sr-users March 2010

sr-users@lists.kamailio.org

60 participants
117 discussions

by Bruno Machado

I have some questions regarding the shared memory. I'm working with the version 1.5.2 and its environment is quite simple. It does'nt storage any informations in the memory, like dialplan, contacts of subscribers, carrierroute, LCR and etc. The only purpose of this server is being a redirect server. Some calls have several possible destinations, reaching easily 12 contacts. The config.h's constant MAX_BRANCHES is set in 12, the default number. Ok. Like I said, almost the requests have more than 12 possible contacts. When it happens, Kamailio logs these messages: /usr/local/kamailio-1.5.2/sbin/kamailio[26188]: ERROR:core:append_branch: max nr of branches exceeded /usr/local/kamailio-1.5.2/sbin/kamailio[26188]: ERROR:pv:pv_set_branch: append_branch action failed Ok. It is not a problem. It is an error but it does not stop the system. The SIP messages still flowing normally. The server works for several days, 24/7 without functional errors, receiving five hundred thousand requests per day . But, usually at the weekends and the clock tells 2:30 am, the company's cell phone rings bringing a message: 'the system is down'. Kamailio shows: /usr/local/kamailio-1.5.2/sbin/kamailio[25492]: ERROR:core:add_avp: no more shm mem /usr/local/kamailio-1.5.2/sbin/kamailio[25492]: ERROR:exec:exec_avp: unable to add avp Kamailio stills running but it is not possible to route any request. All the new incoming calls generates the same error: 'no more shm mem'. I changed the size of shared memory, running Kamailio with '-m 64', but the server stopped again after some days. I see the options: - increase MAX_BRANCHES - increase again 'm' parameter (but it will just add more some days in the running time) - reduce the maximum number of contacts to 11 of the script that returns them to Kamailio So, my doubts are: - is it possible to have a memory leak when the message "max nr of branches exceeded" is showed? - changing '-m parameter' to a higher value, like 256, could affect the performance? tks friends bruno machado ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

14 years, 1 month

[Kamailio-Users] Media Bridge tips and hints

by Uriel Rozenbaum

Hi gentlemen, I'm having a bit of a hard time developing a bridge for media sessions. I have a box with 2 interfaces; eth0 has the private IP p.r.i.v and eth1 has public IP p.u.b.l. When I receive a message in the private IP I try sending the call to a public proxy we have (p.u.b.2) using Carrierroute module but the source IP used is p.r.i.v. It is obvious that p.u.b.2 will not be able to route back messages. # U 192.168.200.11:5060 -> p.r.i.v:5060 INVITE sip:1568767139@p.r.i.v SIP/2.0. Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK5d94bc09. From: "60911000" <sip:60911000@p.r.i.v>;tag=as6d477d46. To: <sip:1568767139@p.r.i.v>. Contact: <sip:60911000@192.168.200.11 <sip%3A60911000(a)192.168.200.11>>. ... # U p.r.i.v:5060 -> p.u.b.2:5060 INVITE sip:1568767139@p.u.b.2 SIP/2.0. Record-Route: <sip:p.r.i.v;lr=on;ftag=as6d477d46>. Via: SIP/2.0/UDP p.r.i.v;branch=z9hG4bKf531.440788e1.0. Via: SIP/2.0/UDP 192.168.200.11:5060 ;rport=5060;received=192.168.200.11;branch=z9hG4bK5d94bc09. From: "60911000" <sip:60911000@p.r.i.v>;tag=as6d477d46. To: <sip:1568767139@p.r.i.v>. Contact: <sip:60911000@192.168.200.11 <sip%3A60911000(a)192.168.200.11>>. ... I try then to set listen=0.0.0.0 so kamailio will be binded to all IPs but then the Route headers and Via use this 0.0.0.0 # U 192.168.200.11:5060 -> p.r.i.v:5060 INVITE sip:1568767139@p.r.i.v SIP/2.0. Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK14a00df7. From: "60911000" <sip:60911000@p.r.i.v>;tag=as502405cf. To: <sip:1568767139@p.r.i.v>. Contact: <sip:60911000@192.168.200.11 <sip%3A60911000(a)192.168.200.11>>. ... # U p.u.b.l:5060 -> p.u.b.2:5060 INVITE sip:1568767139@p.u.b.2 SIP/2.0. Record-Route: <sip:0.0.0.0;lr=on;ftag=as502405cf>. Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK711b.8d28549.0. Via: SIP/2.0/UDP 192.168.200.11:5060 ;rport=5060;received=192.168.200.11;branch=z9hG4bK14a00df7. From: "60911000" <sip:60911000@p.r.i.v>;tag=as502405cf. To: <sip:1568767139@p.r.i.v>. Contact: <sip:60911000@192.168.200.11 <sip%3A60911000(a)192.168.200.11>>. ... In this case the destination UA cannot return the call because it does not have a valid IP to do so. Is there any way to make Kamailio act in a different way? Like letting me modify the IP it will use for Via and Record-Route headers. Be aware that I'm supposed to do some kind of mirror for calls coming from outside the network (origin is public and destination is private). I'm sure I'll be asking some more on this same thread as I test. Thanks in advance! Uriel

14 years, 1 month

[Kamailio-Users] VUC Podcast: The SIP Router Project

by Daniel-Constantin Mierla

Hello, the recording of the VUC session last Friday about SIP Router Project is now available online at: http://www.voipusersconference.org/2010/kamailio3/ Here you find some details about who was in the call: http://www.kamailio.org/w/2010/03/vuc-listen-the-sip-router-project-podcast/ Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

14 years, 1 month

[Kamailio-Users] GSoC 2010 - Call for students

by Daniel-Constantin Mierla

Hello, if you are a student (or you know a student) interested in working with SIP Router over the summer, earning some money as well, please apply for GSoC Conferencing Presence support (or forward this email). The description of the project is done at: http://www.sip-communicator.org/index.php/GSOC2010/Kamailio4575 If you have technical details about the project please ask them on sr-dev(a)lists.sip-router.org. For GSoC related questions, please address to: gsoc(a)sip-communicator.dev.java.net. A good FAQ for applicants is available at: http://www.sip-communicator.org/index.php/GSOC2010/HowToApply Application must be done directly to the google site, link provided in the FAQ. I will be in charge of mentoring this particular project, helping you as much as possible to understand the current presence server architecture and the core API of sip router. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

14 years, 1 month

[Kamailio-Users] Centos 5 kamailio.init-file

by Detlef Pilzecker

Hi, does someone have a kamailio.init file for Centos5? The .../pkg/kamailio/[debian, fedora, ...]/kamailio.init files don't work! Thank you very much! Detlef Pilzecker

14 years, 1 month

[Kamailio-Users] Quick RTPproxy question

by Uriel Rozenbaum

Hi guys, I have some easy doubt about nathelper functions using RTPProxy. I'm trying to bridge from an external IP to an internal IP. The start-line for rtpproxy is: rtpproxy -l PUBLIC_IP/PRIVATE_IP -s udp: 127.0.0.1:7999 -F It starts OK and I see it when kamailio starts. I'm going to use something like Daniel showed on some other mail: if(dst_ip==private) force_rtp_proxy("ocfaei"); else force_rtp_proxy("ocfaei"); if i have the invite from a public IP to someone in private on the request I'll run force_rtp_proxy("ocfaei"); then the reply will be from private to public... should I run "force_rtp_proxy("ocfaei");"? or should it be the same? As allways, thanks for your help. Kind regards, Uriel

14 years, 1 month

[Kamailio-Users] Monitoring Upstream Carrier Health

by Geoffrey Mina

Hello, I am wondering if anyone has a clever way to remotely monitor a Kamailio 1.5 server. I am not looking for the standard monitoring, what I am looking to achieve is catching situations where my upstream carrier is having problems. We have a certain level of 404, 500, 503 errors throughout the day which are not indicative of a major carrier problem. I want to be able to monitor the ratio of properly setup calls to failed setups - so I can know when a carrier is having issues and is responding with many 503 errors. Any push in the right direction would be greatly appreciated. thanks.

14 years, 1 month

[Serusers] ser auth_db registration problem

by Anand Kumar

Hi All, i m trying to register softphone on ser using modparam("auth_db", "db_url", "mysql://baaaa:dddd@xx.x.x.x/ser") modparam("auth_db", "use_domain", 0) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") if(!www_authorize("", "subscriber")) { www_challenge("", "1"); break; } and found the following log but phone is not register . please help me... -- 2(28254) find_mod_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so] 2(28254) get_connection(): Connection found in the pool 2(28254) DEBUG: register_fifo_cmd: new command (DB) registered 0(28252) DEBUG: init_mod_child (1): dispatcher 0(28252) DISPATCHER:init_child #1 / pid <28252> 0(28252) SIP Request: 0(28252) method: <REGISTER> 0(28252) uri: <sip:123.49.54.226> 0(28252) version: <SIP/2.0> 0(28252) parse_headers: flags=1 0(28252) Found param type 232, <branch> = <z9hG4bK-d8754z-3d07d32ccc286b5c-1---d8754z->; state=6 0(28252) Found param type 235, <rport> = <n/a>; state=17 0(28252) end of header reached, state=5 0(28252) parse_headers: Via found, flags=1 0(28252) parse_headers: this is the first via 0(28252) After parse_msg... 0(28252) preparing to run routing scripts... 0(28252) parse_headers: flags=128 0(28252) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(28252) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(28252) parse_headers: flags=256 0(28252) end of header reached, state=9 0(28252) DEBUG: get_hdr_field: <To> [39]; uri=[sip:123456@123.49.54.226<sip%3A123456(a)123.49.54.226> ] 0(28252) DEBUG: to body ["123456"<sip:123456@123.49.54.226<sip%3A123456(a)123.49.54.226> > ] 0(28252) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 0(28252) DEBUG: get_hdr_body : content_length=0 0(28252) found end of header 0(28252) find_first_route: No Route headers found 0(28252) loose_route: There is no Route HF 0(28252) grep_sock_info - checking if host==us: 13==13 && [123.49.54.226] == [123.49.54.226] 0(28252) grep_sock_info - checking if port 5060 matches port 80 0(28252) grep_sock_info - checking if host==us: 13==13 && [123.49.54.226] == [123.49.54.226] 0(28252) grep_sock_info - checking if port 5060 matches port 80 0(28252) check_self: host != me 0(28252) parse_headers: flags=-1 0(28252) DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff 0(28252) parse_headers: flags=-1 0(28252) parse_headers: flags=60 0(28252) t_lookup_request: start searching: hash=57920, isACK=0 0(28252) DEBUG: RFC3261 transaction matching failed 0(28252) DEBUG: t_lookup_request: no transaction found 0(28252) DEBUG: mk_proxy: doing DNS lookup... 0(28252) check_via_address(122.162.142.65, 192.168.1.2, 0) 0(28252) DEBUG: add_to_tail_of_timer[4]: 0x2acf403e64e0 0(28252) DEBUG: add_to_tail_of_timer[0]: 0x2acf403e6500 0(28252) SER: new transaction fwd'ed 0(28252) DEBUG:destroy_avp_list: destroying list (nil) 0(28252) receive_msg: cleaning up 1(28253) DEBUG: timer routine:4,tl=0x2acf403e64e0 next=(nil) 1(28253) DEBUG: retransmission_handler : request resending (t=0x2acf403e6308, REGISTER ... ) 1(28253) DEBUG: add_to_tail_of_timer[5]: 0x2acf403e64e0 1(28253) DEBUG: retransmission_handler : done 1(28253) DEBUG: timer routine:5,tl=0x2acf403e64e0 next=(nil) 1(28253) DEBUG: retransmission_handler : request resending (t=0x2acf403e6308, REGISTER ... ) 1(28253) DEBUG: add_to_tail_of_timer[6]: 0x2acf403e64e0 1(28253) DEBUG: retransmission_handler : done 1(28253) DEBUG: timer routine:6,tl=0x2acf403e64e0 next=(nil) 1(28253) DEBUG: retransmission_handler : request resending (t=0x2acf403e6308, REGISTER ... ) 1(28253) DEBUG: add_to_tail_of_timer[7]: 0x2acf403e64e0 1(28253) DEBUG: retransmission_handler : done 1(28253) DEBUG: timer routine:7,tl=0x2acf403e64e0 next=(nil) 1(28253) DEBUG: retransmission_handler : request resending (t=0x2acf403e6308, REGISTER ... ) 1(28253) DEBUG: add_to_tail_of_timer[7]: 0x2acf403e64e0 1(28253) DEBUG: retransmission_handler : done 1(28253) DEBUG: timer routine:7,tl=0x2acf403e64e0 next=(nil) 1(28253) DEBUG: retransmission_handler : request resending (t=0x2acf403e6308, REGISTER ... ) 1(28253) DEBUG: add_to_tail_of_timer[7]: 0x2acf403e64e0 1(28253) DEBUG: retransmission_handler : done Anand Dayal -- Anand Dayal

14 years, 1 month

[Serusers] proxy health

by Sabatella, Michael

So I sent an email out a few weeks ago regarding the stateful stats in ser 0.9.6. Stateful Statistics Current: 6 (200 waiting) Total: 6387581 (0 local) Replied localy: 6869160 The number waiting is slowly going up. Can somebody point me to a developer of ser I can talk to regarding this? So far there is no affect to any of my users but I want to make sure this does not get out of hand. Thanks Mike ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

14 years, 1 month

[Kamailio-Users] Reminder: VUC Today, SIP Router Project

by Daniel-Constantin Mierla

Hello, a kind reminder for today's audio conference scheduled for 16:00GMT. Check the start time for your zone at: http://vuc.me/next After presenting the achievements so far within SIP Router, with what is new in 3.0.0 release and development version, you have an unique opportunity to ask questions to: - Andrei Pelinescu-Onciul, the creator of SIP Express Router (SER), the architect behind the core (transport layers, memory management, asynchronous processing, timers, etc), who will be also able to answer anything from project's history started in 2001 - Alex Balashov, Kamailio management team member, experienced consultant in building large SIP platforms - myself, as co-founder of Kamailio (OpenSER) You can join via irc on #vuc (note that some of us started to hang out on #sip-router) channel at irc.freenode.net. Dialing to VUC is possible via sip, skype, pstn, web page and more, see: http://vuc.me Five hours to go right now, hear you then! Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

14 years, 1 month

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sr-users March 2010