July 2009 - sr-users - kamailio.org

[Kamailio-Users] Kamailio Revision: 5487 1.4.3-notls (x86_64) -m 4096 -- Crashing Randomly
by Brandon Armstead 17 Jul '09

17 Jul '09

Hey guys, Here is the response from backtrace: Core was generated by `/sbin/kamailio -P /var/run/kamailio.pid -m 4096'. Program terminated with signal 11, Segmentation fault. [New process 29118] #0 0x00007f37499be44d in relay_reply (t=0x7f3645019f88, p_msg=0x857350, branch=0, msg_status=487, cancel_bitmap=0x7fff53aa3428) at t_reply.c:1101 1101 relayed_code=relayed_msg->REPLY_STATUS; (gdb) bt #0 0x00007f37499be44d in relay_reply (t=0x7f3645019f88, p_msg=0x857350, branch=0, msg_status=487, cancel_bitmap=0x7fff53aa3428) at t_reply.c:1101 #1 0x00007f37499bf8fb in reply_received (p_msg=0x857350) at t_reply.c:1404 #2 0x00000000004218fc in forward_reply (msg=0x857350) at forward.c:507 #3 0x000000000045cf35 in receive_msg ( buf=0x744aa0 "SIP/2.0 487 Request Terminated\r\nTo: <sip:+1XXXXXXXXXX@PSTN:5060>;tag=63c9393246a9a5d6i0\r\nFrom: \"XXXXX X & X \" <sip:XXXXXXXXXX@PSTN>;tag=VPSF506071629460\r\nCall-ID: STMMGC012009071521"..., len=997, rcv_info=0x7fff53aa3520) at receive.c:203 #4 0x000000000049c486 in udp_rcv_loop () at udp_server.c:449 #5 0x000000000042882b in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:780 As far as syslog/kernel message: Jul 15 21:07:28 sip-XXXX kernel: [526828.616918] kamailio[29118]: segfault at 40 ip 7f37499be44d sp 7fff53aa32b0 error 4 in tm.so[7f374997f000+4d000] The server load seems to randomly jump to ~ 8 - 10 and this happens, any thoughts / help is appreciated, thanks! One other thing that may be relevant, is we have disabled TCP support on this proxy, because previously (I have another mailing list thread) about, with TCP enabled kamailio will not start due to epoll alloc failure, thanks!

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[Serusers] Ser & RTPproxy does not change the SDP vodeo port.
by Amos Shahar 16 Jul '09

16 Jul '09

hi, I'm using ser 0.9.7 and RTPproxy 1.1 on diffrent servers. the ser.cfg is the nat-rtpproxy.cfg from the iptel site. I have 2 grandstream GXV-300 devices, each behind diffrent NAT, this setup is working fine for voice and RTP is relayed through the rtpproxy properly. The problem is that the video is not working, when ser gets the invite from device A and send it to device B, it change the audio port only but leave the video port the same as the port he got from device A. I'm not sure if it is an issue with the RTPproxy or the nat-helper module. any idea how to debug / resolve? do you aware of other versions of ser/rtpproxy that this setup should work with video? Thanks! Amos

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[Kamailio-Users] received=IP added in Via
by catalina oancea 16 Jul '09

16 Jul '09

Hello, Does anybody know how to disable the received=IP Via parameter that is added by default from kamailio? For the case I want to disable it in, I have no NAT handling. Also, I have: dns=no rev_dns=no Anything else? Thanks Catalina

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[Kamailio-Users] Clik to dial
by ALAEDDINE abbech 16 Jul '09

16 Jul '09

Hello every one I must use Openser alone to do the aplication clik to dial, who have an idea to do this application?? regards, Alasupcom

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[Kamailio-Users] OpenSER stability problems in pilot project
by Munder Albert (CI/ISE) 16 Jul '09

16 Jul '09

Hello, Sorry, next try with posting this message. Attempts with attachment failed due to message size. We are running OpenSER in a pilot project and unfortunately have some stability problems. Any help or hints are appreciated. Project information OpenSER is used in a pilot project with * Appr. 5000 subscriber accounts * Appr. 1200 simultaneously registered users * Signalling encrypted with TLS * Media data encrypted with SRTP * Clients: softphones and hardphones * Re-registration time for clients: 3600 sec OpenSER configuration · Works as stateful SIP Proxy 1 mySQL database 2 Version 1.3.4.-TLS 3 Tcp_children: 100 --> is it recommended to increase this number? 4 Udp_children: 20 5 Tcp_connection_timeout: 3600 6 Shared memory: · -m 512 when error occurred 1 Now set to 1024 Problems * Shared memory consumption Shared memory usage is permanently increasing (about 50 MB per day) Application already crashed twice First messages were, these, repeated thousands of times (5915 times): Jun 17 08:54:52 si-.... /usr/local/sbin/openser[13921]: ERROR:core:tcpconn_new: shared memory allocation failure Jun 17 08:54:52 si-... /usr/local/sbin/openser[13921]: ERROR:core:handle_new_connect: tcpconn_new failed, closing socket And a few of these also (7613 times): Jun 17 08:57:24 si-... /usr/local/sbin/openser[13880]: ERROR:core:tls_accept: some error in SSL: Jun 17 08:57:24 si-... /usr/local/sbin/openser[13880]: ERROR:core:tls_print_errstack: error:1409C041:SSL routines:SSL3_SETUP_BUFFERS:malloc failure * TCP errors, lost SIP messages Examples from error messages: 14.100 times in log file from 17.06.09 Jun 17 04:03:15 si-... /usr/local/sbin/openser[13863]: ERROR:core:tcp_blocking_connect: poll error: flags 18 Jun 17 04:03:15 si-... /usr/local/sbin/openser[13863]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused Jun 17 04:03:15 si-... /usr/local/sbin/openser[13863]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jun 17 04:03:15 si-... /usr/local/sbin/openser[13863]: ERROR:core:tcp_send: connect failed Jun 17 04:03:15 si-.. /usr/local/sbin/openser[13863]: ERROR:tm:msg_send: tcp_send failed Jun 17 04:03:15 si-... /usr/local/sbin/openser[13863]: ERROR:tm:t_forward_nonack: sending request failed Appears at least 20 000 times; and in the day of the last shared memory errors, it was 225.794 times in the log file (note that the number in parenthesis is usually 1 or 2, but on that day it has reached 6): Jun 17 09:01:27 si-.... /usr/local/sbin/openser[13921]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (6) Jun 17 09:01:27 si-... /usr/local/sbin/openser[13921]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (5) * Certificate validation problems TCP traffic is currently significantly increased by some ( appr. 70) clients which failed to validate the TLS certificate. Registration is repeated every 5 sec. Circa 30 thousand per day (on that day, it was 37.162 times in log) Jun 17 04:03:10 si-024lc008 /usr/local/sbin/openser[13801]: ERROR:core:tls_accept: some error in SSL: Jun 17 04:03:10 si-024lc008 /usr/local/sbin/openser[13801]: ERROR:core:tls_print_errstack: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca Mit besten Grüßen | Best regards Albert Munder Robert Bosch GmbH IT Systems Engineering (CI/ISE) Postfach 30 02 20 70442 Stuttgart GERMANY www.bosch.com Tel. +49 711 811-40562 Fax +49 711 811-5113333 Albert.Munder(a)de.bosch.com Robert Bosch GmbH, Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart HRB 14000 Aufsichtsratsvorsitzender: Hermann Scholl; Geschäftsführung: Franz Fehrenbach, Siegfried Dais; Bernd Bohr, Wolfgang Chur, Rudolf Colm, Gerhard Kümmel, Wolfgang Malchow, Peter Marks; Volkmar Denner, Peter Tyroller.

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[Kamailio-Users] no RTP randomly with rtpproxy
by Samuel Muller 16 Jul '09

16 Jul '09

A strange thing happens with RTP proxy since I compiled the v.1.2.0 release ... Radomly, (approx. 1 time per 10 calls) RTP proxy don't grab the caller's original source IP in the SDP ??? So I don't have any RTP, so no sound during the call. typical call schema : sip phone -> mitel ipbx -> kamailio -> audiocodes mediant 2000 -> pstn All the SIP headers and the SDP are ok each time ... the RTP ports are ok too, understood at each point. rtp proxy is launched with these options with a public IP address (and compiled with a modified port range) : /usr/sbin/rtpproxy -s udp:77.246.81.133 35000 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 77.246.81.133 -m 6000 -M 64000 kamailio is ok too in the config : kamailio IPv4 UDP sip.720.fr:46830-> rtpproxy.720.fr:35000 logs of RTP proxy (option -f to screen the debug) : # rtp ok : received command "29638_4 L 3962688592-61598521(a)10.33.146.4 77.246.81.136 35000 0_3962688592-61598523;1 1c1320846358;1" lookup on ports 6008/6010, session timer restarted pre-filling callee's address with 77.246.81.136:35000 sending reply "29638_4 6010 77.246.81.133 " caller's address filled in: 94.198.149.41:50186 (RTP) guessing RTCP port for caller to be 50187 # rtp not ok : received command "29571_2 L 3288148592-61598465(a)10.33.146.4 77.246.81.136 35000 0_3288148592-61598467;1 1c934295778;1" lookup on ports 6000/6002, session timer restarted pre-filling callee's address with 77.246.81.136:35000 sending reply "29571_2 6002 77.246.81.133 " and hop, no "caller's address filled in" ... it's exactly the same type a SIP call (with the same ip phone, etc...). I can precise that rtp proxy is running on the same machine that kamailio (v.1.4.4), but listen on a dedicated sub-if. Does someone knows something about that ? thanks ! .Samuel Muller. sml(a)720.fr

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[Kamailio-Users] help for Multi-Domain
by zhangchao00001 16 Jul '09

16 Jul '09

Dear sir: I write to ask if the kamailio support multi-domain. for example, A UA belong to domain b get into domain a, could it work as well as in domain b? If so, how to configure it? Thank you.

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[Kamailio-Users] sharing audio between SIP client and MSN client
by rachid 16 Jul '09

16 Jul '09

Hello all, I use kamailio to have sharing chat between a SIP client and MSN client Now i want to have a sharing audio between them but i don't know witch module to use?? someone can help me please -- ------------------------------------------------------------ BAHA Rachid Ingénieur Télécoms, INPT-Rabat GSM:(+33) 0619787609

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Re: [Kamailio-Users] softswitch
by peter brooks 16 Jul '09

16 Jul '09

i would like to offer people pc2pc phone service.can i do this with you software do you have a php admin page? can you install the software for a price? _________________________________________________________________ Windows Live™ SkyDrive™: Get 25 GB of free online storage. http://windowslive.com/online/skydrive?ocid=TXT_TAGLM_WL_SD_25GB_062009

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[Serusers] Check out my photos on Facebook
by Tuhin Saiful 16 Jul '09

16 Jul '09

Hi serusers(a)iptel.org, 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, Saiful To sign up for Facebook, follow the link below: http://www.facebook.com/p.php?i=1086097402&k=Z513ZV52PWZCUGAGYKYTWR&r 557896866 was invited to join Facebook by Tuhin Saiful. 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=cebebd&u=557896866&mid=c8ac3aG2140d4a2G0G8 Facebook's offices are located at 1601 S. California Ave., Palo Alto, CA 94304.

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[Kamailio-Users] question about $branch(name) PV
by Klaus Darilion 15 Jul '09

15 Jul '09

Hi! Does $branch(name) also handle the "main branch" ($ru, $du) as a branch, or will it handle only the additional branches? thanks klaus

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[Kamailio-Users] Carrierroute-Kamailio-1.4.3-Strip does not strip in the to: header
by Omar Mendoza 15 Jul '09

15 Jul '09

I have a prefix, and i remove using the Strip in the DB INVITE sip:8135553333@192.168.2.77 SIP/2.0 Record-Route: <sip:99998135553333@198.168.1.170;lr=on;ftag=as1074edd2> Via: SIP/2.0/UDP 198.168.1.170;branch=z9hG4bKa5b6.ce53.0 Via: SIP/2.0/UDP 198.168.1.180:5060;branch=z9hG4bK292133e9;rport=5060 From: "Omar" <sip:5125555555@198.168.1.180>;tag=as1074edd2 To: <sip:99998135553333@198.168.1.170> Contact: <sip:5125555555@198.168.1.180> __________________________ Omar

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[Kamailio-Users] kamctl cr dump - Command not available on Kamailio 1.5.2
by Edson - Lists 15 Jul '09

15 Jul '09

Seems that an error is still there... Kamailio-151:~/sip-server # kamctl cr dump 500 command 'cr_dump' not available Kamailio-151:~/sip-server # kamailio -V version: kamailio 1.5.2-notls (i386/linux) flags: STATISTICS, USE_IPV6, USE_TCP, USE_SCTP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:5901M @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 13:29:16 Jul 14 2009 with gcc 4.3 Edson.

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[Serusers] RTPproxy Fork Command
by Jeremy Valcourt 14 Jul '09

14 Jul '09

Hi, According to the RTPproxy 1.1 release notes you can fork a stream (http://www.rtpproxy.org/wiki/RTPproxy/RELENG_1_1_0_NOTES) : "The new C (copy) command has been introduced allowing to fork off one or both RTP streams in the session to some third-party in real time." However, after spending an afternoon searching the Internet, I can't for the life of me find any documentation on how to use this feature. So, how do you fork a stream with RTPproxy? Any help is greatly appreciated. Jeremy V.

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[Kamailio-Users] OpenSer/Kamailio and A2Billing
by Carlos A. Alvarez 14 Jul '09

14 Jul '09

Folks, First of all I will like to apologize before hand, since I am better with numbers than with words. Therefore I hope you all can at least understand the essence of what I am trying to communicate. ;-) I am trying to set up an application, where I use Kamailio, asterisk, and A2Billing to process pre paid calling cards, and wholesale voip. I already have setup a kamailio service using carrier route and a2billing as a preprocessor. It works beautiful, Great work guys!!!! The questions I have are around the design for resiliency and redundancy. I will like to use another instance of kamailio for a dispatcher function to load balance between asterisk servers, since the is a limitation and performance issues with asterisk. Now I have done some research and some testing and they only way I know I can make it work is by using 302 redirection to redirect the invites from my customers to the corresponding asterisk server. If I use a direct setup using ds_select_dst asterisk gets all "confused" since the request appears to come from the dispatcher ip address. For wholesale ip, I filter a2billing on ip address and not username. My configuration is as follows: <customer>------><dispatcher/kamailio>------><a2billing/asterisk>-----><carrierroute/kamailio>------><carrier> Any thoughts, comments and/or suggestions are more than welcomed! Carlos

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[Kamailio-Users] Kamailio (OpenSER) v1.5.2 Released
by Daniel-Constantin Mierla 14 Jul '09

14 Jul '09

A new release in 1.5 series is out. Kamailio (OpenSER) 1.5.2 is based on the latest version of branch 1.5, including many fixes in code and documentation, therefore those running 1.5.0 or 1.5.1 are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.5.2/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/1.5.2/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.5.2/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Binaries and packages will be uploaded at: http://www.kamailio.org/pub/kamailio/1.5.2/ Modules' documentation: http://www.kamailio.org/docs/modules/1.5.x/ What is new in 1.5.x release series is summarized in the announcement of v1.5.0: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes Note: Kamailio is the name of former OpenSER project. First version under Kamailio name was 1.4.0. Older versions will continue to use OpenSER name. The source tree for current development version is hosted by SIP Router project, you can test it via: http://sip-router.org/wiki/migration/kamailio-3.0-config Many thanks to everybody that helped with this release! Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] reminder: kamailio 1.5.2 today
by Daniel-Constantin Mierla 14 Jul '09

14 Jul '09

Hello, anyone that still has to commit something on SVN branch 1.5 @sourceforge.net, please do it before 14:00GMT, afterwards will start releasing 1.5.2. Thanks, Daniel

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[Kamailio-Users] aliases
by asd 14 Jul '09

14 Jul '09

Hello, I would like to assign multiple DIDs (not sure how it's called in Kamailio) for one of my sip user. This sip user is registering with 7000 as username, and I would like to assign 7001, 7002 numbers to this sip user. I have tried using dbaliases function, so I have added 7001 and 7002 as aliases for 7000. If I call 7002, then alias is retrieved fine from db, I have no problem with that. But 7000 is receiving this call as 7000 (so he see 7000 as a called number, instead of 7002). is there a way to send "called number 7002" to 7000? Kind regards, Dmitri

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[Kamailio-Users] [nat][registrar] setting max_expires dynamically
by Pascal Maugeri 14 Jul '09

14 Jul '09

Hi How can we set the expires value of the REGISTER OK sent back to the clients in a dynamic way ? We set already a value with modparam("registrar", "max_expires", 120) but if the UA has been detected to be behind a NAT we want to apply a different value. Regards, Pascal

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Re: [Kamailio-Users] preparing kamailio 1.5.2
by Daniel-Constantin Mierla 14 Jul '09

14 Jul '09

Hello, thanks for testing and reporting back. Please cc all the time to the list, so everybody interested gets updated about conclusions. Daniel On 14.07.2009 0:05 Uhr, Marco Bungalski wrote: > Hello, Daniel! > > Thank you for you documentation - with this my linux-knowledge increased .-) > > > I've just downloaded an testet - now Kamailio don't crash with a loaded > dialog-module after > a receive of SIP-message containing an empty record-route-field. So good > luck for tomorrow :) > > Message now: > Jul 13 23:59:37 [23474] ERROR:core:print_rr_body: failed to parse RR > Jul 13 23:59:37 [23474] ERROR:dialog:populate_leg_info: failed to print > route records > > > Thanks and cheers, > > Marco > > > -----Ursprüngliche Nachricht----- > Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Gesendet: Montag, 13. Juli 2009 17:05 > An: Marco Bungalski; kamailio > Betreff: Re: AW: AW: [Kamailio-Users] preparing kamailio 1.5.2 > > Hello Marco, > > On 13.07.2009 14:28 Uhr, Marco Bungalski wrote: > >> Hello, Daniel, >> >> i know how to compile, but where to download the version with the bugfix >> included? >> >> > if you fetch from svn the branch 1.5 then it has the fix. Here is a > short description of how the versioning was done so far for kamilio > (openser) from svn: > - when a major version is released, a branch with the first two numbers > in the version is created (e.g., 1.5) > - for each minor version (including the .0 - corresponding to the major > version) a tag is created with name of the three numbers in version > - any fix for the major version is committed in the branch. Once a new > minor release is out, a new tag is created. Therefore, the branch has > all the time the most updated code for a specific major release. The > tags are snapshots at the time of release. > > For the case of 1.5.x series: > - there is one branch 1.5 > - now there are two tags on svn: 1.5.0 and 1.5.1, soon there will be 1.5.2. > - the tarballs are created from a svn checkout done as exampled in the > wiki page I sent to you. > > probably reading a bit about svn (subversion) will help you. Anyhow, > next major version will use GIT. > > Now, just to download the latest 1.5.x version, do: > svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 > kamailio-1.5 > > You need subversion package installed. The continue installation as you > do with the tarball. > > Cheers, > Daniel > >> Hello, >> >> this one should help you: >> http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn >> >> Cheers, >> Daniel >> >> On 07/13/2009 01:30 PM, Marco Bungalski wrote: >> >> >>> Hello, >>> >>> i am sorry to say that i've not testet it because i don't know where to >>> download the fix and how to implementate that :-( >>> >>> But if you give me a link i will test it today (i can simulate the bug) >>> >>> >> and >> >> >>> will send you a feedback. >>> >>> Greetings, >>> >>> Marco >>> >>> >>> >>> >>> Mit freundlichen Grüßen >>> >>> Marco Bungalski >>> >>> >>> 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 >>> web: www.bungalski.de >>> >>> Geschäftsführender Gesellschafter: Marco Bungalski >>> Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586 >>> >>> -----Ursprüngliche Nachricht----- >>> Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Gesendet: Montag, 13. Juli 2009 12:32 >>> An: Marco Bungalski >>> Cc: users(a)lists.kamailio.org >>> Betreff: Re: [Kamailio-Users] preparing kamailio 1.5.2 >>> >>> Hello >>> >>> On 07/13/2009 12:28 PM, Marco Bungalski wrote: >>> >>> >>> >>>> Hello, Daniel! >>>> >>>> >>>> >>>> Will the bug "Kamailio crashes when receiving an empty >>>> Record-Route-field with using the dialog-module" be fixed in 1.5.2? >>>> >>>> >>>> >>>> If yes: Thanks for that :-) >>>> >>>> >>>> >>>> >>> it should be already in the svn (therefore yes). Have you tested it? >>> >>> Cheers, >>> Daniel >>> >>> >>> >>> >> >> > > >

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[Kamailio-Users] new module: mi_rpc
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello, several days ago, a new module named mi_rpc was introduced to source tree. The goal is to be able to execute MI commands using RPC interface from sip router core. One big advantage of this is the executing MI commands using the sercmd tool. sercmd is a command line interface, able to do auto completion for commands, used to change parameters, get insights of core and modules at runtime. If you grabbed the source code (some guidelines at: http://sip-router.org/wiki/migration/kamailio-3.0-config), then: cd utils/sercmd make ./sercmd See the readme for more options. You need to compile and load modules_s/ctl: http://sip-router.org/docbook/sip-router/branch/master/modules_s/ctl/ctl.ht… Back to mi_rpc, one issue that needs to be sorted out is the output format. Right now doing a pretty-format printing which is not suitable for xmlrpc. Feedback is very much appreciated, thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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Re: [Kamailio-Users] How to know if a UAC is behind NAT after fix_nated_register() ?
by Pascal Maugeri 13 Jul '09

13 Jul '09

I answer to myself (it may help others) as I could test successfully the configuration below. Also there is a typo (isbflagset instead of isflagset): so when I detect a UA is behind NAT, I do: if (nat_uac_test("19")) { if (method=="REGISTER") { setbflag(7); fix_nated_register(); ... Then when my B2BUA sends invite to this UA I do (after lookup("location") ): lookup("location"); ... if (is_method("INVITE")) { if (isbflagset(7)) { setflag(5); Regards, Pascal On Mon, Jul 13, 2009 at 2:43 PM, Pascal Maugeri <pascal.maugeri(a)gmail.com>wrote: > I'm not that familiar with flag management. > > so when I detect a UA is behind NAT, I do: > > if (nat_uac_test("19")) { > if (method=="REGISTER") { > setbflag(7); > fix_nated_register(); > ... > > Then when my B2BUA sends invite to this UA I do (after lookup("location") > ): > > if (is_method("INVITE")) { > if (isflagset(7)) { > setflag(5); > > Is that correct ? > > BTW, regarding the flag number, is there a table of reserved numbers ? Can > I use "7" in this case ? > > -pascal > > > > > On Mon, Jul 13, 2009 at 12:00 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> On 07/13/2009 11:54 AM, Pascal Maugeri wrote: >> >>> Hi >>> >>> A quick question, when I apply a fix_nated_register() to a remote user >>> when he/she registers, how can I know later on if this user is behind NAT ? >>> >> set a branch flag for REGISTER before save location. After lookup location >> that flag is recovered and you can test it in the config file. >> >> Cheers, >> Daniel >> >> >>> All my calls go through a B2BUA: >>> >>> Caller ------> kamailio/rtpproxy ------> B2BUA ------> kamailio/rtpproxy >>> ------> Callee >>> >>> When my B2BUA calls a user, I want to know if this user is behind NAT or >>> not. For now all the calls go through rtpproxy but I would like to apply >>> this path only when user is behind NAT. >>> >>> Regards, >>> Pascal >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> 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 >> http://www.asipto.com/ >> >> >

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Re: [Kamailio-Users] preparing kamailio 1.5.2
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello Marco, On 13.07.2009 14:28 Uhr, Marco Bungalski wrote: > Hello, Daniel, > > i know how to compile, but where to download the version with the bugfix > included? > if you fetch from svn the branch 1.5 then it has the fix. Here is a short description of how the versioning was done so far for kamilio (openser) from svn: - when a major version is released, a branch with the first two numbers in the version is created (e.g., 1.5) - for each minor version (including the .0 - corresponding to the major version) a tag is created with name of the three numbers in version - any fix for the major version is committed in the branch. Once a new minor release is out, a new tag is created. Therefore, the branch has all the time the most updated code for a specific major release. The tags are snapshots at the time of release. For the case of 1.5.x series: - there is one branch 1.5 - now there are two tags on svn: 1.5.0 and 1.5.1, soon there will be 1.5.2. - the tarballs are created from a svn checkout done as exampled in the wiki page I sent to you. probably reading a bit about svn (subversion) will help you. Anyhow, next major version will use GIT. Now, just to download the latest 1.5.x version, do: svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio-1.5 You need subversion package installed. The continue installation as you do with the tarball. Cheers, Daniel > Hello, > > this one should help you: > http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn > > Cheers, > Daniel > > On 07/13/2009 01:30 PM, Marco Bungalski wrote: > >> Hello, >> >> i am sorry to say that i've not testet it because i don't know where to >> download the fix and how to implementate that :-( >> >> But if you give me a link i will test it today (i can simulate the bug) >> > and > >> will send you a feedback. >> >> Greetings, >> >> Marco >> >> >> >> >> Mit freundlichen Grüßen >> >> Marco Bungalski >> >> >> 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 >> web: www.bungalski.de >> >> Geschäftsführender Gesellschafter: Marco Bungalski >> Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586 >> >> -----Ursprüngliche Nachricht----- >> Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Gesendet: Montag, 13. Juli 2009 12:32 >> An: Marco Bungalski >> Cc: users(a)lists.kamailio.org >> Betreff: Re: [Kamailio-Users] preparing kamailio 1.5.2 >> >> Hello >> >> On 07/13/2009 12:28 PM, Marco Bungalski wrote: >> >> >>> Hello, Daniel! >>> >>> >>> >>> Will the bug "Kamailio crashes when receiving an empty >>> Record-Route-field with using the dialog-module" be fixed in 1.5.2? >>> >>> >>> >>> If yes: Thanks for that :-) >>> >>> >>> >> it should be already in the svn (therefore yes). Have you tested it? >> >> Cheers, >> Daniel >> >> >> > >

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Re: [Kamailio-Users] preparing kamailio 1.5.2
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello, this one should help you: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn Cheers, Daniel On 07/13/2009 01:30 PM, Marco Bungalski wrote: > Hello, > > i am sorry to say that i've not testet it because i don't know where to > download the fix and how to implementate that :-( > > But if you give me a link i will test it today (i can simulate the bug) and > will send you a feedback. > > Greetings, > > Marco > > > > > Mit freundlichen Grüßen > > Marco Bungalski > > > 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 > web: www.bungalski.de > > Geschäftsführender Gesellschafter: Marco Bungalski > Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586 > > -----Ursprüngliche Nachricht----- > Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Gesendet: Montag, 13. Juli 2009 12:32 > An: Marco Bungalski > Cc: users(a)lists.kamailio.org > Betreff: Re: [Kamailio-Users] preparing kamailio 1.5.2 > > Hello > > On 07/13/2009 12:28 PM, Marco Bungalski wrote: > >> Hello, Daniel! >> >> >> >> Will the bug "Kamailio crashes when receiving an empty >> Record-Route-field with using the dialog-module" be fixed in 1.5.2? >> >> >> >> If yes: Thanks for that :-) >> >> > it should be already in the svn (therefore yes). Have you tested it? > > Cheers, > Daniel > > -- Daniel-Constantin Mierla http://www.asipto.com/

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Re: [Kamailio-Users] preparing kamailio 1.5.2
by Marco Bungalski 13 Jul '09

13 Jul '09

Hello, Daniel! Will the bug "Kamailio crashes when receiving an empty Record-Route-field with using the dialog-module" be fixed in 1.5.2? If yes: Thanks for that :-) Greetings, Marco

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[Kamailio-Users] roadmap to kamailio 3.0
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello, the roadmap to next major release of Kamailio, codenamed 3.0, was sketched during last week IRC meeting as: - freezing in 1-1.5 months - 1-1.5 months testing - release by end of September/ beginning of October Right now, current development state is: - all but one module (seas) were updated to work with the new core - several addons in kamailio core and tm still to be included in GIT master branch - many new features developed: mecached, event_route, xml doc handling ... see: http://sip-router.org/wiki/features/new-in-devel I invite all of you to test, startup guidelines at: http://sip-router.org/wiki/migration/kamailio-3.0-config Any feedback is very much appreciated, thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] How to know if a UAC is behind NAT after fix_nated_register() ?
by Pascal Maugeri 13 Jul '09

13 Jul '09

Hi A quick question, when I apply a fix_nated_register() to a remote user when he/she registers, how can I know later on if this user is behind NAT ? All my calls go through a B2BUA: Caller ------> kamailio/rtpproxy ------> B2BUA ------> kamailio/rtpproxy ------> Callee When my B2BUA calls a user, I want to know if this user is behind NAT or not. For now all the calls go through rtpproxy but I would like to apply this path only when user is behind NAT. Regards, Pascal

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[Kamailio-Users] preparing kamailio 1.5.2
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello, last week during the IRC meeting it was proposed to release kamailio 1.5.2 tomorrow. If anyone of you is aware of some show stopper, please reply. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] two more modules
by Daniel-Constantin Mierla 13 Jul '09

13 Jul '09

Hello, two of the last three modules that were not ported to use the new sip router core are ready now: nat_traversal and siptrace. Particularly, siptrace was refurbished, should catch now forwarded ACK. Before release, there are couple of new features that should get in trunk since sip route core offers support for such extensions: - ability to filter the ACK for negative replies issued by sl module in the script - ability to store stateless forwarded messages (now only stateful ones being stored) Testing is very much appreciated. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] Kamailio 1.4.1 Configuration Issue
by dyerom AD 13 Jul '09

13 Jul '09

Good day everyone. As I am new to kamailio, I would like to ask help on the CFG file. I have successfully installed kamailio 1.4.1 but having problems to set it up on my desired functionality. Below is what i want to achieve. All sip clients (asterisk box, sip enabled box, pbx) should pass sip calls to my kamailio server. Clients connections to my Kamailio are via IP to IP. From my kamailio server i should be able to route them on my carrier, USA carrier and International Carrier. Connections to carrier is via IP to IP. I have set up Kamailio to support MYSQL authentication, its just that i have having problems to put my idea on the kamailio.cfg. Any sample working configurations similar to want I want to achieve is greatly appreciated. Thank you in advance for any support. Dennis Cristobal. jtom25(a)yahoo.com Feel safer online. Upgrade to the new, safer Internet Explorer 8 optimized for Yahoo! to put your mind at peace. It's free. Get IE8 here! http://downloads.yahoo.com/sg/internetexplorer/

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[Kamailio-Users] TCP supervisor process in Kamailio
by Pascal Maugeri 10 Jul '09

10 Jul '09

Hi I recently read the following in order to optimize OpenSER in handling TCP connections: "First, the TCP supervisor process must be given an elevated priority level in order to prevent anomalous behavior due to the Linux scheduler." First of all, as this is quite old paper (it refers to OpenSER 1.2), I'm wondering if such a tuning is still needed for Kamailio 1.5 branch ? If yes, how can I do this ? Regards, Pascal

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[Kamailio-Users] help for multi-domain
by zhangchao00001 09 Jul '09

09 Jul '09

Dear sir: I write to ask if the kamailio support multi-domain. for example, A UA belong to domain b get into domain a, could it work as well as in domain b? If so, how to configure it? Thank you.

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force_rtp_proxy() vis-a-vis BYE
by Alex Balashov 08 Jul '09

08 Jul '09

According to the Kamailio 1.5.x docs, force_rtp_proxy() is deprecated and rtpproxy_offer() and rtpproxy_answer() should be used instead. Does that mean unforce_rtp_proxy() is also deprecated? If so, how should calls be cleaned up once a BYE is received? Is the orthodox approach to simply let the streams time out on their own, or should unforce_rtp_proxy() still be used here? What about when a CANCEL is received while the dialog is an early state -- should unforce_rtp_proxy() be used there too? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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[Serusers] SER generating Via: with branch parameter equal to "0" over half the time
by Frank Durda IV 08 Jul '09

08 Jul '09

A telco equipment vendor has pointed out that when an ACK message pass via our SER servers, a Via: header is added which is normal, but frequently the branch value in that Via: header is "0". As in: Via: SIP/2.0/UDP 10.133.90.x;branch=0 This violates RFC 3261 for being zero and for not being "unique across time and space". (8.1.1.7, page 38). This is showing up in over 50% of the ACK messages being passed through in just one direction. When not equal to zero, the value is like this: Via: SIP/2.0/UDP 10.133.90.x;branch=z9hG4bK095a.b27ac307.0 We don't see branch=0 appearing in any other SIP message type, just ACKs. Is anybody else seeing this and perhaps have a fix? The telco equipment vendor doesn't think it is confusing their equipment, but as it is out spec they want to eliminate it as a source of trouble. The calls are originating in a variety of types of equipment including Acme, Sonus, Asterisk, etc. This is the Via: added by SER, any already present when we got the call are non-zero and look fine. We are running ser-2.0.0-rc1. A sample: ACK sip:6x0x9x0x0x;npdi=yes;rn=6x0x6x9x7x@10.133.0.x:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1x3.9x.1x;branch=0 Via: SIP/2.0/UDP 7x.1x.1x3.1x0:5060;branch=z9hG4bK02Bd8a36ba5a45d76d6 From: <sip:Restricted@7x.1x.1x3.1x0>;tag=gK020a4ae7 To: <sip:6103900202@2x8.3x.2x1.7x>;tag=000a0285+1+11540020+75d5539b Call-ID: 2097284613_75974428(a)7x.1x.1x3.1x0 CSeq: 15498 ACK Max-Forwards: 16 Content-Length: 0 Thanks in advance.

2 3

[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 07 Jul '09

07 Jul '09

Hello, within K project, we had from time to time an IRC meeting for short term planning. IMO it is the time for next one since probably we need at least a kamailio 1.5.2 release. The place was the irc channel #kamailio on freenode.net, now we can host it on #sip-router. Presence is not mandatory and anyone is welcome to join. From the perspective of K project, here are the topics I would like to approach: - kamailio 1.5.2 - show-stoppers, when? - shall we do another branch 1.4-based release - kamailio 3.0 - current status - what is missing, what can be worked-around, prioritizing tasks... - documentation - migration tutorials - release roadmap For sip router: - logistics: anything missing for project now? - what can be done to make adoption easier (new tools, etc...) - suggestions for future development I propose Tuesday, July 7, 2009, we can adjust to fit the best for the people that want to attend. I started a wiki page, feel free to add there: https://sip-router.org/wiki/devel/irc-meetings/2009-07-07 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] SEMS 1.1.1 released
by Stefan Sayer 07 Jul '09

07 Jul '09

(apologies if you receive multiple copies of this mail) Hello, the 111 version of SEMS, the SIP Express Media Server, is now available for download at http://ftp.iptel.org/pub/sems/1.1/1.1.1/ http://ftp.iptel.org/pub/sems/1.1/1.1.1/src/sems-1.1.1.tar.gz 4a6422d09ddadaf9eacd8cae8f0848d5 sems-1.1.1.tar.gz Debian packages for lenny64 are available in the repository: deb http://ftp.iptel.org/pub/sems/debian lenny free deb-src http://ftp.iptel.org/pub/sems/debian lenny free and etch64 in http://ftp.iptel.org/pub/sems/1.1/1.1.1/packages . This is a bugfix release in the 1.1 branch which accumulates fixes for bugs found in 1.1.0 so far. Specifically, this is - fixed Via HF missing the port number in ACK to 200 reply - do not try to scale too short RTP packets - fixed initialization of SSL - caused random crashing of xmlrpc server - fix size() for AmArg struct type - authenticate on both 401 and 407 reply in click2dial - fixed ssl build dependency for DIAMETER client in deb Many Thanks to everyone who contributed with bug reporting and fixes. More information, documentation etc about SEMS can be found at its homepage: http://iptel.org/sems Best Regards Stefan Sayer -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com IPTEGO GmbH Wittenbergplatz 1 10789 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

1 0

[Kamailio-Users] force_rtp_proxy() vis-a-vis BYE
by Alex Balashov 07 Jul '09

07 Jul '09

According to the 1.5.x docs, force_rtp_proxy() is deprecated and rtpproxy_offer() and rtpproxy_answer() should be used instead. Does that mean unforce_rtp_proxy() is also deprecated? If so, how should calls be cleaned up once a BYE is received? Is the orthodox approach to simply let the streams time out on their own, or should unforce_rtp_proxy() still be used here? What about when a CANCEL is received while the dialog is an early state -- should unforce_rtp_proxy() be used there too? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

3 2

Images in the Document "SIP Introduction"
by Antônio Theóphilo 07 Jul '09

07 Jul '09

Hello Everybody Does anyone know how to get the images that are broken in the document "SIP Introduction" ( http://sip-router.org/docbook/sip-router/branch/master/sip/sip_introduction… )?It seems that they are diagrams very important for the understanding of the protocol's dynamics. Antônio Theóphilo

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[Kamailio-Users] SIP-Router Project - how does this affect me?
by Geoffrey Mina 07 Jul '09

07 Jul '09

I am not clear on what exactly the SIP-Router project is, nor how it will affect me? Is Kamailio eventually being replaced with SIP-Router once all the functionalities have been rolled in? Will my entire configuration script need to be reworked to eventually operate with this new platform? Can anyone explain to me why this merge is happening so soon after Kamailio split with OpenSIPS? All this turmoil feels very destructive to me.

3 2

[Kamailio-Users] How should I change Calling Party Number for a call?
by Uriel Rozenbaum 07 Jul '09

07 Jul '09

Hi Guys, I was wondering how can I implement a proxy with Kamailio that can manage privacy and normalization of the calling party. In an INVITE message I can get the calling party name (or number) in several places: 1. From Header 2. Remote-Party ID 3. P-Asserted-Identity 4. P-Preferred-Identity How should I act in order to normalize or block the Calling Party to be passed to subsequent hops? AFAIK I can change the From Header as long as the tag isn't modified. Is this really true? What about RPID or PAI? Should I Change the aliases there as well? Thanks in advance to all! Uriel

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[Kamailio-Users] strange fr_inv_timer behaviour
by hans＠onsmail.nl 07 Jul '09

07 Jul '09

Hi, When setting fr_inv_timer to 60s and the phone is not answered kamailio generates a timeout after 20s ? If fr_inv_timer is set to 90s the timeout is generated after 30s ? Doing a level 7 debug I received the following tm timer related output Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1755]: DBG:tm:set_timer: relative timeout is 500000 Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1755]: DBG:tm:insert_timer_unsafe: [4]: 0xafa6abc4 (16900000) Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1755]: DBG:tm:set_timer: relative timeout is 3 Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1755]: DBG:tm:insert_timer_unsafe: [0]: 0xafa6abe0 (51) Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1754]: DBG:tm:set_timer: relative timeout is 60 Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1754]: DBG:tm:insert_timer_unsafe: [1]: 0xafa6abe0 (108) Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1751]: DBG:tm:set_timer: relative timeout is 60 Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1751]: DBG:tm:insert_timer_unsafe: [1]: 0xafa6abe0 (108) Jun 3 15:07:12 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:utimer_routine: timer routine:4,tl=0xafa6abc4 next=(nil), timeout=16900000 <phone is ringing ....> Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:timer_routine: timer routine:1,tl=0xafa6abe0 next=(nil), timeout=108 Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:final_response_handler: stop retr. and send CANCEL (0xafa6aa78) Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:t_should_relay_response: T_code=180, new_code=408 Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:t_pick_branch: picked branch 0, code 408 Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:t_check: start=0xafa6aa78 Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:t_check: transaction already found! Jun 3 15:07:32 sip-proxy-01 /usr/local/sbin/kamailio[1758]: DBG:tm:pv_get_tm_reply_code: reply code is <408> version: kamailio 1.5.1-notls (i386/linux) flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:5868M @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 14:40:00 Jun 3 2009 with gcc 4.1.2 Kind regards Hans

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[Kamailio-Users] Polycom BLF presence
by Daniel Corbe 07 Jul '09

07 Jul '09

Anyone have any hints on how to make Polycom BLF presence working with Kamailio? -Daniel

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[Kamailio-Users] RtpProxy: start_recording()
by Iñaki Baz Castillo 06 Jul '09

06 Jul '09

Hi, I'm using start_recording() function of nathelper module. I use "-R" option in rtpproxy daemon in order not to record RTCP. start_recording() is called in route and onreply_route. After the call ends I see the following files in the recording directory: 283K 2009-07-06 13:41 1e54fbae418504a40744ad765718792a(a)212.230.19.1=38a3b929-co1782-INS001;1.a.rtp 275K 2009-07-06 13:41 1e54fbae418504a40744ad765718792a(a)212.230.19.1=38a3b929-co1782-INS001;1.o.rtp 0 2009-07-06 13:41 1e54fbae418504a40744ad765718792a(a)212.230.19.1=38a3b929-co1782-INS001;2.a.rtp 0 2009-07-06 13:41 1e54fbae418504a40744ad765718792a(a)212.230.19.1=38a3b929-co1782-INS001;2.o.rtp Some questions: - "1.a.rtp" is one direction and "1.o.rtp" the other, right? - What are the last two files (0 size)? - How to extract the audio from these files? note that the used codec could be G729, ALAW, GSM... Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [SR-Users] [sr-dev] LCR/CR and prefix mask (REGEX?) [moved to SR-Users]
by Edson - Lists 06 Jul '09

06 Jul '09

I looked around... CR uses the same approach that LCR on prefix definitions (or, at least the README don't say/explain that it's different). So I went to look on REGEX module... Iñaki, this module has Your signature... let me ask You what exactly are this REGEX groups? Could I use something like: - Use RURI and recover associated "REGEX-group number"; - Ask LCR or CR to look for routes to prefix="REGEX-group number"; - proceed routing fail-over as usual... Edson Juha Heinanen escreveu: > Iñaki Baz Castillo writes: > > > I'd really like to see some mechanism to make routing decisions based > > on RURI username itself, rather than just no its prefix. Sure it > > could be a good feature. > > lcr module works on longest prefixes of r-uri username (= telephone > number). it was tried once operate on regular expressions, but it was > not a good idea. if you want something else than longest prefixes, > check if some other module (like drouting or cr) could provide that. > > -- juha > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >

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Re: [Kamailio-Users] Problem upgrading from 1.4.x to 1.5.1 notls
by Uriel Rozenbaum 06 Jul '09

06 Jul '09

Henning, how can I configure Kamailio to use some other path to find modules? i should make modules differently as well, right? I'll copy back the list so anyone can answer that. Rgds, Uriel On Wed, Jul 1, 2009 at 9:35 AM, Henning Westerholt < henning.westerholt(a)1und1.de> wrote: > On Mittwoch, 1. Juli 2009, you wrote: > > It's a CentOS 5.3 i386. > > > > # gcc --version > > gcc (GCC) 4.1.2 20071124 (Red Hat 4.1.2-42) > > Copyright (C) 2006 Free Software Foundation, Inc. > > This is free software; see the source for copying conditions. There is NO > > warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR > PURPOSE. > > > > # uname -a > > Linux sbci1pri.cpsnet.com.ar 2.6.18-92.1.18.el5 #1 SMP Wed Nov 12 > 09:30:27 > > EST 2008 i686 i686 i386 GNU/Linux > > > Hi Uriel, > > > thanks, but this is all fairly common, so i doubt your environment is > guilty here. > > > > Do you have some guidelines to erase all Kamailio files from the system? > > I'm sure its a build problem. > > > Sorry, this depends where you installed the server, you basically need to > track all files that were installed. I missed the "make uninstall" also a > few times. :/ > > > But you could just try to install the new version in another directory, > like /tmp/kamailio or something like this. If you then configure the server > to only use modules from this dir it should be not conflict with the old > installation. > > > Henning >

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[Serusers] rtpproxy: How to force it to send packets out a specific interface?
by Frank Durda IV 06 Jul '09

06 Jul '09

I am running SER/rtpproxy on some systems with multiple cores and multiple ethernet interfaces. I am running multiple copies of rtpproxy, each with an IP address associated with a particular interface, as in one rtpproxy is using em1 which is 208.33.44.3 and em2 is 208.33.44.4 which another rtpproxy listens at, while ser listens in em0 at 208.33.44.2. These three (actually ten) interfaces pass through ethernet switches and then on to a router with a default gateway of 208.33.44.1 and then off to the Internet or wherever. (In my setup, each rtpproxy has a second interface pointing into a private network, so only one side of a call appears on any given interface and five face toward the Internet.) SER+rtpproxy correctly divides RTP traffic for calls to 208.33.44.3 and 208.33.44.4 in the SDP payloads, and callers send their RTP packets to the two rtproxies IP addresses without problems. That part works just fine. The problem is that rtpproxy is using the default route for transmission, so everything that goes towards the Internet goes out via em0, so em1 and em2 don't transmit anything. This means that this one link is overwhelmed by the traffic that should have been divided across multiple links. Rtpproxy is allowing the local route table and the default route to handle the outgoing packets and so the packets are all going out the same interface. In my case, em0 was 100Mbit/sec (completely sufficient for SIP messages, while em1 and em2 were gigabit links to handle RTP, but they are getting used only in one direction. Even if em0 was upgraded to Gbit, it would still not be able to handle the potential for 2+ Gbit of data from the rtproxies. So, is there any way to force rtproxy to perform what I believe they used to call "reflector mode", where the answers go out the same interface they come in on? (This used to be common on UDP-based NFS file servers.) A number of communication programs also allow this behavior as an option, but rtpproxy doesn't appear to do it or it isn't turned on by default. Because the destinations for the RTP are out on the Internet somewhere, I don't believe I can solve this via "route" or "ifconfig" settings, but would be delighted if someone knows different or knows of a setting change in rtpproxy that will do this so that load balancing will work. The OS in question is FreeBSD 7.2, but I have the same issue on 6.2 and 6.3. Thanks in advance!

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Re: [Kamailio-Users] SIP-Router Project - how does this affect me?
by Alex Balashov 06 Jul '09

06 Jul '09

That said, this page does speak to a number of the concerns, even if somewhat vaguely: http://sip-router.org/answers/ > >> I am not clear on what exactly the SIP-Router project is, nor how it >> will affect me? Is Kamailio eventually being replaced with SIP-Router >> once all the functionalities have been rolled in? Will my entire >> configuration script need to be reworked to eventually operate with >> this new platform? Can anyone explain to me why this merge is >> happening so soon after Kamailio split with OpenSIPS? All this >> turmoil feels very destructive to me. > > This is a confusing topic, and could definitely use additional discussion > and official clarification. > > I have been made to understand that Kamailio will not be replaced by > sip-router. It is rather that sip-router is a a "common core" that > integrates both Kamailio and sip-router modules, and that both projects > will share this common core going forward, but remain distinct. The > analogy provided to me by Daniel was of Linux kernels vs. distributions. > > This seems fair enough, but it doesn't adequately explain who sip-router > is intended for and who Kamailio is intended for, and who should use which > and why and under what circumstances. It does also seem that most of the > development and documentation energy is being concentrated into sip-router > integration, which leaves open the question of what the future of Kamailio > is. > > This definitely needs to be discussed more. Thank you very much for > raising it, Geoffrey. > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (678) 237-1775 > > > > _______________________________________________ > 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 > -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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[Kamailio-Users] Branch Flag / Location Lookup
by Brandon Armstead 04 Jul '09

04 Jul '09

Hey guys, First off thanks for any help in advanced! I'm saving a nat branch flag into usrloc when the user registers, now each user has two registrations to the same username in usrloc, upon lookup("location"), inside of branch_route[] it appears only one of them have the nat branch flag set, while the other does not? Is there any case where a branch flag could somehow get mangled, or what to look for, or what may cause this? Also one other thing from a database outlook (the cflag) is equivalent on both AOR's. However again at the time of branch_route, one flag is i.e. 000800, and the other 000820, any thoughts are appreciated, thanks!

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Re: [Kamailio-Users] [LCR] About ping
by Iñaki Baz Castillo 04 Jul '09

04 Jul '09

2009/3/23 Alexandr Dubovikov <shurik(a)start4.info>: > On Mon, Mar 23, 2009 at 11:15:52AM +0100, Iñaki Baz Castillo wrote: >> 2009/4/1 Alexandr Dubovikov <shurik(a)start4.info>: >> > On Fri, Mar 20, 2009 at 10:21:20PM +0100, Andreas Heise wrote: >> >> Hello Iñaki, > > Hi Inaki, > >> > >> > Hi all, >> >> >> >> you should ask Alexandr he has introdused this feature with rev5452, >> >> but I'm not sure if he is on the lists all the time, so I'll forward your >> >> question >> >> to him... >> > >> > sorry, I am currently offline and couldn't answer directly to the list. >> > >> > anyway. I agreed, 180 seconds it's too big ping interval, but it couldn't be >> > also less 32 seconds. >> > >> > >> > http://www.ietf.org/rfc/rfc3261.txt >> > >> > 17.2.2 Non-INVITE Server Transaction >> > >> > >> > T1 = 500 ms. >> > >> > >> > Timer J 64*T1 for UDP Section 17.2.2 Wait time for >> > 0s for TCP/SCTP non-INVITE request >> > >> > so, the timer for "completed" is 32 seconds and couldn't be less. >> > >> > >> > of course, you can use less value, but on own risk :-) >> > RFC 3261 are not recommendet it :) >> >> Thanks for pointing it out. >> However, if a gw takes so long time (~32 seconds) to respond I prefer >> to consider it as "offline" :) > > don't forget to disable ping for this gateway, because finaly you will have > a DDOS effect of the "offline" gw. :-) Well, If the host is already offline then it doesn't matter to kill it more XD Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Problem in adding new user in database
by ajay_ei 03 Jul '09

03 Jul '09

Hi, I am a new user of openser. I have installed and configured openser on my local system. when i am trying to add a new user in database i am getting following error: [root@Ajay-PC sbin]# openserctl add ajay ajay ajay@localhost database engine 'MYSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 check_db_alias: alias counter=0 ERROR 1054 (42S22) at line 1: Unknown column 'phplib_id' in 'field list' ERROR: introducing the new user 'ajay' to the database failed Kindly help me. Thanking you. -- View this message in context: http://www.nabble.com/Problem-in-adding-new-user-in-database-tp24307793p243… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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sr-users July 2009