users/2009-January/021426.html you can implement keep-alive/syncronization mechanisms within your farm of sip routers. Any other ideas? Let's see the best one for using the new stuff from uac and the rtimer module... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

15 years, 3 months

[Kamailio-Users] new features in htable module

by Elena-Ramona Modroiu

Hi, htable module has been enhanced with couple of features that give more flexibility in working as config file cache system. - auto-expire - to each value stored in hash table can be assigned an expire value. If no update was done in that time interval, the value will be automatically removed. One new pseudo-variable class has been added to manage the expire values: $shtex(htable=>key) - it can be used to get/set the time-out interval. To get this feature on, the hash table definition must include autoexpire attribute greater than 0. - sht_rm_name_re(htable=>regexp) - delete from hash table all entries that match the name against regexp - sht_rm_value_re(htable=>regexp) - delete from hash table all entries that match the value (string) against regexp - reload the content of a hash table from database at runtime upon MI command. The hash table definition must include the database table name, then you can issue: 'kamctl fifo sht_reload htname' to reload the hash table with name 'htname'. Check the documentation at: http://kamailio.org/docs/modules/devel/htable.html Regards, Ramona

15 years, 3 months

[Kamailio-Users] Registered user

by michel freiha

Dear All, I need to ask please about which function should I use in order to check while making a call if the user who is dialing the number is making the call from a registered account or not? Regards

15 years, 3 months

[Kamailio-Users] new feature: send sip requests from config file

by Daniel-Constantin Mierla

Hello, uac module provides a new function that allow sending sip requests from configuration file: - uac_send_req() http://kamailio.org/docs/modules/devel/uac.html#id2467912 It opens the door for a new bunch of things you can do now. I can think of now: 1) send IM alerts on special events route { $uac_req(method)="MESSAGE"; $uac_req(ruri)="sip:alert@kamailio.org"; $uac_req(furi)="sip:server@kamailio.org"; $uac_req(hdrs)="Content-Type: text/plain\r\n"; $uac_req(body)="SIP request from:" + $si + ":" + $sp; uac_send_req(); } 2) turn lamps on for some sip phones on some events - e.g., snom: http://wiki.snom.com/Features/LED_Remote_Control 3) talk server-to-server - here one example can be building a platform with a central server for accounting Accounting is done at 200ok. So each routing server will build a csv list (time,callid, from tag, to tag, invite from uri, invite r-uri): onreply_route[1] { if(status!="200") return; $uac_req(method)="ACCOUNTING"; $uac_req(ruri)="sip:store@accounting.kamailio.org"; $uac_req(furi)="sip:server@server1.kamailio.org"; $uac_req(hdrs)="Content-Type: text/accounting-csv\r\n"; pv_printf($uac_req(body), "$TS,$ci,$ft,$tt,$T_req($fu),$T_req($ru)"); uac_send_req(); } note that $T_req() is given by tm module, also with Kamailio trunk: http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:devel#tm_module_p… The accounting server: route{ # you should check the source IP also if(method=="ACCOUNTING" && $rU="store") { sql_query("ca", "insert into accounting(timeval,callid,ftag,ttag,src,dst) values ('$(rb{s.select,0,,})','$(rb{s.select,1,,})', ...)", "ra"); send_reply("200", "Stored"); } } ... and put your imagination work ... there are plenty of new and cute features you can add on your service with this functionality ... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

15 years, 3 months

[Kamailio-Users] TCP ZeroWindow and TCP Window Full

by Gawith

Hi, I see enormous TCP problems on my systems. All of a sudden, a perfectly working TCP session uses ZeroWindow packets, later I see TCP Full Window packets and finally the peer resets the connection. In the logs I see a lot of "core:handle_io: empty fd map" messages. Has anyone experienced similar problems? Regards.

15 years, 3 months

[Kamailio-Users] SIP Digest Access Authentication RELAY survey

by Victor Pascual Ávila

Hi, excuse me if this message is not directly related to Kamailio. I'm just wondering if folks could share with me if (and how) they have prevented the "SIP Digest Access Authentication RELAY" in their networks (and what worked for them or not). NAT boxes reduce dramatically the scenarios for a successful attack. Otherwise, some might be mitigating the attack by means of forcing UAs to use outbound proxies while others might be reducing the attack incentives by means of message integrity. Any comment would be appreciated, -- Victor Pascual Ávila

15 years, 3 months

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sr-users January 2009