- sr-users - kamailio.org

[Serusers] Check out my photos on Facebook
by Sarfaraz Chougule 24 Mar '10

24 Mar '10

Hi, I set up a Facebook profile where I can post my pictures, videos and events and I want to add you as a friend so you can see it. First, you need to join Facebook! Once you join, you can also create your own profile. Thanks, Sarfaraz To sign up for Facebook, follow the link below: http://www.facebook.com/p.php?i=520931674&k=54EUZ6Q3QT2BXJMFYDYVPVWZWUKJ64&r Already have an account? Add this email address to your account http://www.facebook.com/n/?merge_accounts.php&e=serusers%40iptel.org&c=71a7… was invited to join Facebook by Sarfaraz Chougule. If you do not wish to receive this type of email from Facebook in the future, please click on the link below to unsubscribe. http://www.facebook.com/o.php?k=679489&u=557896866&mid=213a1b7G2140d4a2G0G8 Facebook's offices are located at 1601 S. California Ave., Palo Alto, CA 94304.

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[Kamailio-Users] Establish Call on different network.
by john david 24 Mar '10

24 Mar '10

Hello all, I am using kamailio server, it works properly when connection is in between two clients between same lan. But when I connecting between two client who are on different proxy and behind different NAT, call has been established but no audio or video get transferred. What are the changed i suppose to do, or any parameters should get enabled while compilation, Thanks in advance.

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[Kamailio-Users] dinner in berlin, thursday, march 25, 19:00
by Daniel-Constantin Mierla 23 Mar '10

23 Mar '10

Hello, couple of developers from SIP and VoIP area are in Berlin this week and plan to meet for dinner and beer Thursday, March 25, 19:00 at the traditional place by now: Lemke Brauhaus Luisenplatz 1, 10585 Berlin Across the corner with Charlottenburg Castle http://www.brauhaus-lemke.com/index.php?area=4 If you are around, just pop up. You can write me if you need more details about how to get there. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] Shared memory
by Bruno Machado 23 Mar '10

23 Mar '10

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

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[Kamailio-Users] Media Bridge tips and hints
by Uriel Rozenbaum 23 Mar '10

23 Mar '10

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

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[Kamailio-Users] VUC Podcast: The SIP Router Project
by Daniel-Constantin Mierla 22 Mar '10

22 Mar '10

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/

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[Kamailio-Users] GSoC 2010 - Call for students
by Daniel-Constantin Mierla 22 Mar '10

22 Mar '10

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/

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[Kamailio-Users] Centos 5 kamailio.init-file
by Detlef Pilzecker 22 Mar '10

22 Mar '10

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

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[Kamailio-Users] Quick RTPproxy question
by Uriel Rozenbaum 22 Mar '10

22 Mar '10

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

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[Kamailio-Users] Monitoring Upstream Carrier Health
by Geoffrey Mina 22 Mar '10

22 Mar '10

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.

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[Serusers] ser auth_db registration problem
by Anand Kumar 19 Mar '10

19 Mar '10

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

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[Serusers] proxy health
by Sabatella, Michael 19 Mar '10

19 Mar '10

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.

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[Kamailio-Users] Reminder: VUC Today, SIP Router Project
by Daniel-Constantin Mierla 19 Mar '10

19 Mar '10

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/

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[Kamailio-Users] Kamailio - SIP Router on GSoC 2010
by Daniel-Constantin Mierla 19 Mar '10

19 Mar '10

Hello, I have the pleasure to announce that our project has an accepted application for Google Summer of Code 2010. The proposed application was done within SIP Communicator Organization and is related to extending the presence server for conference calls notifications. See more: http://www.sip-communicator.org/index.php/GSOC2010/Kamailio4575 http://www.sip-communicator.org/index.php/Development/Gsoc2010 The friends from SEMS project will be part as well, with separate application, implementing the audio mixer part. More details will be published soon. Start thinking about good candidates (they must be students (college or university)), we need them to get the project accepted for implementation. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] uac_replace_from unexpected behavior
by Brandon Armstead 18 Mar '10

18 Mar '10

Hello, As always thank you ahead of time for your help and input! I am currently calling uac_replace_from("", "") in effort to "leave uri" and "toss away display name" Which does seem to work....... for the initial INVITE However upon receiving an ACK with an empty display, however "" <- quotations, it does not clear the display "" which is causing issues with one of my upstream vendors. Example / Scenario: From: "" <sip:uri@host> Expected Result upon uac_replace_from("",""): From: <sip:uri@host> Current Result: From: "" <sip:uri@host> As you can see it is not stripping the "" empty display quotes. Any thoughts / ideas / suggestions to get my desired affect? Sincerely, Brandon Armstead

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[Kamailio-Users] Rtpproxy/Kamailio modification to support high capacity encryption, transcoding
by Vikram Ragukumar 18 Mar '10

18 Mar '10

Hello, We're making initial modifications to rtpproxy to support high channel capacity transcoding and encryption. At this point we want to get some general idea of the scope of changes needed for rtpproxy and Kamailio... so we're starting with small steps. We've been studying rtpproxy source and our current thinking is to add a sub-structure to the existing rtpp_session structure (defined in rtpp_session.h). The new sub-structure would include: -encryption options (type, key length, salt size, type of key mgt protocol, etc) -encode / decode options (type, VAD/CNG, VIF size, etc) Any comments or advice on this approach appreciated. Not sure whether to start a separate thread, but also there is the issue of what changes are necessary to Kamailio to support sending updated commands to rtpproxy. Would modifying Nathelper alone be sufficient? Thanks and Regards, Vikram. PS : I'm posting on Kamailio's mailing list because it seems that both Kamailio and rtpproxy developers closely follow this list.

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[Kamailio-Users] SIP Router Project of VUC, March 19
by Daniel-Constantin Mierla 18 Mar '10

18 Mar '10

Hello, this Friday, March 19, late afternoon, the weekly VoIP User Conference is hosting a session about SIP Router project. My goals are to present the achievements so far within SIP Router projects, what is new in Kamailio 3.0 release and plans for the future. More details can be found at: http://www.kamailio.org/w/vuc-the-sip-router-project/ You can join the audio conference via sip, skype, pstn line or other several options presented on http://vuc.me site. There is a irc channel available for it: #vuc on irc.freenode.net. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] kamailio logo artwork
by Daniel-Constantin Mierla 18 Mar '10

18 Mar '10

Hello, courtesy of Elio Rojano (http://www.sinologic.net), very nice logo artworks are available now, using openser or sip-router thematics. You can browse at: http://www.asipto.com/gallery/v/ksr-artwork/ Available for download as well at: http://www.kamailio.org/pub/ksr-artwork/ If you are good at graphic design and have new ideas, please submit. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] Kamailio 3.0.0 Install Error
by Nathaniel L Keeling 17 Mar '10

17 Mar '10

I am trying to install the new release of Kamailio 3.0.0 on Solaris 10. When I perform the install, I am getting this error. Here is the command that I am entering for the install: make prefix=/usr/local/kamailio-3.0.0 INSTALL=install group_include="standard standard-dep postgres" CPU=ultrasparc install install mode make[2]: Entering directory `/usr/local/src/kamailio-3.0.0/lib/kcore' make[2]: Nothing to be done for `install-if-newer'. make[2]: Leaving directory `/usr/local/src/kamailio-3.0.0/lib/kcore' touch /usr/local/kamailio-3.0.0/lib/kamailio/modules_k/speeddial.so install -m 755 speeddial.so /usr/local/kamailio-3.0.0/lib/kamailio/modules_k make[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/modules_k/speeddial' mkdir -p /usr/local/kamailio-3.0.0/etc/kamailio/ # other configs /bin/sh: syntax error at line 1: `;' unexpected make: *** [install-cfg] Error 2 I am new to Kamailio and would appreciate any help. Thanks Nathaniel

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Re: [Kamailio-Users] Is Timestamp supported in kamailio?
by Iñaki Baz Castillo 17 Mar '10

17 Mar '10

2010/3/16 mustafa rifaee <mustafa.rifaee(a)gmail.com>: > Hi all, > I would like to asking you Hi, it's just better to keep the thread in the maillist. > about the Time-stamp header in SIP messages, and > why this header does not supported in most open SIP servers, Is it > recommended or not? > Is this fact true ?? I think The problem is that both UAS and UAC must have > the same system time to generate a useful result. So therefore both or all > SIP devices (think about an intermediate proxy) must have NTP activated. That's not true. The request can contain a header like: TimeStamp: 25 and the proxy/server replies in the 100 Trying: TimeStamp: 25 0.001 This means that it has token 0.001 seconds to the server in order to process the request and generate the 100 response. -- Iñaki Baz Castillo <ibc(a)aliax.net> -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] [OT] MSN presence flow
by Iñaki Baz Castillo 17 Mar '10

17 Mar '10

Hi, does somebody have a MSN protocol flow related to presence rules or buddies management? This is, I would like to know how MSN protocol imlements some tasks as: - Adding a buddy. - Blocking a buddy for presence. - Blocking a contact (not a buddy) for presence. Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] kamailio crashes
by Panagiotis Skoulikaritis 17 Mar '10

17 Mar '10

Hello Daniel I do have quite a few core files, please send me the gdb commands. Regards Panagiotis. Daniel-Constantin Mierla wrote: > Hello Panagiotis, > > On 03/16/2010 02:21 PM, Panagiotis Skoulikaritis wrote: >> Hello Daniel >> >> I will need time to recreate the problem, >> attached are the only traces I kept. >> on the amaze-4.cap the calls are originated by a softphone registered >> on the kamailio >> on the crash.cap the calls are originated from the "PSTN". >> >> we do account the $pd and the $pn >> >> I reply to you directly since i don't want to give the traces on the >> mailing list, I hope you will understand. > > I asked to be sent private, it is ok. > >> >> Also for workaround Alex took out the PDT module from the route and >> the kamailio do not crash anymore. > > Hmm, so you say pdt module is related? That is pretty small and old > module... > > Do you still have the core file? I can send you some gdb commands to > get more details out of it. > > Thanks, > Daniel > >> >> Regards >> >> Panagiotis >> >> >> Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> can you send me ngrep/pcap file with ip addresses so I can match >>> which 200ok is causing the problem (coming from B or coming from >>> Asterisk)? The backtrace shows ip while the sip trace is masked. >>> >>> Also, I would need a bit more info from the core file. Please keep >>> one around. The issue seems to be related to P-Asserted-Identity >>> header, but I couldn't find any such header in the sip trace you sent. >>> >>> Are you accounting the PAI header? >>> >>> Thanks, >>> Daniel >>> >>> On 03/10/2010 03:30 PM, Panagiotis Skoulikaritis wrote: >>>> Dear Marius >>>> >>>> The scenario is as follows: >>>> >>>> 1. A Call is placed by a sip subscriber "A" >>>> 2. kamailio forwards the call to the asterisk server >>>> 3. Asterisk plays an IVR message on the subscriber "A", creates a >>>> new call to a "virtual" number which is forwarded to the kamailio >>>> server, and plays an ivr to this leg as well when the call is >>>> answered, then it connects the two calls. >>>> 4. Kamailio translates the "virtual" number to the pstn number of >>>> subscriber B >>>> >>>> >>>> I have attached a picture of the above scenario. >>>> >>>> The modules that are loaded are: >>>> >>>> loadmodule "db_mysql.so" >>>> loadmodule "mi_fifo.so" >>>> loadmodule "mi_datagram.so" >>>> loadmodule "sl.so" >>>> loadmodule "tm.so" >>>> loadmodule "rr.so" >>>> loadmodule "pv.so" >>>> loadmodule "maxfwd.so" >>>> loadmodule "usrloc.so" >>>> loadmodule "registrar.so" >>>> loadmodule "textops.so" >>>> loadmodule "uri_db.so" >>>> loadmodule "siputils.so" >>>> loadmodule "xlog.so" >>>> loadmodule "acc.so" >>>> loadmodule "dispatcher.so" >>>> loadmodule "pdt.so" >>>> loadmodule "dialplan.so" >>>> loadmodule "siptrace.so" >>>> loadmodule "dialog.so" >>>> loadmodule "sqlops.so" >>>> loadmodule "userblacklist.so" >>>> loadmodule "htable.so" >>>> loadmodule "uac.so" >>>> >>>> >>>> The config that does all the routing is : >>>> >>>> route[10] { >>>> >>>> xlog("alx ------- This is Route 10 -------"); >>>> >>>> if($rU =~ "^.*%+") >>>> { >>>> xlog("alx ------- The number contains %23 "); >>>> $rU = $(rU{re.subst,/^(.*)%23(.*)/\1\2/}); >>>> #$rU = $(rU{s.unescape.user}); #It changes the %23 >>>> to # !! >>>> xlog("alx ------- The perl $rU ------- "); >>>> } >>>> >>>> if($rU =~ "^.*#+") >>>> { >>>> xlog("alx ------- The number contains #"); >>>> $rU = $(rU{re.subst,/^(.*)#(.*)/\1\2/}); >>>> #$rU = $(rU{s.unescape.user}); #It changes the %23 >>>> to # !! >>>> xlog("alx ------- The perl $rU ------- "); >>>> } >>>> >>>> if(prefix2domain("2", "0")) { >>>> >>>> $var(dial_grp) = $(rd{s.select,0,.}{s.int}); # >>>> Dialplan group prefix for routing >>>> $var(num_pr) = $(rd{s.select,1,.}{s.int}); # The >>>> number of digits that prefix has >>>> $var(num_translation) = >>>> $(rd{s.select,2,.}{s.int}); # Called number translation >>>> $avp(s:port_translation) = >>>> $(rd{s.select,3,.}{s.int}); # Port number translation >>>> #$var(test_var) = $(rd{s.select,4,.}{s.int}); # >>>> Future property >>>> >>>> $avp(s:cust_prefix) = $(rU{s.substr,0,$var(num_pr)}); >>>> $rU = $(rU{s.substr,$var(num_pr),0}); >>>> >>>> xlog("alx ------- The new rU is $rU and properties $rd >>>> -------"); >>>> >>>> if($var(num_translation) == 1) >>>> { >>>> if($sht(a=>$rU)!=null){ >>>> >>>> $rU = $sht(a=>$rU); >>>> xlog("alx ------- Translation Done. DST >>>> num=$rU ----------"); >>>> >>>> } else { >>>> xlog("alx ------- Translation NOT Done >>>> ----------"); >>>> } >>>> >>>> >>>> #xlog("alx ------- We have DST number >>>> translation for user fU $avp(s:frm_user_name) ----------"); >>>> #if(dp_translate("31", "$rU/$rU")) >>>> #{ >>>> # xlog("alx ------- Translation Done. >>>> DST num=$rU ----------"); >>>> #} else { >>>> # xlog("alx ------- Translation NOT >>>> Done ----------"); >>>> #} >>>> } >>>> >>>> if(dp_translate("$var(dial_grp)", "$rU/$rU")) >>>> { >>>> xlog("alx ------- The $rU and with >>>> attributes :$avp(s:dest) -------\n"); >>>> >>>> $var(i) = 0; >>>> while($(avp(s:dest){s.select,$var(i),.})!="#") >>>> { >>>> $avp(s:dstgrp) = >>>> $(avp(s:dest){s.select,$var(i),.}{s.int}); >>>> $var(i) = $var(i) + 1; >>>> xlog("alx ------- The >>>> avp(s:dstgrp)=$avp(s:dstgrp) var(i)=$var(i) -------"); >>>> } >>>> >>>> # backup the username so we can use >>>> different prefixes >>>> $avp(s:user) = $rU; >>>> >>>> # select destination from first group >>>> >>>> if(ds_select_domain("$avp(s:dstgrp)", "4")) >>>> { >>>> >>>> if($(ru{uri.param,prefix})!=null) >>>> { >>>> $ru >>>> = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; >>>> >>>> } else { >>>> $ru >>>> = "sip:" + $avp(s:user) + "@" + $rd; >>>> } >>>> } >>>> >>>> $avp(s:dstgrp) = null; >>>> xlog("alx ------- The final >>>> RURI is $ru ------- "); >>>> if($avp(s:port_translation) >>>> == 1) >>>> { >>>> rewriteport("5061"); >>>> } >>>> t_on_failure("3"); >>>> t_relay(); >>>> exit; >>>> >>>> } >>>> >>>> >>>> >>>> >>>> } >>>> >>>> >>>> } >>>> >>>> Attached is the trace >>>> >>>> Regards. >>>> >>>> P. >>>> >>>> marius zbihlei wrote: >>>>> Panagiotis Skoulikaritis wrote: >>>>>> Hello Daniel >>>>>> >>>>>> the kamailio version is 1.5.3 >>>>>> >>>>>> Regards >>>>>> >>>>>> P. >>>>> Hello, >>>>> >>>>> Can you give us more details like the sip message that generates >>>>> the coredump (or if every sip message received generates the >>>>> core), if your config does something more out of the >>>>> ordinary(let's say exotic). Can we reproduce it ? >>>>> >>>>> It would also be helpful if you specify the list of modules you >>>>> have loaded. >>>>> >>>>> Cheers, >>>>> Marius >>>>>> >>>>>> Daniel-Constantin Mierla wrote: >>>>>>> Hello, >>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (OpenSER) - Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> -- >>> Daniel-Constantin Mierla >>> Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 >>> * http://www.asipto.com/index.php/sip-router-masterclass/ >>> > > -- > Daniel-Constantin Mierla > Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 > * http://www.asipto.com/index.php/sip-router-masterclass/ >

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Re: [Kamailio-Users] Is Timestamp supported in kamailio?
by Iñaki Baz Castillo 16 Mar '10

16 Mar '10

2010/3/16 mustafa rifaee <mustafa.rifaee(a)gmail.com>: > Hi Castillo, > I would like to asking you about the Time-stamp header in SIP messages, and > why this header does not supported in most open SIP servers, Is it > recommended or not? > Is there any open source support the time stamp header. > > Is this fact true ?? I think The problem is that both UAS and UAC must have > the same system time to generate a useful result. So therefore both or all > SIP devices (think about an intermediate proxy) must have NTP activated. > > Please Help me; I think I *already* helped you as you *already* asked me directly. Please, keep the threads in the maillist, nobody likes to give private free support. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Is Timestamp supported in kamailio?
by mustafa rifaee 16 Mar '10

16 Mar '10

Hi all; I would like to ask you about the timestamp in kamailio, Is it supported or not ? if yes how i can enable it ? Please Help me; Thanks. Best Regards Mustafa Rifaee

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[Kamailio-Users] [dialog module] dialog is not terminated when no ACK is received
by Iñaki Baz Castillo 16 Mar '10

16 Mar '10

Hi, using Kamailio 1.5.4. I use dlg_manage() for an INVITE. 200 Ok is replied by the callee but the UAC doesn't send the ACK (due to a crash). The dialog remains in Kamailio dialog memory/table in state 3 and would expire after default_timeout (which usually is 3600 seconds or more, unsuitable for this case). Yes, it could occur that the proxy is not doing loose_routing, but in that case it doesn't make sense to use dialog module, so shouldn't dialog module expire dialogs in state 3 after ~32 seconds? I'm using profiles_with_value to limit the number of calls per user, so this issue is a bit important, as a UAC not sending the ACK for a 200 means one less available channel for this user during dialog module default_timeout. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] maintenance this afternoon
by Daniel-Constantin Mierla 15 Mar '10

15 Mar '10

Hello, some small maintenance tasks will be performed afternoon today to the server hosting the mailing lists and kamailio.org site, if you notice some downtime, try again a bit later. Thanks, Daniel

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[Kamailio-Users] kamailio crashes
by Panagiotis Skoulikaritis 14 Mar '10

14 Mar '10

Dear fellow kaimailio users We have a kamailio server which crashes. below is the backtrace from the core files any idea why the kamailio is crashing Regards Panagiotis core.29568 Mar 10 09:27 #0 fm_status (qm=0x73a040) at mem/f_malloc.c:609 #1 0x0000000000423d5c in sig_usr (signo=15) at main.c:563 #2 <signal handler called> #3 0x00000037e3cd4711 in __recvfrom_nocancel () from /lib64/libc.so.6 #4 0x00000000004790cc in udp_rcv_loop () at udp_server.c:408 #5 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 core.19350 Mar 10 09:05 (gdb) backtrace #0 free_to (tb=0x775c00) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad2432de100) at parser/hf.c:187 #2 0x00002ad23fe3e525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad2432dcf58, rpl=0x772d28, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad23fe47b46 in t_reply_matching (p_msg=0x772d28, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad23fe47fa2 in t_check (p_msg=0x772d28, param_branch=0x7ffff9c016bc) at t_lookup.c:964 #5 0x00002ad23fe58ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x772d28) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bK45f7.70f91294.0;received=77.247.97.11\r\nV", len=920, rcv_info=0x7ffff9c017a0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7ffff9c019b8) at main.c:774 (gdb) core.29567 Mar 10 09:27 gdb) backtrace #0 free_to (tb=0x776460) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad1805fa100) at parser/hf.c:187 #2 0x00002ad17d15a525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad1805f8f58, rpl=0x771920, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad17d163b46 in t_reply_matching (p_msg=0x771920, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad17d163fa2 in t_check (p_msg=0x771920, param_branch=0x7fff8bf6a8cc) at t_lookup.c:964 #5 0x00002ad17d174ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771920) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKddc9.fa58f7e.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bKb973f6a69c9bea270e9db867dd7cc90f\r\nRecord-Route: <si"..., len=919, rcv_info=0x7fff8bf6a9b0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 (gdb) core.5326 Mar 10 09:02 (gdb) backtrace #0 free_to (tb=0x772c48) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2b2578e22560) at parser/hf.c:187 #2 0x00002b257597b525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2b2578e213b8, rpl=0x771b50, code=<value optimized out>) at sip_msg.h:54 #3 0x00002b2575984b46 in t_reply_matching (p_msg=0x771b50, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002b2575984fa2 in t_check (p_msg=0x771b50, param_branch=0x7fff7cdc63ac) at t_lookup.c:964 #5 0x00002b2575995ac2 in reply_received (p_msg=0x772c48) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771b50) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKec2f.dc3d97a7.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bK47703457194f5be415efc231f6b3e923\r\nRecord-Route: <s"..., len=918, rcv_info=0x7fff7cdc6490) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=5, argv=0x7fff7cdc66a8) at main.c:774 (gdb) quit

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[Kamailio-Users] Loadbalancer/outbound_proxy and ACK problem
by Pavel Miskov 13 Mar '10

13 Mar '10

Hello list, let me first show my scenario: UAC_A ---> LB ---> | | PROXY+REG #1 | or | PROXY+REG #2 UAC_B <--- LB <--- | LB cfg: modparam("path", "use_received", 1) loose_route(); #for REG+PROXY if(is_method("REGISTER")){ add_path_received(); ds_select_dst("1", "0"); fix_nated_register(); } else { record_route(); add_path(); ds_select_dst("1", "0"); fix_nated_register(); } PROXY+REG cfg: ------------- modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) INVITE if working fine until ACK. UAC_A sends ACK to LB "ACK sip:UAC_B@LB_IP:5678;" insted of "ACK sip:UAC_B@UAC_B_PUB_IP:". In "200 OK" which PROXY sends to LB Contact is rewritten with ip addres of LB "Contact: <sip:UAC_B@LB_IP:5678;" and this is why UAC_A sends wrong ACK. Obviously I miss something so if someone could point me where to look. Thanks Pavel U UAC_A_PUB_IP:31488 -> LB_IP:5678 INVITE sip:UAC_B@test.com SIP/2.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport..Max-Forwards: 70..Contact: <sip:UAC_A@UAC_A_PRIV_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID:ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U LB_IP:5678 -> UAC_A_PUB_IP:31488 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488;received=UACs_PUB_IP..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Server: Kamailio (1.5.1-tls (x86_64/linux))..Content-Length: 0.... # U LB_IP:5678 -> REG_PROXY_IP:5166 INVITE sip:UAC_B@test.com SIP/2.0..Record-Route: <sip:LB_IP:5678;lr=on>..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Max-Forwards: 69..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Path: <sip:LB_IP:5678;lr>.... # U REG_PROXY_IP:5166 -> LB_IP:5678 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Server: Kamailio (1.5.1-tls (x86_64/linux))..Content-Length: 0.... # U REG_PROXY_IP:5166 -> LB_IP:5678 INVITE sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d SIP/2.0..Record-Route: <sip:REG_PROXY_IP:5166;lr=on;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Via: SIP/2.0/UDP REG_PROXY_IP:5166;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Route: <sip:LB_IP:5678;lr;received="sip:UAC_B_PUB_IP:31468">..Max-Forwards: 68..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Path: <sip:LB_IP:5678;lr>.... # U LB_IP:5678 -> REG_PROXY_IP:5166 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP REG_PROXY_IP:5166;branch=z9hG4bKa1b1.c7ca19a7.0;rport=5166;received=REG_PROXY_IP..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Server: Kamailio (1.5.1-tls (x86_64/linux))..Content-Length: 0.... # U LB_IP:5678 -> UAC_B_PUB_IP:31468 INVITE sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d SIP/2.0..Record-Route: <sip:REG_PROXY_IP:5166;lr=on;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.56778065.0..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Max-Forwards: 67..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Path: <sip:LB_IP:5678;lr>.... # U UAC_B_PUB_IP:31468 -> LB_IP:5678 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.56778065.0..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport =5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@UAC_B_PRIV_IP:31 468;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41.. Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U LB_IP:5678 -> REG_PROXY_IP:5166 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip: LB_IP:5678;lr=on>..Contact:<sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U REG_PROXY_IP:5166 -> LB_IP:5678 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To:"UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U LB_IP:5678 -> UAC_A_PUB_IP:31488 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U UAC_B_PUB_IP:31468 -> LB_IP:5678 SIP/2.0 200 OK..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.56778065.0..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@UAC_B_PRIV_IP:31468;r instance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U LB_IP:5678 -> REG_PROXY_IP:5166 SIP/2.0 200 OK..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact:<sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U REG_PROXY_IP:5166 -> LB_IP:5678 SIP/2.0 200 OK..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U LB_IP:5678 -> UAC_A_PUB_IP:31488 SIP/2.0 200 OK..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com >;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U UAC_A_PUB_IP:31488 -> LB_IP:5678 ACK sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d SIP/2.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-0472a03b5d1aa106-1---d8754z-;rport..Max-Forwards: 70..Route: <sip:LB_IP:5678;lr>..Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Contact: <sip:UAC_A@UAC_A_PRIV_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 ACK.. # U LB_IP:5678 -> REG_PROXY_IP:5166 ACK sip:REG_PROXY_IP:5166;lr;nat=yes SIP/2.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.2..Via: SIP/2.0/UDP LB_IP:5678;rport=5678;received=LB_IP;branch=z9hG4bKa1b1.46778065.2..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-0472a03b5d1aa106-1---d8754z-;rport=31488..Max-Forwards: 68..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4 YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 ACK..

3 5

[Kamailio-Users] What is the best module for routing INVITE based on DID in R-URI?
by Geoffrey Mina 13 Mar '10

13 Mar '10

Hello, I am putting up a Kamailio server which will do nothing but route INVITE requests from my upstream carrier to individual offices on my side. The office locations will NOT be registered SIP UAs, but other Kamailio proxy servers. What I want to have is a database of DIDs associated with a forwarding IP:Port and/or SRV records. 5555551212 ==> 1.2.3.4:5060 5555551213 ==> 1.2.3.5:5060 etc. Any guidance on what the best approach to achieving my goal would be much appreciated. Thanks, Geoff

4 12

[Kamailio-Users] need help to use the Kamailio register to remote sip proxy
by Cucku Cucku 12 Mar '10

12 Mar '10

Hi all i found the Kamailio supports register to remote sip proxy: http://www.mail-archive.com/users@lists.kamailio.org/msg07451.html it mentions to use : SQL to create the mysql table is in utils/kamctl/mysql/uac-create.sql but i didnt find the sql script My sip proxy version : sercmd> core.version Server: kamailio (3.0.1 (i386/linux)) 679736 my util folder : [root@localhost kamailio-3.0.1]# ls utils/kamctl/mysql/ acc-create.sql domainpolicy-create.sql purple-create.sql alias_db-create.sql drouting-create.sql registrar-create.sql auth_db-create.sql group-create.sql rls-create.sql avpops-create.sql htable-create.sql siptrace-create.sql carrierroute-create.sql imc-create.sql speeddial-create.sql cpl-create.sql lcr-create.sql standard-create.sql dialog-create.sql msilo-create.sql uri_db-create.sql dialplan-create.sql pdt-create.sql userblacklist-create.sql dispatcher-create.sql permissions-create.sql usrloc-create.sql domain-create.sql presence-create.sql please help Thank you Muốn đổi mới cách xưng hô với ông xã/ bà xã của bạn? Xem có bao nhiêu cách xưng hô giữa vợ và chồng nhé! http://vn.answers.yahoo.com

3 3

[Kamailio-Users] Stateless failover
by Santiago Soares 12 Mar '10

12 Mar '10

We're using Kamailio as a stateless load balancer, and would like to add failover support. Is it possible to have failover support in stateless mode? What i want is that when a server doesn't answer to a request, Kamailio set its state to probing, and start to ping it until it answer, then setting its state to active again. And that only active servers are selected by ds_select_dst(). Is it possible? How can I do this? Thank you. Santiago

4 4

[Kamailio-Users] Advanced Call Scenario / One Way Audio
by Brandon Armstead 12 Mar '10

12 Mar '10

Hello, As always, thank you for all / any help and input you may provide in advance. Call Scenario: UA1 -> REGISTRAR-01 -> Kamailio-01 -> Asterisk (New Call-ID + Asterisk in Media Path) -> Kamailio-01 -> REGISTRAR-02 -> UA2 UA1 is behind NAT UA2 is behind NAT The purpose of this is when using a shared "USRLOC" database to simulate calls from "PSTN" to generate both legs of the call, i.e. incoming and outgoing, and also allow for easier / cleaner "traversal" This aids from scenario's happening where UA1 calls UA2 (while UA1 exists on P1 and UA2 exists on P2) this prevents P1 -> UA2, and forces P2 -> UA2 We determine that this is a call from P1 to P2 (internal call) and thus create this "bridge / interconnection" We are running into a problem it seems with one way audio, i.e. the CALLEE can hear the CALLER, however the CALLER CAN NOT hear the CALLEE. REGISTRAR-01 AND REGISTRAR-02 are both "proxying" RTP As well as the initial Asterisk in "the middle" SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Perhaps there are better ways to go about this, let me know if I am way off course, thank you! Sincerely, Brandon Armstead

3 2

[Kamailio-Users] IP Authentication
by Joao Gomes Pereira 12 Mar '10

12 Mar '10

Hello to all Is is possible to implement IP authentication in Kamailio? Does Kamailio has a manual to do that? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

5 8

basic sip-router installation tutorial
by Meftah Tayeb 12 Mar '10

12 Mar '10

hi all, please could anyone give me a simple step by step tutorial to setup sip-router in a debian OS with mysql and other things? from jit, no zip file... thanks

4 3

[Kamailio-Users] Handling 3XX messages
by Bruno Machado 11 Mar '10

11 Mar '10

Hi! I have a Kamailio 1.5.2 handling 3XX messages. The commands below are executed in a failure route. If the first contact is not reachable or even a negative answer is received, the algorithm will call a second failure route, executing the 't_next_contacts'. Following this logic, the server will try several endpoints. The first 200 OK received from one this endpoints will establish the call. If all the contacts are tried, the server will finish the session with sending an error to the source. get_redirects("15:15","Redirect"); if(!t_load_contacts()) { t_reply("500", "Server Internal Error - Cannot load contacts"); exit; }; if(!t_next_contacts()) t_reply("404", "Not found"); else t_relay(); Ok... I found just one problem. If the 3XX message brings 5 different contacts, (all them with different q value), Kamailio just try 4 of them. Contact: sip:20182#556250982222@aaa.aaa.aaa.aaa;q=0.015, sip:3441#556250982222@bbb.bbb.bbb.bbb;q=0.014, sip:3441#556250982222@ccc.ccc.ccc.ccc;q=0.013, sip:50014#556250982222@ddd.ddd.ddd.ddd;q=0.012, sip:777#556250982222@eee.eee.eee.eee;q=0.011 After the 300 message, the server sends a invite to the first contact. Seconds later, analysing the debug screen, I see: DBG:tm:t_next_contacts: next contact is <sip:3441#556250982222@bbb.bbb.bbb.bbb The INVITEs are sent to this IP. After some seconds the server calls the third contact: DBG:tm:t_next_contacts: next contact is <sip:3441#556250982222@ccc.ccc.ccc.ccc And so on until this, the last contact: DBG:tm:t_next_contacts: next contact is <sip:777#556250982222@eee.eee.eee.eee ERROR:core:add_avp: 0 ID or NULL NAME AVP ERROR:tm:t_next_contacts: setting of fr_inv_timer_avp failed The proxy sent INVITEs for all them, except the last one because this failure. The function 't_next_contact' returns an error value. Somebody has a hint about this? Regards, bruno machado ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

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[Kamailio-Users] Advanced Call Scenario / One Way Audio
by Brandon Armstead 11 Mar '10

11 Mar '10

**resending as I am not sure if this made it out the first time, I do not believe it did, if this is a duplicate -- my apologies.** Hello, As always, thank you for all / any help and input you may provide in advance. Call Scenario: UA1 -> REGISTRAR-01 -> Kamailio-01 -> Asterisk (New Call-ID + Asterisk in Media Path) -> Kamailio-01 -> REGISTRAR-02 -> UA2 UA1 is behind NAT UA2 is behind NAT The purpose of this is when using a shared "USRLOC" database to simulate calls from "PSTN" to generate both legs of the call, i.e. incoming and outgoing, and also allow for easier / cleaner "traversal" This aids from scenario's happening where UA1 calls UA2 (while UA1 exists on P1 and UA2 exists on P2) this prevents P1 -> UA2, and forces P2 -> UA2 We determine that this is a call from P1 to P2 (internal call) and thus create this "bridge / interconnection" We are running into a problem it seems with one way audio, i.e. the CALLEE can hear the CALLER, however the CALLER CAN NOT hear the CALLEE. REGISTRAR-01 AND REGISTRAR-02 are both "proxying" RTP As well as the initial Asterisk in "the middle" SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Perhaps there are better ways to go about this, let me know if I am way off course, thank you! Sincerely, Brandon Armstead

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[Kamailio-Users] 400 Bad Request
by Ernest Mavrel 10 Mar '10

10 Mar '10

3 2

[Kamailio-Users] cdr_id not registered in acc
by Anders 10 Mar '10

10 Mar '10

Hi, (Kamailio 1.5 + freeradius + cdrtool) After a call is finished, it's INVITE and BYE are registered just fine in the acc table, but the records in acc never receive a cdr_id, and (therefore) nothing is inserted the cdrs table. Ideas on where to start looking? Thanks, Anders

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[Kamailio-Users] DEFINE question
by Uriel Rozenbaum 10 Mar '10

10 Mar '10

Hi guys, The question's simple, is there any pre-processor command to DEFINE constants? Thanks! Uriel

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[Kamailio-Users] OpenSER+Asterisk+BLF
by Mark Sayer 10 Mar '10

10 Mar '10

We are needing to modify the configure of a currently operating OpenSER to properly relay the SUBSCRIBE and SIP-NOTIFY messages that are sent between Asterisk and a phone that supports BLF (like the Snom 300 or Yealink T26). Our setup includes an OpenSER 1.2 & Asterisk 1.4.17 in the same box. OpenSER performs all registration, authentication and NAT. Asterisk handles the media and the accounting. In a pure Asterisk environment a "hint" would be setup in the Asterisk extensions.conf file and the phone (UA) would SUBSCRIBE to that HINT. Once Asterisk has registered that UA to the HINT then it sends SIP-NOTIFY messages to the UA as the status of the channel changes (available, ringing, busy). Our current openser.cfg file makes no mention of either SUBSCRIBE or NOTIFY which is an obvious reason that my Asterisk installation never registers the UA to the HINT. Is anyone interested in getting paid to fix this for us (we're too stupid to do it ourselves) or to offer another solution for controlling BLF in this setup. Thanks, Mark

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[Kamailio-Users] small typo in rewriteport example on Kamailio Wiki
by Andreas Heise 09 Mar '10

09 Mar '10

Hello, there is a small typo in rewriteport example on Kamailio Wiki should be rewriteport("5070"); instead of rewriteport(5070); http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#rewriteport_s… regards, Andreas

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[Kamailio-Users] Kamailio (OpenSER) v3.0.1 Released
by Daniel-Constantin Mierla 08 Mar '10

08 Mar '10

Hello, the first patch release for 3.0 series is out as version 3.0.1. It includes the fixes to issues discovered since 3.0.0. Database structure and configuration file compatibility are preserved so the upgrade from 3.0.0 is straightforward. Links and more details are available at: http://www.kamailio.org/w/2010/03/kamailio-v3-0-1-released/ Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] [OT] BYE sip:SIMPLE&XCAP@ietf.org SIP/1.0
by Iñaki Baz Castillo 08 Mar '10

08 Mar '10

Hi, if somebody is bored and has 5-10 minutes of spare time I suggest to read the mail I've sent to sip-implementors maillist, full of hate and fury: https://lists.cs.columbia.edu/pipermail/sip-implementors/2010- March/024529.html Enjoy. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Serusers] ser numbers going up
by Sabatella, Michael 08 Mar '10

08 Mar '10

Hi all, Running ser 0.9.6. I notice these numbers going up and never going back down: I am a bit concerened with the red text. Is there any reason why that does not come down to 0. Do I need to restart the ser process once a month or something around that timing? So far I have not heard of any issues with user call setup but I just want to remain proactive. Serctl moni outputs the following: Server: Sip EXpress router (0.9.6 (i386/linux)) Now: Mon Mar 8 10:05:18 2010 Up Since: Fri Jan 8 11:24:17 2010 Up time: 5092861 [sec] Transaction Statistics Current: 1 (176 waiting) Total: 5624219 (0 local) Replied localy: 6049354 Completion status 6xx: 0, 5xx: 121, 4xx: 308504, 3xx: 0,2xx: 5316846 Stateless Server Statistics 200: 13145729 202: 0 2xx: 0 300: 0 301: 0 302: 0 3xx: 0 400: 42 401: 0 403: 0 404: 350766 407: 16536998 408: 0 483: 0 4xx: 2 500: 1 5xx: 0 6xx: 0 xxx: 0 failures: 72 Michael Sabatella IPC Systems Network Services Engineer Office: 1-201-253-2280 Cell: 1-908-380-5321 @ IPCGlobalNetworks.com<http://www.ipcglobalnetworks.com/> ________________________________ 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.

1 0

[Kamailio-Users] Get all the values for dialogs with a specific profile
by Iñaki Baz Castillo 08 Mar '10

08 Mar '10

Hi, by running "kamctl fifo profile_get_size togw" I can see the number of dialog tagged as "togw". Also if I run "kamctl fifo profile_get_size togw 999888777" I can see the number of dialogs tagged as "togw" with value "999888777". Is there any way to display all current values for dialogs with a specific profile/tag (i.e. "togw")? The idea is building a simple script to display the current number of calls per user (having that the value applied to dialog profiles means the calling user). Thanks for any suggestion. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] new website look
by Daniel-Constantin Mierla 08 Mar '10

08 Mar '10

Hello, http://www.kamailio.org has a new look! The former site was built with a cms system that discontinued the development, providing no longer updates for newer technologies and nicer design. We hope the new design helps to locate project resources easier and makes site navigation more straightforward via comprehensive list of links in the right sidebar. Credits go to Elio Rojano for the logo artwork and to developers of GPL Wordpress theme Cordobo Green Park which was used as start for the new design. If you are skilled in web design and can build a more personalized look on top of wordpress, please contact us. Cheers, Daniel PS. Do not forget, if you are tomorrow in London, UK, consider attending Present and Future of SIP Routing to get latest update about the project: http://www.kamailio.org/w/present-and-future-of-sip-routing-2010-london/ -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

1 0

[Kamailio-Users] Dispatcher ds_list results in flag=P
by Geoffrey Mina 08 Mar '10

08 Mar '10

Hello, I have a weird issue with Dispatcher module. When I do a ds_list I get some of my destinations returned with flag=P. It appears that when they are in this state, they are not included in the list of destinations to send traffic to. I have tried restarting Kamailio, and these certain servers always return to flag=P. I have confirmed the servers are up and running fine. They are communicating without issue at the SIP and network level. Any idea what could cause this? Thanks,

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Re: [Kamailio-Users] Users Digest, Vol 57, Issue 85
by Zhe 07 Mar '10

07 Mar '10

Help. Please remove me from this mail list. Thanks<br><br>在2010-02-27 19:00:01，users-request@lists.kamailio.org 写道： >Send Users mailing list submissions to > users(a)lists.kamailio.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > >You can reach the person managing the list at > users-owner(a)lists.kamailio.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > > 1. remove parameter (Elgar Onel) > 2. Exotic case (but real) in which RtpProxy doesn't work > (I?aki Baz Castillo) > 3. Re: Exotic case (but real) in which RtpProxy doesn't work > (Daniel-Constantin Mierla) > 4. Re: Exotic case (but real) in which RtpProxy doesn't work > (I?aki Baz Castillo) > 5. Re: Exotic case (but real) in which RtpProxy doesn't work > (Daniel-Constantin Mierla) > 6. Re: Exotic case (but real) in which RtpProxy doesn't work > (I?aki Baz Castillo) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Fri, 26 Feb 2010 03:00:25 -0800 (PST) >From: Elgar Onel <elgaronel(a)yahoo.com> >Subject: [Kamailio-Users] remove parameter >To: users(a)lists.kamailio.org >Message-ID: <444271.31134.qm(a)web114207.mail.gq1.yahoo.com> >Content-Type: text/plain; charset="us-ascii" > >Dear friends, > >how to remove parameters from INVITE sip address? > >tia, e. > > > > >

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[Kamailio-Users] 3.0 Modules
by dotnetdub 07 Mar '10

07 Mar '10

Hi List, I see two modules in particular that I could really do with running on 1.5.3 - I have started testing 3.0 but will not put into production just yet. Call control and topoh Would it be possible to backport these to 1.53 ? Would there be much to it.?. Call Control in particular is something that is very useful for us. We will eventually move to 3.0 of course as it is amazing product with lots of really nice new features just been burned badly before by moving to something so new. TIA. Stephen.

2 1

[Kamailio-Users] 10 years sip express router (ser)
by Daniel-Constantin Mierla 05 Mar '10

05 Mar '10

Hello, yesterday at CeBIT I've run into Fraunhofer booth and a banner attracted my attention, photo at: http://www.asipto.com/gallery/v/cebit2010/P3030721.JPG.html?g2_imageViewsIn… According to the middle right bubble, 2000 was the first year FhG presented SER at CeBIT. Given that, means the project should celebrate (at least) 10 years these days?!? I was among FhG staffers at CeBIT 2002, presenting latest SER at that time. According to CVS repository on BerliOS, files such as Makefile and main.c are dated 2001. Officially, SER was GPL-ed and open sourced in 2002 (year when ser project was registered to berlios.de software forge). Andrei, more insights from non-open-source era? I got into the project in January 2002, would be interesting to trace the birthday... Cheers, Daniel

2 1

[Kamailio-Users] local 408 from fr_timer and fr_inv_timer
by Robert McGilvray 04 Mar '10

04 Mar '10

I'm looking for a way to distinquish between locally generated 408s triggered by the timers. My problem is that I'm using gateway failover in a failure route and on 408s it would retry phones multiple times if they didn't answer (no voicemail etc) - so fr_inv_timer was triggered after receiving a provisional reply. I need to failover in case a gateway doesn't respond with provisional (it's set at 5 seconds currently). Will this work, is there a cleaner solution? I haven't had the oppurtunity to test it so I may have some syntax errors. onreply_route { if (t_check_status("1[0-9][0-9]")) { xlog("L_NOTICE", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu TO $ru -> Provisional Response --> $T_reply_code\n"); setflag(FLAG_PR_RECEIVED); } } failure_route[ROUTE_DP_FAILURE] { if (t_local_replied("last") && t_check_status("408") && isflagset(FLAG_PR_RECEIVED)) { xlog("L_INFO", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu TO $ru -> Local 408 after provisional response, giving up\n"); exit; } } Thanks! Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] Present and Future of SIP Routing 2010 London
by Daniel-Constantin Mierla 04 Mar '10

04 Mar '10

Hello, I am pleased to announce a dedicated event in London, Tuesday, March 9 afternoon, where to present the version 3.0 and latest developments in Kamailio and SIP Router projects. The event includes as well presentations from various players in the VoIP and SIP market, more details at: http://www.kamailio.org/mos/view/Present-and-Future-of-SIP-Routing-2010-Lon… It is a free event, the number of seats is limited, room being available courtesy of Sun Microsystems (address 55 King William St, London EC4), therefore the first come first serve policy is applied. You can also enroll for a presentation, just send a short description about the topic in the registration email. Hope to see many of you in London next week, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Serusers] Is this a SER Problem?
by Samson Luk 03 Mar '10

03 Mar '10

I use SPA3102 with a iptel.org VoIP a/c, whenever my Firewall external IP changed (SPA3102's WAN port only connected to the internal LAN with a static IP, router function not used) the iptel.org VoIP a/c will report registration failed. I have to remove the proxy domain field and re-insert it then reboot to fix the problem, just wonder if this is a problem on the SER side or on the SPA side. I am sorry if this is not a correct appropriate place to ask such a question.

1 0

Re: [Kamailio-Users] Kamailio and SIP <-> XMPP gateway
by Daniel-Constantin Mierla 03 Mar '10

03 Mar '10

On 03/02/2010 11:25 AM, Even André Fiskvik wrote: > Thanks for the answers Daniel. > > On 1. mars 2010, at 12.23, Daniel-Constantin Mierla wrote: > > >> Hello, >> >> On 02/28/2010 06:43 PM, Even André Fiskvik wrote: >> >>> Reading through various earlier discussion though, I've found an issue that I wonder wether has been solved or not. >>> In some mailing-lists I saw that only text/plain MESSAGEs was supported through the xmpp gateway, is this still true? >>> >>> >> yes and 'no' :-). Yes because nothing was done in this respect to the module itself. 'No' because now (kamailio 3.0) you can use transformations to get content of xml documents and if the body is xhtml. >> > That's good to know. > > >>> If that's the case, is there any way to get CounterPath eyeBeam/Bria to work in the decribed scenario? >>> As to my knowledge it can only send text/html ? >>> >>> >> You can build a new message with uac module or update the existing SIP message (content type, body), do msg_apply_changes(): >> http://kamailio.org/docs/modules/3.0.x/modules_k/textops.html#id2513318 >> >> and they use xmpp functions. >> > I did a "quick" hack to support relying the MESSAGE by just setting the content-type to text/plain for now to do some testing. > The error about unsupported content-type thus went away. > > >> You can use ngrep to sniff the network to see connection between kamailio (openser) and xmpp server as well as run kamailio in debug mode in order to get more verbose log messages for each operation. >> > I've tried sniffing traffic on the server, and it doesn't seem like kamailio tries to connect to the XMPP server at all :/ > Any clues what might be causing this? Did you run in debug mode? Can you post the log messages? > I've attached my kamailio config-file as well if anyone would take You sent the email only to me. Did you intend it for mailing list? I recommend to keep sending to mailing list apart of messages with private data. In this way many can reply and other can learn from discussion. > a look at it (the main config is in xmpp.cfg) and also a diagram showing how the components are realated to each other. > I will try to look over configs when I get some time -- because of a trip ahead I am pretty constrained today. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] preparing for kamailio 3.0.1
by Daniel-Constantin Mierla 03 Mar '10

03 Mar '10

Hello, I think it is time to prepare for first minor release (3.0.1) in 3.0 series. We are approaching two months since kamailio 3.0.0, period within several issues were reported and fixed. Please report anything you are aware is not currently working in 3.0 to sr-dev(a)lists.sip-router.org or bug tracker: http://sip-router.org/tracker/ As for date, end of this week or beginning of next week would be a good time? Other opinions? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Serusers] Need a way to substitute SIP Response Code 504 for something else
by Frank Durda IV 03 Mar '10

03 Mar '10

There is a commercial switch vendor who has recently decided to make two changes that combined are quite evil: 1, They decided to devivate from RFC 3398 and when a given call INVITE timer expire (any of the SS7 ones, plus the SIP ones, including any limit specified in the INVITE itself), instead of translating the ISUP Release Cause 102 into a SIP Response Code of 504 (Gateway Timeout), they send out a 503 (Service Unavailable) instead. (503 is commonly used in low-cost carriers and the commercial equipment they use to trigger a route-advance, where the call doesn't end but is relaunched via another slightly more expensive route/carrier who perhaps can get the call to the destination. Some of these low-cost outfits may retry the same call with five or six carriers before reluctantly sending to the local access tandem which is usually the most expensive choice, but almost certain to work. This probably isn't what the inventors of the 503 code had in mind, but that is how it is used. A minority use 404 the same way.) 2. The same vendor implemented an automatic "Address Reach" testing mechanism for outbound calls. This thing means that when sending a call to a distant switch, if a single or small number of calls are returned from that distant switch with a SIP Response Code of 504 from the destination, the sending switch will take that destination switch completely out of route ("blacklist") and not send any calls at all to that destination until the blacklist is manually reset. So a result that applies to one call is fatal to all subsequent calls. This also means that if you sent the INVITE Expires timer to a value less than the number of seconds the calling party is willing to let the called number ring (called party doesn't have voice mail or it doesn't pick up for several rings), you trigger a 504 and bingo, you have killed the route between those two switches. So your own choices for how long to wait on ring-back for a call dictate how quickly it could kill the entire route for that call. Now, the two changes taken together are also somewhat convienent because this equipment maker just happened to make it impossible for their own equipment to ever return a 504 and do can never trip off their own booby-trap. So in a way what these two changes effectively create a not-our-hardware-detector, triggering route failures only when in contact with RFC-compliant equipment made by somebody else. I think that behavior is anti-competitive, and probably illegal in quite a few countries (notably the EU), but that is what exists at the moment. This creation started getting deployed in the past few months. So, to get around this nonsense, I need a way in SER to detect that a 504 has come in and change it to something else before sending it on. I won't quibble about the "proxies can't forward a 503", so I won't try that. I would like to turn the 504 into a 404, which is better than having route get pulled all over the place. Now, the last time (a year or so ago) I asked about trying to change a SIP Response Code in SER, the suggested ser.cfg "fix" created all sorts of strange warnings in the logs about timers expiring and I was told to not worry about that. Well, it seemed to cause a lot more problems than just warnings like memory leaks or cores caused by something, so I ended up having to hard-code the translation I wanted into the SER source code and not use rules in ser.cfg to do this. I'm writing now to (A) alert people about what this major brand switch maker is doing with regard to 504s so you will recognize it if you encounter it, but also (B) to see if maybe has a more elegant way to change SIP Responses via the ser.cfg file has emerged. The obvious things like subst subst_uri don't appear to work, obviously busily editing the non-existent INVITE method message and ignoring the reply message. Using a reply/drop combination to generate a new reply and kill the real one doesn't seem to work either. As I recall it ends up sending the response specified in the reply, and then passing the original response as well. "drop" apparently doesn't kill it completely and I suspect reply is really meant to be used only during handling of the method messages, and not the response messages. Suggestions appreciated.

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[Kamailio-Users] kam 1.5/asterisk 1.6 hints/blf?
by Robert McGilvray 02 Mar '10

02 Mar '10

Please excuse the noob questions, I'm not very familiar with presence. Users are registered to kamailio, authenticated against radius with LDAP backend. I'd like to setup some queues on asterisk for helpdesk call center but in order to do it correctly asterisk needs the device state (ringing, in use etc). I assume I'll need to use realtime ldap and setup device entries for all my phones, then use hints on the extensions as if they were local. If I understand correctly this will make asterisk subscribe to kamailio (I will use kams IP in host field), kam then has the responsibility of handling the subscribe/publish/notify stuff. What needs to be done on the kam side? Thanks! Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] Kamailio (1.5 rev5834) and/or rtpproxy gets frozen
by Iñaki Baz Castillo 01 Mar '10

01 Mar '10

Hi, I'm experimenting an annoying issue in a Debian Lenny 32 bits under a DELL 850 (2 cores). Same occured before with Debian Etch 32 and 64 bits in other host (also DELL 850). Kamailio (1.5 rev 5834) behaves as a load balancer in front of 2 PSTN gateways. There are ~200 calls and for all of them rtpproxy is applied. Sometimes, with no reason (not just in the moments of highest traffic) kamailio gets totally frozen, this is, it doesn't reply to SIP messages, neither relays them. Then Kamailio cannot be killed (just with -9). Sometimes it occurs when performing a fifo command (i.e. "kamctl fifo address_reload"). After killing it, kamailio cannot be started again. It starts and logs the usual logs (to syslog) but after "rtp proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled" there is no more logs. Then if I do a "ps aux | grep kamailio" I just see a single process, it's like the children fail when connecting to rtpproxy and die but the master process doesn't realize of it and remains alive showing no error at all. So I suspect from rtpproxy as even when kamailio was killed rtpproxy process consumes lot of CPU. I'll try with rtpproxy-1.2.1 (the latest stable version) as before I was using a old "truk" version (unfortunatelly I've no idea about how to know the CVS revision of a CVS working directory...). Anyhow I also suspect that the problem could be in the server itself, as sometimes the above problem occurs when reloading iptables and so on, very very strange, no a definitive cause or reason, very annoying. So nothing is clear for me. But I know that the problem occurs: - In two DELL 850, one with 32 and the other with 64 bits. - In both Debian Leeny and Etch. - With kamailio 1.5 rev 5834. - With rtpproxy rev XXX (how to get the last modification date in a CVS working dirrectory?) So, the fact is that I'm completely lost, no idea of what is happening. Any suggestion? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Project Update, London Meeting, March 9
by Daniel-Constantin Mierla 01 Mar '10

01 Mar '10

Hello, Tuesday, March 9, I am going to be in London and plan to organize a social networking event for folks around SIP/VoIP, Kamailio (OpenSER) & SIP Router. One is going to be the usual evening dinner/pub meeting, see pics from some past events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ I addition, if there are people interested, I would like to have a session in the afternoon (2-3 hours) to give an update about Kamailio (OpenSER) 3.0, present and future development of SIP Router. Time is rather short, but I encourage attendees to prepare short presentations as well, about their products and services, to have more information exchanged there. Let me know if you wish to attend such event in order to find a proper place to host it (drop me an email). Participation is free for everybody, at the dinner, the classic rule, everyone is going to pay for himself/herself. Looking forward to meeting many of you in London. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] Kamailio and SIP <-> XMPP gateway
by Even André Fiskvik 01 Mar '10

01 Mar '10

Hi. I've just started looking at kamailio, and I'm experimenting a bit with configuring a SIP <-> XMPP gateway with it using the pua_xmpp module. The configuration part hasn't beed straight-forward, but I think I'm starting to get somewhere. Both kamailio and XMPP/Jabber are new technologies for me serverside, so I don't have a great deal of experience with either. In short, what I'm trying to accomplish at this stage is: - Have CounterPath eyeBeam/Bria to register to my kamailio server - Have Aastra 6755i register to my kamailio server - Connect an XMPP client to my XMPP server - The CounterPath eyeBeam/Bria, Aastra and the XMPP client should all be able to see each others presence - The CounterPath eyeBeam/Bria should be able to IM other eyeBeams/Brias - The CounterPath eyeBeam/bria should be able to IM the XMPP clients and the other way around Reading through various earlier discussion though, I've found an issue that I wonder wether has been solved or not. In some mailing-lists I saw that only text/plain MESSAGEs was supported through the xmpp gateway, is this still true? If that's the case, is there any way to get CounterPath eyeBeam/Bria to work in the decribed scenario? As to my knowledge it can only send text/html ? Could anyone with a good knowledge of Kamailio and the various modules pua_xmpp, xmpp, pua, pua_bla comment on this kind of setup is at all possible with kamailio and it's current state? Is there any way I can verify that opensips connects to the XMPP server correctly? I tried connecting FreeSWITCH to my XMPP server (Openfire) in component mode, and it would show up connected in Openfire's administration GUI, however I can not see the same when I try to configure kamailio. If anyone has experience with this setup and have another XMPP server they have tested it with, I would also want to try to swap Openfire with that (given they also have some guidance on XMPP server configuration). Best regards, Even André Fiskvik

2 1

[Kamailio-Users] remove parameter
by Elgar Onel 26 Feb '10

26 Feb '10

Dear friends, how to remove parameters from INVITE sip address? tia, e.

1 0

[Kamailio-Users] ratelimit enhancements
by Daniel-Constantin Mierla 26 Feb '10

26 Feb '10

Hi Marius, since you did some updates to this module, I am opening for debate some needed enhancements I did during 3.0 testing phase and want to get opinions how to get in the code repo. Practically is a new module I named for now ratelimit2 and my last idea is to get it named pipelimit in the trunk. The reason for a new module are some major changes. The module uses the same algorithm but its core is overhaul. - definitions of pipes are loaded from database - there can be unlimited number of pipes - pipes are identified by string names - should be possible to reload pipes at runtime (iirc, not yet in) - new pipes can be added at runtime - functions accept variables to identify the pipe Since I never used queues from this module and haven't spent time to understand the concept behind, this functionality is completely missing. The old module might be good to keep in place, probably many people are using it in this form. So, proposals? What is the way to go on? Common code (algorithms) can be made lib at some point. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] tracking new features
by Daniel-Constantin Mierla 25 Feb '10

25 Feb '10

Hello, a reminder for developers to write a short note for each new feature either to NEWS file in source tree or to wiki page: http://sip-router.org/wiki/features/new-in-devel Makes life much easier when the next major release is going to happen and help people to spot quickly what new features are already available in devel branch, hopefully attracting some of them to go for it and test. Now the wiki page is updated with most of NEWS content for 3.1. Thanks, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

1 0

[Kamailio-Users] CeBIT 2010
by Daniel-Constantin Mierla 25 Feb '10

25 Feb '10

Hello, I am going to visit CeBIT 2010 in Hannover next week. Drop me an email if you are around and want to meet and chat about Kamailio, SIP Router projects, etc. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

1 0

[Kamailio-Users] Storing nonscalar types (lists)
by Alex Balashov 24 Feb '10

24 Feb '10

Are AVP arrays still the only way to create a list of scalars as of K 3.0.0? Or are there other, more graceful ways to accommodate this need now? AVPs are a very useful feature, though the syntax is not documented in 'avpops' or the cookbook except rather incidentally: $var(i) = 0; while(is_avp_set("$(avp(s:somename)[$var(i)])")) { xlog("L_INFO", "Value at index $var(i): $(avp(s:somename)][$var(i)])\n"); $var(i) = $var(i) + 1; } Is there a better way to handle this now, perhaps via some features imported from SER? Script variables ($var(...)) do not support subscripts judging by the cookbook; do they? Are there other options I am unaware of? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

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[Kamailio-Users] Logging of db_extra parameters + serial forking
by Alex Balashov 24 Feb '10

24 Feb '10

Here is a problem I am sure is not new (1.5.x): modparam("acc", "db_extra", "gateway_id=$avp(s:gateway_id)") route { ... $avp(s:gateway_id) = 13; $ru = "some.gw1"; t_on_failure("1"); if(!t_relay()) sl_reply_error(); } # Serial forking rollover route failure_route[1] { if(t_was_cancelled()) exit; $avp(s:gateway_id) = 14; $rd = "some.gw2"; append_branch(); t_relay(); } The problem is, when serial forking happens, the CDR rows in 'acc' still show a value of 13 for 'gateway_id'. This is obviously because the value is sampled at the time of the initial stateful relay processing and is not updated afterward. Any easy/elegant fix? I can think of several fixes that don't involve using 'acc', obviously, but I'd rather avoid that... -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

1 1

Re: [SR-Users] function calling in the config script with $var() as attribute(s)
by Klaus Feichtinger 24 Feb '10

24 Feb '10

Klaus, Alex, you're right. I've seen Klaus's suggestion only after writing my email message. This is the best solution and solves my problem! regards, Klaus > > > Am 24.02.2010 15:41, schrieb Klaus Feichtinger: > > Hello SR-users and -developers, > > > > I have the question if parameters for a function in the config script > > (e.g. "t_relay(host, port)") in general can be set with a variable > > ($var(xy)). Because of having a set of SIP servers / gateways > distributed > > over a wide area and therefore keeping the configuration as simple as > > possible, I would like to define a custom variable, assign it to a $var > > and use this variable as attribute for a relay function. In this case > the > > variable would represent a string of an IP-address (e.g. "10.1.1.55"). > > When this would work I only have to adapt the custom variables on top of > > each config script and could let the rest "as it is". > > > > > > First practical tests have shown, that the functions are NOT happy with > these "attributes". It can be used without any problems for if-else- > > statements, but I fear not for functions.... > > > > The practical scenario looks like: > > > > tc.ip11 = "10.16.65.101" desc ""; > > [...] > > route { > > $var(test) = $sel(cfg_get.tc.ip11); > > [...] > > if (!t_relay($var(test), "5061")) > > A workaround for t_relay: > > $du = "sip:"+$var(test)+":5061"; > t_relay(); > > > klaus > > > > [...] > > } > > > > This configuration resulted in following error messages: > > > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: syntax error > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 38-41: Function parameter with integer > value not allowed#012 > > > > > > Please, could anybody clarify if this should work or not? If not - is > any > > alternative possibility available for "automating" function attributes > > instead of re-writing each function using the same string attribute? > > > > Thanks in advance, > > > > Klaus > > -- GMX DSL: Internet, Telefon und Entertainment für nur 19,99 EUR/mtl.! http://portal.gmx.net/de/go/dsl02

1 0

function calling in the config script with $var() as attribute(s)
by Klaus Feichtinger 24 Feb '10

24 Feb '10

Hello SR-users and -developers, I have the question if parameters for a function in the config script (e.g. "t_relay(host, port)") in general can be set with a variable ($var(xy)). Because of having a set of SIP servers / gateways distributed over a wide area and therefore keeping the configuration as simple as possible, I would like to define a custom variable, assign it to a $var and use this variable as attribute for a relay function. In this case the variable would represent a string of an IP-address (e.g. "10.1.1.55"). When this would work I only have to adapt the custom variables on top of each config script and could let the rest "as it is". First practical tests have shown, that the functions are NOT happy with these "attributes". It can be used without any problems for if-else- statements, but I fear not for functions.... The practical scenario looks like: tc.ip11 = "10.16.65.101" desc ""; [...] route { $var(test) = $sel(cfg_get.tc.ip11); [...] if (!t_relay($var(test), "5061")) [...] } This configuration resulted in following error messages: <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: syntax error <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 38-41: Function parameter with integer value not allowed#012 Please, could anybody clarify if this should work or not? If not - is any alternative possibility available for "automating" function attributes instead of re-writing each function using the same string attribute? Thanks in advance, Klaus -- Sicherer, schneller und einfacher. Die aktuellen Internet-Browser - jetzt kostenlos herunterladen! http://portal.gmx.net/de/go/chbrowser

3 4

[Kamailio-Users] Kamailio 3.0.0: configuration request regarding VARIABLES usage (custom variables + $var(x))
by Klaus Feichtinger 24 Feb '10

24 Feb '10

Hello Kamailio-3 users, I have two questions regarding usage of variables within the config file (as published in the NEW FEATURES of Kamailio 3.0.0). The reason why I want to use variables instead of standard configuration lines is the maintainability of my SIP servers. I have a group of about 20 sip servers distributed over a wide area. Especially routing rules might change during operation of the servers and must be adapted on all servers. Therefore I want to use variables, defined at the beginning of each config file. Within the config file / the different routing blocks I want to reference to these variables. That would be much easier for automatic config script adaptation with a linux script (e.g.) than parsing the whole config file for special text phrases. QUESTION 1: ========== Is it impossible using a variable for requesting a SRC_IP address? With my current configuration I always see an entry in the SYSLOG that looks as follows: "<core> [route.c:1531]: BUG: comp_ip: invalid type for src_ip or dst_ip (20)" The configuration lines look like: $var(lst_gwIpRange) = "10.16.48.6[0-2]"; [...] if (src_ip=~$var(lst_gwIpRange)) { [...] } Is maybe a transformation missing/necessary to avoid the error message (which causes a malfunction)? Can anybody give me a hint? QUESTION 2: ========== The second problem I found is, that the config variables can be configured within a route block only and not - as expected - e.g. at the top of the config file (in the same way as the custom variables). I can't use the custom variables as they are, because I will use more complex variables, which are a mixture of up to three custom variables. These "complex" variables will be used in more than one routing block. Therefore I have to define these variables in every routing block...... Is an alternative (easier) solution with a single definition available, too? An example of my problem is as follows: # custom variables defined on top of the kamailio.cfg file tst.loc_ipaddr = "10.10.1.1" desc "IP addr of the local SIP Server" tst.loc_prefix = "11" desc "prefix of this SRV" tst.distRu1 = "51[0-9].*" desc "variable representing ID1" # the final variable should look like "sip:1151[0-9].*@10.10.1.1" # and is used in 3 routing blocks route[] $var(test) = "sip:" + $sel(cfg_get.tst.loc_prefix) + $sel(cfg_get.tst.distRu1) + "@" + $sel(cfg_get.tst.loc_ipaddr); [...] route[TEST1] $var(test) = "sip:" + $sel(cfg_get.tst.loc_prefix) + $sel(cfg_get.tst.distRu1) + "@" + $sel(cfg_get.tst.loc_ipaddr); [...] route[TEST2] $var(test) = "sip:" + $sel(cfg_get.tst.loc_prefix) + $sel(cfg_get.tst.distRu1) + "@" + $sel(cfg_get.tst.loc_ipaddr); [...] Thanks in advance for any hints! Regards, Klaus Feichtinger -- Sicherer, schneller und einfacher. Die aktuellen Internet-Browser - jetzt kostenlos herunterladen! http://portal.gmx.net/de/go/atbrowser

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[Kamailio-Users] 3.0.0 and snmpstats
by Kelvin Chua 24 Feb '10

24 Feb '10

hi guys, anybody here tried using snmpstats with 3.0.0? i keep on getting "No such Object" already verified that kamailio is talking to agentx master. net-snmp version 5.3.3 compiled from source also disabled openssl and md5 in net-snmp to prevent any undefined symbols been reading the mailing lists and so far nothing helped. seems like only a couple of guys are using this module? Kelvin Chua

2 4

[Kamailio-Users] Freeing sqlops results
by Alex Balashov 23 Feb '10

23 Feb '10

Should sql_result_free() always be called, no matter what the circumstance, in order to discharge memory allocated to query result sets and/or metadata? Or is it not necessary in certain situations, for example when script is broken via 'exit', i.e. sql_query("db", "SELECT ...", "res"); if(! $dbr(res=>rows)) { # Is it necessary to call sql_result_free() here? sl_send_reply("404", "Not Found"); exit; } Are results implicitly freed at the end of route script execution anyway? Are they tied to transaction persistence in any way? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

3 3

[Kamailio-Users] Forward calls from Asterisk to SIP provider via Kamailio for termination
by Slot Zero 23 Feb '10

23 Feb '10

Hello, I am a Kamailio noob :). I am trying to get Asterisk to forward calls to my SIP provider via Kamailio. The same machine is running Kamailio and Asterisk. I do not want to consume credentials as they have to be passed on all the way to my SIP provider. There is no NAT of any sorts. SIP Phone/Users connect to Asterisk.I do not need to authenticate when forwarding call from Asterisk to Kamailio as they are both running on the same server but I do need to make sure that Kamailio dials and forwards 011+number to be sent from local host port 5062(Asterisk listener) to SIP provider only. I have 6 Public IP addresses mapped on the server. I want to use the force_send_socket to allow me to change source IP of SIP requests when being sent to the SIP provider on the basis of credentials username in the request. I have pasted my config below. Please tell me what am I doing wrong here. In the kamctlrc file i have SIP_DOMAIN=localhost Thank you #------CONFIG BEGINS------------------ mpath="/lib/kamailio/modules_k/" debug=3 fork=yes children=4 auto_aliases=no alias=localhost alias=192.168.10.1 alias=192.168.10.2 alias=192.168.10.3 alias=192.168.10.4 alias=192.168.10.5 alias=192.168.10.6 disable_tcp=yes loadmodule "sl.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "/lib/kamailio/modules/tm.so" loadmodule "textops.so" modparam("rr", "enable_full_lr", 1) route { # Sanity Check # ------------ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if(msg:len>2048) { sl_send_reply("413", "message too large to be forwarded over UDP without fragmentation"); exit; }; # Record Route and NAT Preset # -------------------- if (method != "REGISTER") { record_route(); }; # Loose Route # ----------- if (loose_route()) { route(1); return; }; # Call Type Processing # -------------------- if (uri != myself) { route(1); return; }; if (uri == myself) { if (method == "BYE") { route(4); return; } else if (method == "CANCEL") { route(4); return; } else if (method == "INVITE") { route(3); return; } else if (method == "NOTIFY") { sl_send_reply("200", "Understood"); return; } else if (method == "OPTIONS") { sl_send_reply("200", "Got it"); return; } }; route(1); } # Default Message Handling # ----------------------- route[1] { t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } # INVITE Message Handling # ---------------------------------- # ---------------------------------- route[3] { if (uri =~ "^sip:011[0-9]@*") { rewritehostport("sip.voipprovider.com:5060"); if (search("^(Contact|m): .*user01*(a)(127\.0\.0\.1|localhost)")) { force_send_socket(192.168.0.2:5060); }; route(1); return; }; } # CANCEL and BYE Message Handling # ---------------------------------- route[4] { route(1); }

3 8

[Kamailio-Users] Rtpproxy hook
by Vikram Ragukumar 22 Feb '10

22 Feb '10

Hello, I am trying to encrypt/decrypt all packets belonging to an RTP stream. I am using Kamailio v1.5 with rtpproxy, where Kamailio forces the rtpstream through rtpproxy. Does rtpproxy provide a hook so that i can gain access to each RTP packet to perform encryption/decryption ? I do not want to end up having a patch for rtpproxy that i need to update whenever there is a new release of rtpproxy. -- Thanks and Regards, Vikram Ragukumar.

2 1

[Kamailio-Users] Error in Registration with TLS on
by Hemanshu Patel 22 Feb '10

22 Feb '10

Dear friends, Since last few days i am working on Kamailio with TLS support. I had followed each and every steps in installation docs...created certificates as well. Then i started testing the server with TLS on using SIPP. First i didnt added any certificate to SIPP, and Registration wasnt happening... When i added a certificate and key to SIPP....it started working fine....i was been able to test Registrations Successfully. Then i started working with one open source soft phone supporting TLS named mumble. IT Supports. Now i hadnt added any certificate to Mumblem. In my settings of kamailio i have set clietn_verify = 0 and require_client_certificate = 0. So without certificate as well i should be able to Authenticate my self successfully. Instead it gives following error in kamailio log: Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:print_ip: tcpconn_new: new tcp connection to: 172.16.16.218 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_new: on port 58125, type 3 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: entered: Creating a whole new ssl connection Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: looking up socket based TLS server domain [172.16.16.218:5091] Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_find_server_domain: socket based TLS server domain found Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: found socket based TLS server domain [172.16.16.218:5091] Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: Setting in ACCEPT mode (server) Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_add: hashes: 929, 1 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:handle_new_connect: new connection: 0x7fd6f4a58208 23 flags: 0002 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:send2child: to tcp child 0 0(3296), 0x7fd6f4a58208 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:handle_io: received n=8 con=0x7fd6f4a58208, fd=18 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_add: io_watch_add(0x73a0a0, 18, 2, 0x7fd6f4a58208), fd_no=1 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:tls_update_fd: New fd is 18 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: ERROR:core:tls_accept: SSL_accept failed: SSL_ERROR_SSL Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_del: io_watch_del (0x73a0a0, 18, -1, 0x10) fd_no=2 called Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: releasing con 0x7fd6f4a58208, state -2, fd=18, id=1 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: extra_data 0x7fd6f4a683a0 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:handle_tcp_child: reader response= 7fd6f4a58208, -2 from 0 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_destroy: destroying connection 0x7fd6f4a58208, flags 0002 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_close: closing SSL connection Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_update_fd: New fd is 23 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_shutdown: shutdown successful Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_clean: Cleanup function entered Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:print_ip: tcpconn_new: new tcp connection to: 172.16.16.218 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_new: on port 58126, type 3 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: entered: Creating a whole new ssl connection Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: looking up socket based TLS server domain [172.16.16.218:5091] Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_find_server_domain: socket based TLS server domain found Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: found socket based TLS server domain [172.16.16.218:5091] Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: Setting in ACCEPT mode (server) Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_add: hashes: 930, 2 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:handle_new_connect: new connection: 0x7fd6f4a58208 23 flags: 0002 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:send2child: to tcp child 0 0(3296), 0x7fd6f4a58208 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:handle_io: received n=8 con=0x7fd6f4a58208, fd=18 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_add: io_watch_add(0x73a0a0, 18, 2, 0x7fd6f4a58208), fd_no=1 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:tls_update_fd: New fd is 18 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: ERROR:core:tls_accept: SSL_accept failed: SSL_ERROR_SSL Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_del: io_watch_del (0x73a0a0, 18, -1, 0x10) fd_no=2 called Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: releasing con 0x7fd6f4a58208, state -2, fd=18, id=2 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: extra_data 0x7fd6f4a683a0 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:handle_tcp_child: reader response= 7fd6f4a58208, -2 from 0 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_destroy: destroying connection 0x7fd6f4a58208, flags 0002 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_close: closing SSL connection Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_update_fd: New fd is 23 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_shutdown: shutdown successful Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_clean: Cleanup function entered And in Mumble soft phone log it gives me following Error: [9:50 AM] Welcome to Mumble. [9:50 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. [9:51 AM] Reconnecting. [9:51 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. [9:51 AM] Reconnecting. [9:51 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. [9:51 AM] Reconnecting. [9:51 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. Can any one suggest what could be the problem? My Server works great with SIPP with TLS....so i dont think theres any config related error and i have set client_require_certificate = 0 thats for sure.... In real life scenario, hard or soft phones wont have certificates...so they should be able to connect to server and authenticate/Authorize themselves if server has valid certificate.But its not happening. So i need help from experienced guys.... -- Regards, Hemanshu Patel M: 09601295238

2 2

[Kamailio-Users] acc
by Anders 22 Feb '10

22 Feb '10

Hi, Probably a simpel fix some...not for me ;-) I get this here below in the log when I try to start Kamailio on a new installation. I suspect something is wrong with the acc module. What I read elsewhere is that it needs to recompile... - could that be, and if so, how would I recompile only a module? Thanks!! //Anders Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_config> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 140, column 25-26: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_flag> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 141, column 27-28: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_missed_flag> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 142, column 20-21: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_extra> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 157, column 2-3: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:main: bad config file (4 errors)

2 1

[Kamailio-Users] Demerits of using normal Authentication insted of radius or diameter auth
by Hemanshu Patel 21 Feb '10

21 Feb '10

Hello guys..... Recent now i had started taking keen interest in kamailio. I was testing kamailio, and i had test it under two scenario. first of all let me tell u about my configuration. I am testing on normal Desktop pc, installed kamailio,mysql, radius everything on same pc. I am using MYSQL anyway for both sort of testing. When i test with radius authentication, means using radius_www_authorize at max i am getting around 500 cps for Registration, but when i use normal www_authorize i am getting around 2000 CPS.... whopping difference guys..... So, i was wondering what is merits and demerits of not using Radius based authentication. Is it insecure if i use normal Authentication? Its not in my knowledge that radius do encryption...so i dont think by using radius it provides more security. What is your opinions about it? -- Regards, Hemanshu Patel M: 09601295238

3 3

[Kamailio-Users] Username of P-Asserted-Identity header
by Marco Bungalski 19 Feb '10

19 Feb '10

Hi, all! Is there an easy possibility to get the Username of the P-Asserted-Identity header? When i use $ai i get for example "sip:+4942317769999@testdomain.de". I need only "+4942317769999" in a variable. I know for example that i can use $fU and get only the Username (and $fu gives me the whole From URI).... Cheers, Marco Marco Bungalski GmbH Traversale 5 27283 Verden Telefon: +49 4231 - 776 9999 Fax: +49 4231 - 776 9998 Mobil: +49 172 4204774 e-mail: Marco(a)Bungalski.de www.bungalski.de www.t-m-net.de Geschäftsführender Gesellschafter: Marco Bungalski Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586

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[Kamailio-Users] CANCEL goes bananas!
by Anders 19 Feb '10

19 Feb '10

Ok, "bananas" might not be the right word, but it resembles what is happening ;-) When I get a CANCEL from my customer to Kamailio, it is not passed on to my vendor - this is with a specific customer using VoipSwitch. The result is that the phone called keeps ringing even though it's been (attempted) cancelled, and a sideeffect is that it goes nuts at the carrier side because it can send anymore and returns 480 or 486. Below is the dialog from the Session Progress to when the call is cancelled - and then it goes bananas. Any ideas? Thanks a lot! U 2010/02/12 14:40:25.435793 MyKamailioIP:5060 -> MyCustomerIP:5060 SIP/2.0 183 Session Progress. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Contact: <sip:84370551184517324@MyVendorIP:5060>. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. CSeq: 1 INVITE. Content-Type: application/sdp. Via: SIP/2.0/UDP MyCustomerIP:5060;rport=5060;received=MyCustomerIP;branch=z9hG4bk120246101250121320412156. Content-Length: 210. . v=0. o=prxams02 0 0 IN IP4 MyVendorIP. s=sip call. c=IN IP4 62.93.140.13. t=0 0. m=audio 42972 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 2010/02/12 14:40:30.625842 MyCustomerIP:5060 -> MyKamailioIP:5060 CANCEL sip:890912551184517324@MyKamailioIP SIP/2.0. CSeq: 1 CANCEL. Via: SIP/2.0/UDP MyCustomerIP:5060;branch=z9hG4bk120246101250121320412156. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. Contact: <sip:MyCustomerIP:5060;transport=udp>. Content-Length: 0. . U 2010/02/12 14:41:25.496605 MyVendorIP:5060 -> MyKamailioIP:5060 SIP/2.0 480 Temporarily Unavailable. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Contact: <sip:84370551184517324@MyVendorIP:5060>. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. CSeq: 1 INVITE. Via: SIP/2.0/UDP MyKamailioIP;branch=z9hG4bKf18c.fa7b81d2.0. Via: SIP/2.0/UDP MyCustomerIP:5060;rport=5060;received=MyCustomerIP;branch=z9hG4bk120246101250121320412156. Content-Length: 0. . U 2010/02/12 14:41:25.496888 MyKamailioIP:5060 -> MyVendorIP:5060 ACK sip:84370551184517324@MyVendorIP:5060 SIP/2.0. Via: SIP/2.0/UDP MyKamailioIP;branch=z9hG4bKf18c.fa7b81d2.0. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. CSeq: 1 ACK. Max-Forwards: 70. User-Agent: Kamailio. Content-Length: 0.

2 3

[Kamailio-Users] Testing of Kamailio TLS with Sipp TLS
by Hemanshu Patel 19 Feb '10

19 Feb '10

Hello friendsm, I am testing TLS feature of both kamailio and Sipp. I first downloaded kamailio 1.5.x TLS supported version, uncommented TLS=1 from Makefiel and then build the kamailio. first i test kamailio without TLS with sipp for registration and everything works file. Then i follow "http://www.kamailio.org/docs/tls-devel.html" and creates rootCA, user certificates and all configuration parameters to kamailio.cfg file Config paras are as below: /* uncomment the following lines to enable TLS support (default off) */ disable_tls = no listen = tls:172.16.16.218:5091 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/data/hemanshu/install/kam-tls/etc/kamailio/tls/user/user-cert.pem" tls_private_key = "/data/hemanshu/install/kam-tls/etc/kamailio/tls/user/user-privkey.pem" tls_ca_list = "/data/hemanshu/install/kam-tls/etc/kamailio/tls/user/user-calist.pem" and restart kamailio server It works and i can see via netstat on port 5091. but when i starts sipp it gives me following error. [hemanshu@localhost sipp.3.1]$ ./sipp -sf ./data/rauth.xml -inf ./data/user.csv -r 1 -m 1 -trace_err -trace_stat -nd -fd 1 -i 172.16.16.218 172.16.16.218:5091 -t l1 2010-02-18 13:51:40:244 1266481300.244432: FI_init_ssl_context: SSL_CTX_use_certificate_file failed. I know i have built sipp with TLS support, then i can not figure out where is the problem. Sipp says user certification file failed ..but client doesnt need any certification file...or does it? i even tried with tls_verify_client = 1 , tls_require_client_certificate = 1, playing with different combinations...but still same ans. Can anyone suggest me what could be wrong? Have i made any mistake in configuring kamailio or theres some problem in SIPP. -- Regards, Hemanshu Patel M: 09601295238

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[Kamailio-Users] Issue with dialog module
by Santiago Gimeno 19 Feb '10

19 Feb '10

Hi, I'm using kamailio 1.5 from the branch. In a conditional forwarding scenario with this call flow: Phone A Kamailio Phone B Phone C | | | | | | | | | | | | |INVITE | | | |------------->| | | |100 Trying | | | |<-------------| | | | |INVITE | | | |------------->| | | |100 trying | | | |<-------------| | | |180 Ringing | | | |<-------------| | |180 Ringing | | | |<-------------| | | | |486 Busy Here | | | |<-------------| | | |ACK | | | |------------->| | | |INVITE | | | |---------------------------->| | |100 trying | | | |<----------------------------| | |180 Ringing | | | |<----------------------------| |180 Ringing | | | |<-------------| | | | |200 OK | | | |<----------------------------| |200 OK | | | |<-------------| | | |ACK | | | |------------->| | | | |ACK | | | |---------------------------->| | | | | | | | | What I'm observing is that when Kamailio receives the 486 from B, the dialog between A and B is not destroyed, so when the call is finally established between A and C there are two active dialogs as I could check with `kamctl fifo get_statistics active_dialogs`. Shouldn't there be only one active dialog? Is this a problem in my script or a problem with the dialog module? Thanks in advance, Santi

3 3

[Serusers] tell me module of ser for nideo streaming..
by city spider 18 Feb '10

18 Feb '10

i want to use ser for streaming . please tell me the specific module for streaming... in streaming me using RTSP (Real Time Streaming Protocol) suggest me the specific module that will be added to fulfil my purpose.

1 0

[Kamailio-Users] Help me on 'No matching subscription dialog found'
by Asterisk User 18 Feb '10

18 Feb '10

Hi Group, I have one more problem regarding a SIP subscribe packet. When my phone registers first time, it gets OK for SUBSCRIBE sent from phone. But after couple of hours it(phone) keeps on sending subsequent SUBSCRIBE requests but the response from Kamailio is 481 Subscription does not exist. Here are my packets, =======================> For first SUBSCRIBE SUBSCRIBE sip:ABC.com SIP/2.0 Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-4082dbb From: DEF <sip:307@ABC.com>;tag=802bf23e23f4741 To: DEF <sip:307@ABC.com> Call-ID: 9f465e4b-2207f239(a)172.18.100.89 CSeq: 30004 SUBSCRIBE Max-Forwards: 70 Contact: DEF <sip:307@172.18.100.89:5060> Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 ------------------------------------------------------- SIP/2.0 202 OK Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-4082dbb;rport=5060;received=220.224.237.54 From: DEF <sip:307@ABC.com>;tag=802bf23e23f4741 To: DEF <sip:307@ABC.com>;tag=0cca86077d334bd53b00d2c8eab3d5fb.dfeb Call-ID: 9f465e4b-2207f239(a)172.18.100.89 CSeq: 30004 SUBSCRIBE Expires: 3600 Contact: <sip:192.168.94.30:5060> Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 =======================> For subsequent SUBSCRIBEs SUBSCRIBE sip:ABC.com SIP/2.0 Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-69712d2e From: DEF <sip:307@ABC.com>;tag=5d3bd6c2989f0839 To: DEF <sip:307@ABC.com>;tag=0cca86077d334bd53b00d2c8eab3d5fb.dfeb Call-ID: 7f6d2c36-97569155(a)172.18.100.89 CSeq: 55947 SUBSCRIBE Max-Forwards: 70 Contact: DEF <sip:307@172.18.100.89:5060> Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 ------------------------------------------------------- SIP/2.0 481 Subscription does not exist Via: SIP/2.0/UDP 172.18.100.89:5060;branch=z9hG4bK-69712d2e;rport=5060;received=220.224.237.54 From: DEF <sip:307@ABC.com>;tag=5d3bd6c2989f0839 To: DEF <sip:307@ABC.com>;tag=0cca86077d334bd53b00d2c8eab3d5fb.dfeb Call-ID: 7f6d2c36-97569155(a)172.18.100.89 CSeq: 55947 SUBSCRIBE Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 ---------------------------------------------------------------------------------------------------------------------------------------------- ===>The error which I get on kamailio is, ERROR:presence:get_database_info: No matching subscription dialog found in database ===>Have already loaded the module presence.so modparam("presence", "db_url", "mysql://XYZ:XYZ@192.168.90.12/openser") modparam("presence", "max_expires", 3600) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "db_update_period", 100) modparam("presence", "to_tag_pref", 'pres') modparam("presence", "expires_offset", 10) modparam("presence", "server_address", "sip:192.168.94.30:5060") modparam("presence", "fallback2db", 1) modparam("presence", "subs_htable_size", 11) modparam("presence", "pres_htable_size", 11) loadmodule "presence_mwi.so" loadmodule "presence_xml.so" modparam("presence_xml", "force_active", 1) modparam("presence_xml", "db_url", "mysql://XYZ:XYZ@192.168.90.12/openser") --------------------------------------------------------------------------------------------------------------------------------------------- Please help me on this. --SM

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[Kamailio-Users] starting
by Andrew Matthews 17 Feb '10

17 Feb '10

I'm looking for a tutorial of some sort to help me with getting openser running. What I need from it, are a basic proxy with both IP and username/password authentication, and an lcr. I've searched the net and can't seem to find what I need. I'm new to openser, I came from the asterisk world. If anyone can point me in the right direction I'd appreciate it. Thanks, Exstatica

2 1

Number Portability Functionality
by Asim Riaz 17 Feb '10

17 Feb '10

Hi list, Can someone explain me about PDB server? what I understand that PDB server have all the routing information for all the numbers and SIP-Router receive that information and route the call accordingly but how PDB server receive the routing data? Do we need to load this routing data manually? What happen if the routing information for a number is changed, do we need to update manually? Thanks, Asim

3 2

[Kamailio-Users] Inbound call no way audio
by RAJNIKANT VANZA 17 Feb '10

17 Feb '10

Hi Friends, Outbound calling works fine. When I receive an inbound call, the phone rings, but when I pick up the phone, the conversation does not get established. I have call from inside NAT extension number to Outside NAT extension ( Outboud call ) then call get established and two-way audio work fine but when I have call from Outside NAT extension to Inside NAT extension (Inbound call ) then phone rings, but when pickup the phone, audio is not present. During inboud call RTP packets are not transmit whereas Outboud call RTP packets transmit successful. Sometimes, Inbound and Outbound call only one-way audio present. Can anybody help me, or point me to a previous thread where this issue was discussed and solved? Thanks in advance. -- Best Regards, Rajnikant Vanza

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[Kamailio-Users] What is the best (and secure) way to change FROM_URI using regexp?
by Fernando Frota M Morais 16 Feb '10

16 Feb '10

Before sending this message I looked for an answer and found some options, two of them are followed: 1) SUBST function, sample listed here: if (is_present_hf("User-Agent")) { remove_hf("User-Agent"); remove_hf("P-Asserted-Identity"); subst("/^From: \".+\" (<sip: *[a-zA-Z0-9_\.-]+ <at> .*>)/From: \1/ig"); subst("/^To: \".+\" (<sip: *[a-zA-Z0-9_\.-]+ <at> .*>)/To: \1/ig"); } 2-a) uac_replace: if (from_uri=~"^sip:10804007\@voip.mydomain") { xlog("L_ALERT", "TESTANDO - Original From is ($fu-$fn)\n"); # uac_replace_from("sip:031" . $fU . "\(a)voip.mydomain"); uac_replace_from("Frota10804007","sip:03110804007@voip.mydomain"); xlog("L_ALERT", "TESTANDO - From changed to ($fu-$fn)\n"); }; OR : 2-b) uac _replace #2: if (from_uri=~"^sip:10804007\@.*") { xlog("L_ALERT", "Route 21 - Original from is ($fu-$fn)\n"); $avp(from)=$fu; # $avp(to)=$ru; # avp_pushto("$ru", "$avp(from)"); avp_subst("$avp(from)", "/(sip:)/sip:031/"); uac_replace_from("$avp(from)"); # xlog("L_ALERT", "Route 21 - From changed to ($fu-$fn)\n"); # avp_pushto("$ru", "$avp(to)"); }; } OR Looks like uac_replace is safer then "subst" function, because it can be reverted, but I did not found a way to use concatenation or any other kind of string operation as a parameter for it. I don't want to start a discussion, I need only a few lines showing me how to change from_uri_from from "sip:108xxxx" to "sip:031108xxxx". Thanks in advance, -- Fernando Frota Machado de Morais Divisao de Redes de Comunicacao Centro de Computacao Universidade Federal de Minas Gerais Brasil Tel. +55(31)3409.4007 Fax. +55(31)3409.4004

2 1

[Kamailio-Users] What is wrong with this NOTIFY packet
by Asterisk User 15 Feb '10

15 Feb '10

Hi Group, I know this issue has to do bit more with SIP and off from kamailio problems, but just wanted to know your opinions on this. I have my sip extensions registered to kamilio and Asterisk works as a voicemail server. Now to implement MWI i send NOTIFY packet from Asterisk to Kamailio and from kamailio to SIP extension (Grandstream GXP2000). here is the packet that travels from kamailio to extension and the response sent from extension to kamilio. =========================== NOTIFY sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com> SIP/2.0 Max-Forwards: 10 Record-Route: <sip:172.18.100.74;lr;nat=yes> Via: SIP/2.0/UDP 172.18.100.74;branch=z9hG4bK3c3.e1003f03.0 Via: SIP/2.0/UDP 127.0.0.1:45382 ;received=172.18.100.73;branch=z9hG4bK.4f83530d;rport=45382;alias Via: SIP/2.0/UDP 172.18.100.73:5060 From: <sip:111@arzoo.com <sip%3A111(a)arzoo.com>> To: <sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com>> Contact: <sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com>> Call-ID: 4d61cf9e505d40e905032a18329d61ec(a)172.19.100.42 CSeq: 1 NOTIFY User-Agent: VoiceMail Event: message-summary Content-Type: application/simple-message-summary Content-Length: 39 =========================== SIP/2.0 415 Unacceptable Content-Type Via: SIP/2.0/UDP 172.18.100.74;branch=z9hG4bK3c3.02003f03.0 Via: SIP/2.0/UDP 127.0.0.1:43416 ;received=172.18.100.73;branch=z9hG4bK.7be73901;rport=43416;alias Via: SIP/2.0/UDP 172.18.100.73:5060 Record-Route: <sip:172.18.100.74;lr;nat=yes> From: <sip:111@arzoo.com <sip%3A111(a)arzoo.com>> To: <sip:1111111@arzoo.com <sip%3A1111111(a)arzoo.com>>;tag=3c6ba3e4a93d6864 Call-ID: 4d61cf9e505d40e905032a18329d61ec(a)172.19.100.42 CSeq: 1 NOTIFY User-Agent: Grandstream GXP2000 1.1.4.14 Accept: application/sdp, application/simple-message-summary, application/octet-stream, application/dialog-info+xml, application/pidf+xml, message/sipfrag;version=2.0 Content-Length: 0 =========================== Please guide me in right direction to get rid of 'Unacceptable Content-Type' problem. Thanks in advance, --SM

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[Kamailio-Users] Problems with parallel forking to different users
by Anton Roman 15 Feb '10

15 Feb '10

Hi all, I'm trying to implement parallel forking to different users registered on a Kamailio server. I want to fork the call to a number of different users (not including the called user). To do this I added the piece of code below to the default 3.0.0 config file: ... # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting setflag(4); #dialog aware if(avp_db_load("$ru/username","$avp(s:fork)")) { avp_pushto("$ru/username","$avp(s:fork)/g"); }; } .... This is usr_preference table content: +----+------+----------+--------+-----------+------+-------+---------------------+ | id | uuid | username | domain | attribute | type | value | last_modified | +----+------+----------+--------+-----------+------+-------+---------------------+ | 4 | | 1000 | | fork | 0 | 1001 | | | 3 | | 1000 | | fork | 0 | 1005 | | +----+------+----------+--------+-----------+------+-------+---------------------+ Kamailio correctly forks the call to 1001 and 1005, but the Request-URI of the 1005 user is not correctly set, because the proxy IP is used instead of the IP where the user is registered (10.1.2.15). You can see the INVITEs below. Could anybody tell me what I'm doing badly? Thanks for your time, best regards Anton ---------------------------------------------------------------------------------------------------------------------------------------- # U +0.000328 10.1.20.103:5060 -> 10.1.3.18:5061 INVITE sip:1001@10.1.3.18:5061 SIP/2.0 Record-Route: <sip:10.1.20.103;lr=on;did=c34.3ed81d77> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKc40b.ec3572a3.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-213aa77 From: "1002" <sip:1002@10.1.20.103 <sip%3A1002(a)10.1.20.103> >;tag=280da520d15d62cfo1 To: "1000" <sip:1000@10.1.20.103 <sip%3A1000(a)10.1.20.103>> Call-ID: 8973c34-427096cb(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 69 Contact: "1002" <sip:1002@10.1.3.15:5061> Expires: 240 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 202 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 2003696 2003696 IN IP4 10.1.3.15 ... # U +0.000052 10.1.20.103:5060 -> 10.1.20.103:5060 INVITE sip:1005@10.1.20.103 <sip%3A1005(a)10.1.20.103> SIP/2.0 Record-Route: <sip:10.1.20.103;lr=on;did=c34.3ed81d77> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKc40b.ec3572a3.1 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-213aa77 From: "1002" <sip:1002@10.1.20.103 <sip%3A1002(a)10.1.20.103> >;tag=280da520d15d62cfo1 To: "1000" <sip:1000@10.1.20.103 <sip%3A1000(a)10.1.20.103>> Call-ID: 8973c34-427096cb(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 69 Contact: "1002" <sip:1002@10.1.3.15:5061> Expires: 240 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 202 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 2003696 2003696 IN IP4 10.1.3.15 ... ----------------------------------------------------------------------------------------------------------------------------------------

2 1

[Kamailio-Users] Registration Timeout Retransmission
by Hemanshu Patel 13 Feb '10

13 Feb '10

Hello to all users, I am very new to kamailio and just started using it. I have installed it and my phones are getting register successfully. But sometimes i am seeing a scenario, which makes my phones unusable for sometime. please see the condition as below | phone |--------------| Kamailio | | REGISTER-----------------> <----------------------401 | REGISTER-----------------> Tricky condition here | timeout <--------------200 OK phone shows itself as unregistered | REGISTER----------------> <--------------------401 After several retries phone stops to retry till say next 20 seconds, during that time though there's an entry in location table in kamailio (i am using mysql) my phone shows itself as unregistered phone. Due to load or any other conditions Server takes a bit time to send 200OK back to phone and During that time timer expires and phone sends REGISTER once again. now phone will keep sending REGISTER request with same nonse and other value, and kamailio is configured to not to entertained same credentials with nonce if it is authenticated once. So kamailio generates new 401 Message with new nonce value to which phone understands as the previous reply thet he send for previous 401 was rejected, and the most probable reason is wrong user credentials..... Hope you guys have got my problem? i can not allow kamailio to let use the same credentials+ response with nonce if it has used it once for security reason. Then what could be the solution for this. I am very good in Programming, but just that i have just started to use kamailio and not very much aware about its architecture. -- Regards, Hemanshu Patel M: 09601295238

3 5

[Serusers] stateful statistics
by Sabatella, Michael 12 Feb '10

12 Feb '10

I know somebody else posted this a while ago but I did not see any response. I tried to research it but could not come up with any answers and would like to get an understanding of this . On the stateful statistics what does 130 waiting and 6 current mean Stateful Statistics Current: 6 (130 waiting) Total: 3540496 (0 local) Replied localy: 3809867 Is it recommended to restart the process once a week or once a month? I am running ser 0.9.6 Thanks for the help. Michael ________________________________ 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.

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[Kamailio-Users] Error in header - where and how?
by Anders 12 Feb '10

12 Feb '10

Hi, My carrier doesn't like the invites that come from two specific customers I have, who both use something called VoipSwitch. It returns from the carrier as a Bad Request. Below is the dialog - I'm not sharp enough to see what's wrong - anyone? Thanks! //Anders U 2010/02/01 17:53:42.355689 CustomerIP:5060 -> MyProxyIP:5060 INVITE sip:890912551128650136@MyProxyIP SIP/2.0. CSeq: 1 INVITE. Via: SIP/2.0/UDP CustomerIP:5060;branch=z9hG4bk01025810150801381470578. From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. Call-ID: GW6d4c945d2c573c27@CustomerIP. To: <sip:890912551128650136@MyProxyIP>. Contact: <sip:CustomerIP:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 250. Remote-Party-ID: "SERVER BR"<sip:69912390@customerDNS>;privacy=off;screen=no. . v=0. o=VoipSwitch 9298 9298 IN IP4 CustomerIP. s=VoipSIP. i=Audio Session. c=IN IP4 CustomerIP. t=0 0. m=audio 8298 RTP/AVP 18 101. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2010/02/01 17:53:42.355990 MyProxyIP:5060 -> CustomerIP:5060 SIP/2.0 100 Giving a try. CSeq: 1 INVITE. Via: SIP/2.0/UDP CustomerIP:5060;branch=z9hG4bk01025810150801381470578;rport=5060;received=187.45.218.17$ From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. Call-ID: GW6d4c945d2c573c27@CustomerIP. To: <sip:890912551128650136@MyProxyIP>. Server: Kamailio. Content-Length: 0. . U 2010/02/01 17:53:42.356010 MyProxyIP:5060 -> CarrierIP:5060 INVITE sip:84370551128650136@CarrierIP:5060 SIP/2.0. Max-Forwards: 10. Max-Forwards: 5. Record-Route: <sip:MyProxyIP;lr=on;nat=yes;did=e4c.e1f33ad5>. CSeq: 1 INVITE. Via: SIP/2.0/UDP MyProxyIP;branch=z9hG4bK139a.a74691c5.0. Via: SIP/2.0/UDP CustomerIP:5060;rport=5060;received=CustomerIP;branch=z9hG4bk0102581015080138147057$ From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. Call-ID: GW6d4c945d2c573c27@CustomerIP. To: <sip:890912551128650136@MyProxyIP>. Contact: <sip:CustomerIP:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 250. Remote-Party-ID: "SERVER BR"<sip:69912390@customerDNS>;privacy=off;screen=no. . v=0. o=VoipSwitch 9298 9298 IN IP4 CustomerIP. s=VoipSIP. i=Audio Session. c=IN IP4 CustomerIP. t=0 0. m=audio 8298 RTP/AVP 18 101. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2010/02/01 17:53:42.608917 CarrierIP:5060 -> MyProxyIP:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP MyProxyIP;branch=z9hG4bK139a.a74691c5.0. Via: SIP/2.0/UDP CustomerIP:5060;rport=5060;received=CustomerIP;branch=z9hG4bk0102581015080138147057$ From: "SERVER BR" <sip:001184517324@CustomerIP:5060>;tag=010258101508381470562. To: <sip:890912551128650136@MyProxyIP>. Call-ID: GW6d4c945d2c573c27@CustomerIP. CSeq: 1 INVITE. Content-Length: 0.

2 4

[Kamailio-Users] Drouting Question
by Sven Schulz 12 Feb '10

12 Feb '10

Hello, Ive been testing Kamailio 3.0 and drouting. In the documentation it states (Section 1.5 Routing Rule Processing) ³The module will execute serial forking for each address in chain. The next address in chain is used only if the previously has failed.² What exactly defines a failure? Does it use ping or OPTIONS to determine if a GW is busy or down? Id like to know exactly how drouting figures this out. Sven Schulz Penn State University Telecommunications and Network Services 814.865.6116 sip:sven@psu.edu

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[Kamailio-Users] acc module
by Denis Putyato 12 Feb '10

12 Feb '10

Hello everybody! I have a such problem with ACC module. Calls was accounted but only INVITE and BYE requests has appeared in acc table. ACK is absent. Kamailio 3.0 was installed from GIT. Thank you for any help loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "tm.so" loadmodule "sl.so" loadmodule "tmx.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" loadmodule "siptrace.so" loadmodule "dialplan.so" loadmodule "lcr.so" loadmodule "sqlops.so" loadmodule "carrierroute.so" loadmodule "drouting.so" loadmodule "kex.so" # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 29) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "acc_to_tag_column", "to_tag") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "acc_time_column", "time") modparam("acc", "db_url", "xxxxxxxxxxxxxxxxxxxxxxxx") modparam("acc", "db_extra", "src_user=$avp(i:2);src_domain=$si;dst_ouser=$avp(i:1);dst_user=$rU;dst_doma in=$rd") route { sip_trace(); setflag(30); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); setflag(29); } if (is_method("ACK")) { xlog("L_NOTICE", "ACK detected"); # I see this line in syslog setflag(1); setflag(29); } route("relay"); exit; } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { setflag(1); route("relay"); exit; } else { exit; } } } }

1 0

[Kamailio-Users] acc & radius
by Kris Amy 12 Feb '10

12 Feb '10

Hi all, Is there anything special required to get radius support into the acc module? I have tried the following... make distclean make config modules_include="mediaproxy auth_radius group_radius uri_radius misc_radius" group_include="mysql standard" make deb But alas it still seems to not have it. I have confirmed that the radius libraries are installed aswell. ii libradiusclient-ng-dev 0.5.6-1 Enhanced RADIUS client library development f ii libradiusclient-ng2 0.5.6-1 Enhanced RADIUS client library Cheers, Kris

2 1

[Kamailio-Users] Extra byte in Content-length header
by Vikram Ragukumar 11 Feb '10

11 Feb '10

Hello, I am testing a cellphone software that registers with an Asterisk server through an outbound proxy (Kamailio+rtpproxy). I am capturing network traffic using wireshark and i notice that the SIP Register message from the cellphone has an extra byte in the Content-Length header, which causes Kamailio to discard the message with the following error output : - ERROR:core:parse_content_length: parse error near char [13][^M] - ERROR:core:get_hdr_field: bad content_length header Could it be that this byte was appended somewhere downstream(possibly at the wireless router..) or can i be sure that the byte was added during SIP Message construction by the cellphone software ? Thanks in advance, Vikram.

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Re: [Kamailio-Users] Siremis Accounting
by Daniel-Constantin Mierla 11 Feb '10

11 Feb '10

Hello, please use public mailing list for such questions. Direct messages get long delayed or lost in most of the cases. Mailing list has plenty of developers and community users that can help you faster. Besides that, new people can benefit of older discussions. On 2/11/10 2:14 AM, Kris Amy wrote: > Hi Daniel, > > It would seem to me that the stored procedure on > http://siremis.asipto.com/install-accounting/ is for PostgreSQL. I'm > going to try and getting it working with MySQL. it is for mysql. Cheers, Daniel > > Just thougth you may want to update the page. > > -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 * http://www.asipto.com/

4 3

Re: [Kamailio-Users] Siptrace and Kamailio 3.0.0
by Daniel-Constantin Mierla 11 Feb '10

11 Feb '10

On 2/11/10 11:16 AM, Kris Amy wrote: > On Thu, Feb 11, 2010 at 8:09 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > ok, then the only point in mod_init of siptrace that causes > initialization error without printing a log message is when you > haven't set the trace_flag parameter: > http://kamailio.org/docs/modules/3.0.x/modules_k/siptrace.html#id2520001 > > Cheers, > Daniel > > > Hi Daniel, > > That fixed it. > > Perhaps there is a bug in that it's not picking up the default of 0? in the sources the default is -1, which means disabled -- docs needs to be updated. Thanks for pointing, Daniel -- Daniel-Constantin Mierla eLearning class for Kamailio 3.0.0 *http://www.asipto.com/

1 0

[Kamailio-Users] SDP IP address not rewritten
by Vikram Ragukumar 11 Feb '10

11 Feb '10

Hello, I am running Kamailio version 1.5 and rtpproxy version 1.2. I have enabled NAT support in Kamailio.cfg. I have a UA behind NAT (UA IP:192.168.2.101), and the UA registers with an Asterisk server (Public IP) through an outbound proxy running Kamailio and rtpproxy with Public IP. Registration succeeds, however when making an outbound call from the UA to a cellphone, there is only one sided voice. This is happening because Asterisk is addressing rtp streams to the UA's local IP address instead of its PublicIP:port. Why is the IP address not being rewritten in the SDP ? Is there an alternate way of invoking rtpproxy (with flags) such that the IP address in the SDP is rewritten ? Thanks and Regards, Vikram. PS : Below is the main routing section of my kamailio.cfg route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } 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; } } sl_send_reply("404","Not here"); } exit; } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (isflagset(5)) { setbflag(6); # setbflag(7); } if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); setflag(6);#VR } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } }

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[Kamailio-Users] [kamailio-users] Method NOTIFY not supported ...
by Chandrakant Solanki 11 Feb '10

11 Feb '10

Hi below line is in my configuration... loadmodule "pua.so" modparam("pua", "db_url", "mysql:// kamailio:kamailio@172.16.0.10:3306/kamailio") modparam("pua", "db_table", "pua") ---- ---- if(is_method("NOTIFY") && src_ip==172.16.0.15) { xlog("L_INFO", "NOTIFY Method found\n"); if(!t_relay()){ sl_reply_error(); } } ---- ---- I found following error in log... Feb 10 15:14:40 kamailio_1 /sbin/kamailio[5597]: Method not supported - M=NOTIFY RURI=sip:172.16.0.15 F=sip:2222222@pbxusers.net<sip%3A2222222(a)pbxusers.net>T=sip:172.16.0.15 IP=172.18.100.89 ID= 17dd4bb9-81d132a0(a)172.18.100.89 -- Regards, Chandrakant Solanki

2 7

[Kamailio-Users] Fwd: Siptrace and Kamailio 3.0.0
by Kris Amy 11 Feb '10

11 Feb '10

---------- Forwarded message ---------- From: Kris Amy <kris(a)amy.id.au> Date: 2010/2/11 Subject: Re: [Kamailio-Users] Siptrace and Kamailio 3.0.0 To: Daniel-Constantin Mierla <miconda(a)gmail.com> On Thu, Feb 11, 2010 at 8:09 PM, Daniel-Constantin Mierla <miconda(a)gmail.com > wrote: > Hello, > > ok, then the only point in mod_init of siptrace that causes initialization > error without printing a log message is when you haven't set the trace_flag > parameter: > http://kamailio.org/docs/modules/3.0.x/modules_k/siptrace.html#id2520001 > > Cheers, > Daniel >

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