- sr-users - kamailio.org

Presence on Cisco/Linksys handsets
by Mark Boyce 29 Nov '12

29 Nov '12

Hi All, I've been trying to get some Cisco/Linksys SPA phones working with Kamailio (current stable). All seems to be ok apart from getting Presence/BLF/SCA working. The phones are set to Server_Type : RFC3265_4235 Line Key 2 Extension : Disabled Line Key 2 Share Call Appearance : Private Line Key 2 Extended Function : fnc=blf+sd+cp;sub=1001@ourserver.com;ext=1001@ourserver.com Where 1001 is the other phone. The phone appears to be sending event type 'call-info' rather than dialog etc. Phone is being sent '489 Bad Event' and the error in the logs is : presence [subscribe.c:1007]: Unsupported event header field value call-info Whatever I set the Server Type to on the phones it still tries to send call-info events on subscribes Has anyone got Presence working with Cisco SPA handset? The module needed seems to be available on OpenSIPs http://www.opensips.org/html/docs/modules/1.8.x/presence_callinfo.html Thanks! Mark

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Kamailio getting started
by Scott, Matt 29 Nov '12

29 Nov '12

Anybody got a good tutorial or starter project for Kamailio? We have Siremis 3 installed, but there is 0 help on that? Trying to solve 2 scenarios: -asterisk to softphone through kamailio -softphone to asterisk through kamailio. I ultimately want to use Kamailio as an edge router Any help is appreciated. Matt Scott

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Kamailio HA with Virtual IP and multiple Asterisk servers for dispatching
by Rumen Mihailov 29 Nov '12

29 Nov '12

Hi all, Please bear with me as this is my first post to a mailing list ever so something might be quite wrong. My question is: How do we make Kamailio HA ? For the moment I’ve setup MySQL cluster with MySQL proxies that provide 1 Virtual IP that I use to connect Kamailio to (for usrloc and dispatcher) I have a couple of asterisk servers that Kamailio servers are supposed to send calls to. I have two kamailio servers that share the same IP address with keepalived. Both Kamailio servers are bind to the Virtual IP. The problem is: When Kamailio 1 is active calls are correctly dispatched to both the asterisk servers correctly. When Kamailio 2 is active calls are only sent to one of the asterisk servers. ‘kamctl dispatcher show’ shows the correct servers on both machines. The two Kamailio servers share EXACTLY the same config files. Do you guys have any thoughts on this ? p.s. Thank you for creating such a wonderful product. Thank you !!! Best regards, Rumen

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Call from SipML5 -> PSTN
by Raj Roy Ghandhi 28 Nov '12

28 Nov '12

Hi, >From all of your support now I can call from 1. IP Phone <--> IP Phone 2. Web Page <--> Web Page 3. IP Phone -> PSTN without any issue But when I try to call from Web Page to PSTN then it tries to call sip:00xxxxxx89078@mysipdomain.com and that time out. Trying to figure out how to get this work ? Can anybody guide me on this please. Best Regards, Roy.

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Re: [SR-Users] CRLF from Kamailio server
by Klaus Darilion 28 Nov '12

28 Nov '12

forwarded to mailing-list .... -------- Original Message -------- Subject: Re: [SR-Users] CRLF from Kamailio server Date: Mon, 26 Nov 2012 16:34:26 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: miconda(a)gmail.com To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Hello, I haven't checked actually the parts for tcp, but recently I added the feature of removing the location records if there is no reply to OPTIONS keepalives in nathelper, and it was only for UDP. Also, I think the nat ping is 4 bytes (zeros) or so. Might be another option from tcp, like: http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tcp_keepalive Cheers, Daniel On 11/26/12 3:54 PM, Klaus Darilion wrote: > IIRC tcp_crlf_ping is only about the Ping-Pong mechanism: client sends > PING=CRLF, server responds with PONG=CRLFCRLF. > > Thus, server should only response on client CRLF. Further the PONG is > always double-CRLF. > > So I suspected it is NAT keepalive - maybe someone added > NAT-keep-alive for TCP :-) > > regards > Klaus > > On 26.11.2012 11:05, Daniel-Constantin Mierla wrote: >> Hello, >> >> this might be actually related to tcp connection keepalive: >> >> http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tcp_crlf_ping >> >> IIRC, nathelper pings only udp contacts. >> >> Cheers, >> Daniel >> >> On 11/26/12 10:54 AM, Klaus Darilion wrote: >>> >>> >>> On 26.11.2012 05:13, Dmytro Bogovych wrote: >>>> Greetings. >>>> May anyone give me advice? >>>> I have working Kamailio with TLS. >>>> It is used by iOS softphone - i developed own one based on >>>> resiprocate. >>>> All signalling is TLS. >>>> Almost everything is fine. >>>> >>>> But looks kamailio sends CRLF messages sometimes. >>>> If it happens when softphone is in background mode - it is killed >>>> by iOS >>>> (as iOS prohibits too often awakes from background mode). >>>> >>>> How I can disable these messages? >>>> I see natping_interval = 30 and ping_nated_only=0 >>>> Should I set natping_interval to zero ? >>> >>> Either set the interval to 0 or set ping_nated_only=1 and make sure to >>> not set the NAT flag for the iphone clients. >>> >>> A problem may be firewalls/NAT between the iPhone and Kamailio, as >>> they may drop the TCP connection is there is no activity for too long >>> time. >>> >>> regards >>> Klaus >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] Call from SipML5 -> PSTN
by Peter Dunkley 28 Nov '12

28 Nov '12

You need some sort of media gateway or server. I don't know of any that currently support this. Peter Raj Roy Ghandhi <roy.gandhi(a)gmail.com> wrote: >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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memcached loading problem
by Alex Balashov 28 Nov '12

28 Nov '12

Hello, So, I built the memcached module against the latest Git production branch of memcached and libmemcache-1.4.0.rc2 (the latest available), on Debian i386. But when I try to load the Kamailio memcached module: Nov 28 03:47:44 diminuendo-1 kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/memcached.so>: /usr/lib/libmemcache.so.0: undefined symbol: mcm_buf_len I find that very odd, because the symbol is quite clearly exported in the library: diminuendo-1:~# nm /usr/lib/libmemcache.so.0 | grep -i mcm_buf_len U mcm_buf_len Does anyone have any idea what the underlying linking issue is here? -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Websocket with Kamailio 3.2.2 release
by Raj Roy Ghandhi 28 Nov '12

28 Nov '12

Hi, I am trying to integrate the websocket module into release 3.2.2. But I am unable to do that. Please guide me to get it done. Regards, Roy.

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Eavesdropping SRTP sessions
by Mino Haluz 28 Nov '12

28 Nov '12

Hi, maybe it is not that kamailio related question, but I dont know any other place with such good voip professionals ;) I have kamailio and mediaproxy. Clients are BudgetTone 200 (Grandstream) and CSipSimple. I am forcing clients to use SRTP but it does not support adding any certificate on both sides. SRTP call is working fine. The question is, in this case, is man-in-the-middle attack possible? Maybe I should study SRTP more, but basically, if there are no certificates, there is no method how to be 100% sure that the media goes directly between clients. Is it true? Thanks for response, Mino

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Managing expiration values with memcached
by Alex Balashov 28 Nov '12

28 Nov '12

According to the memcached module docs, setting $mctex(key) can be used to modify the expiration value of a key. Should this be done after or before setting the key itself with $mct(...)? I have tried it before and the assignment didn't work, so clearly the key must first exist in memcache: $mctex(bgn=>$avp(s:bgn)) = $sel(cfg_get.bgn_expiry); $mct(bgn=>$avp(s:bgn)) = $dbr(ra=>[0,0]); Nov 28 05:23:41 diminuendo-1 /usr/local/sbin/kamailio[32089]: ERROR: <core> [lvalue.c:354]: setting pvar failed Nov 28 05:23:41 diminuendo-1 /usr/local/sbin/kamailio[32089]: ERROR: <core> [lvalue.c:407]: assignment failed at pos: (164,37-164,69) OK, so that makes sense. I tried it the other way: $mct(bgn=>$avp(s:bgn)) = $dbr(ra=>[0,0]); $mctex(bgn=>$avp(s:bgn)) = $sel(cfg_get.bgn_expiry); But when I look on my memcached server, the expiration value is not being modified. It's still the default 'expire' value set in the memcached modparam. I ran the memcached server with maximum verbosity and verified that no TOUCH command is taking place, and that the SET command for the key contains the default expiration value, not the one I set with $mctex(...). So, it seems to me that the module is not committing this new expiration value when it is written to $mctex(...). Is this a bug, or am I missing something? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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problem with kamailio and libradiusclient-ng/freeradius-xs
by Rabary 27 Nov '12

27 Nov '12

Hi all, When I have this line in kamailio.cfg: modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") Kamailio failed to start: kamailio error, failed to start (/var/run/kamailio/kamailio.pid exists). I already try to stop kamailio, an remove this file but don't work In log,I have: Nov 27 20:13:39 voip-core-1 kernel: [14318.988658] kamailio[8370]: segfault at 0 ip 00007f0b0b94ca5d sp 00007fffa08f9460 error 4 in libradiusclient-ng.so.2.0.0[7f0b0b946000+c000] Nov 27 20:13:39 voip-core-1 kamailio: ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe When I remove just this line and restart kamailio, it works, I don't know what is the problem because before I can do accounting with freeradius with the same config I have kamailio 3.3.2, kamailio-radius-modules 3.3.2,libradiusclient-ng2 0.5.6-1.1 and on 2nd server freeradius-xs and freeradius-xs-mysql 1.1.7-4 I put in attached file a few config of kamailio and radiusclient Regards -- Rabary Teddy Inutile d'imprimer ce mail

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Disconnection from Kamailio
by Dmytro Bogovych 27 Nov '12

27 Nov '12

Greetings. I observe strange behavior with Kamailio. My iphone 3gs run softphone based on resiprocate. First 10-20 minutes everything works fine. After it the connection is closed from server (select() loop returns read notification, read attempt returns zero bytes). Ok, softphone closes old socket and create new one by new registration. But this connection is terminated from server in 4-5 seconds after registration finish (200 OK). Softphone makes next attempt to register - connection is closed after registration again. So it is infinite loop. After several attempts iOS kills application as it awakes too often in background mode. I have not logs from server yet; but maybe anyone had similar issue? Thank you :)

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send 100 provisioning delayed
by pars3c 27 Nov '12

27 Nov '12

Hi , The proxy send the "automatic" provisioning message 100 "Trying" , near 1-2 seconds later the INVITE message, so the client send 3 times the same invite (correct , 500 ms timer). Is there any parameter to set , for the time between the INVITE and "100" message ? Is correct to disable the automatic send, and set after the invite, sl_send_reply("100", "Trying"); ? Thanks

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Kamailio HA with Virtual IP and multiple Asterisk servers for dispatching
by Rumen Mihailov 27 Nov '12

27 Nov '12

Hi all, Please bear with me as this is my first post to a mailing list ever so something might be quite wrong. My question is: How do we make Kamailio HA ? For the moment I've setup MySQL cluster with MySQL proxies that provide 1 Virtual IP that I use to connect Kamailio to (for usrloc and dispatcher) I have a couple of asterisk servers that Kamailio servers are supposed to send calls to. I have two kamailio servers that share the same IP address with keepalived. Both Kamailio servers are bind to the Virtual IP. The problem is: When Kamailio 1 is active calls are correctly dispatched to both the asterisk servers correctly. When Kamailio 2 is active calls are only sent to one of the asterisk servers. 'kamctl dispatcher show' shows the correct servers on both machines. The two Kamailio servers share EXACTLY the same config files. Do you guys have any thoughts on this ? p.s. Thank you for creating such a wonderful product. Thank you !!! Best regards, Rumen

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Kamailio crash
by Ricardo Martinez 27 Nov '12

27 Nov '12

Hello In the last three days my kamailio process has crashed two times. I obtained the core file and see it, this is the output. Can someone check what is the problem? Core was generated by `/usr/local/sbin/kamailio -m 1024'. Program terminated with signal 11, Segmentation fault. #0 core_hash (cmd_tree=<value optimized out>, param=<value optimized out>) at ../../lib/kcore/hash_func.h:56 56 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; Missing separate debuginfos, use: debuginfo-install filesystem-2.4.30-2.1.el6.x86_64 glibc-2.12-1.25.el6_1.3.x86_64 keyutils-libs-1.4-1.el6.x86_64 krb5-libs-1.9-9.el6_1.2.x86_64 libcom_err-1.41.12-7.el6.x86_64 libselinux-2.0.94-5.el6.x86_64 mysql-libs-5.1.52-1.el6_0.1.x86_64 nss-softokn-freebl-3.12.7-1.1.el6.x86_64 openssl-1.0.0-20.el6_2.1.x86_64 pcre-7.8-3.1.el6.x86_64 zlib-1.2.3-26.el6.x86_64 (gdb) bt #0 core_hash (cmd_tree=<value optimized out>, param=<value optimized out>) at ../../lib/kcore/hash_func.h:56 #1 process_mi_params (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_hash.c:1057 #2 mi_terminate_dlgs (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_hash.c:1165 #3 0x00007fc789996370 in run_mi_cmd (rx_sock=11, tx_sock=12) at ../../lib/kmi/mi.h:77 #4 mi_datagram_server (rx_sock=11, tx_sock=12) at datagram_fnc.c:525 #5 0x00007fc789998509 in datagram_process (rank=<value optimized out>) at mi_datagram.c:346 #6 0x00007fc78999a09c in mi_child_init (rank=<value optimized out>) at mi_datagram.c:296 #7 0x00000000004dbd61 in ?? () #8 0x00007fc78d99aab8 in ?? () #9 0x00007fc78d0181cd in pkg_proc_update_real_used (data=<value optimized out>) at pkg_stats.c:141 #10 0x000000000052b423 in ?? () #11 0x00007fc78d8a2830 in ?? () #12 0x0000000000000000 in ?? () (gdb) I’m using kamailio [root@pxh ~]# kamailio -V version: kamailio 3.2.3 (x86_64/linux) 2a00c6-dirty with dialog module, hash tables, radius authorization. Hope someone could help me. Regards, Ricardo.-

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plans for next major release
by Daniel-Constantin Mierla 27 Nov '12

27 Nov '12

Hello, following the devel IRC meeting from last week, I am starting the discussion about the next major release planning. There were two relevant options: 1) freeze by end of January (2013) and release by March 2) freeze by end of December (2012) and release by February Many devels announced quite some work ahead, so the 1st option seemed not to be enough for finishing intended work, which may add some extra delays (several estimated to finish by end of February or March). So the 2nd came into debate, to do an early release as many people seemed very interested in getting websockets support out in a stable release. Both of the options fit in our releasing policy (8 to 10 months), first one with potential to delays. We do have now lot of new stuff in devel branch, with other ongoing work planned to be pushed before end of year. A suggestion was to do the following major release a bit shorter, if it proves to collect a lot of additions quickly -- maybe by summer 2013. As not all devels were in the meeting, but also because it impacts the user community, I am looking to get some feedback on the mailing lists as well. That will allow to make the decision quicker based on showed interest. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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One active registration per sip instance.
by Krishna Kurapati 26 Nov '12

26 Nov '12

Hi, I have 3.x Kamailio with GRUU support. I see SIP Instance in the contact. However, I see a problem. When I kill the SIP client on a device and restart, it sends registration on a different port. The server instead of updating the contact in user location table, is creating a new contact for the same instance. Do I need to turn on something to get the expected behavior? Here is the output from kamctl ul show. Thanks Krish Kura AOR:: 100227074 Contact:: sip:100227074@10.132.60.70:50845;transport=TCP;ob Q= Expires:: 3132 Callid:: ATZJcr.HqXxwWdsPmanjERetcdeRVeZy Cseq:: 4399 User-agent:: n/a Received:: sip:166.137.120.37:48873;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b3602b-50e-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50839;transport=TCP;ob Q= Expires:: 3029 Callid:: oYsIYUyP3N77MKSoQ3EDChEph.lD4fjB Cseq:: 37489 User-agent:: n/a Received:: sip:166.137.120.37:61763;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b3602b-507-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50803;transport=TCP;ob Q= Expires:: 1527 Callid:: b4IeJXybn6Yr.jB8sc-Kz8CJ0kt0hNbO Cseq:: 30014 User-agent:: n/a Received:: sip:166.137.120.37:36879;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f5c-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50818;transport=TCP;ob Q= Expires:: 2199 Callid:: DM0Z1d-ZenGHknjmT.7Cle5cos7Uhs7H Cseq:: 22942 User-agent:: n/a Received:: sip:166.137.120.37:39471;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f6f-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50815;transport=TCP;ob Q= Expires:: 1833 Callid:: VJy3SNE66U1U971u5BEeqbYJi6a-puT- Cseq:: 64347 User-agent:: n/a Received:: sip:166.137.120.37:35872;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f5a-2 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1

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new developer: Andrew Mortensen
by Daniel-Constantin Mierla 26 Nov '12

26 Nov '12

Hello, I want to announce that a new developer got GIT write access to repository: Andrew Mortensen from University of Pennsylvania, USA - he has contributed a new module: sca (Shared Call Appearances for Broadsoft extensions). The module is right now in a personal branch, soon to me merged in master branch. His git commit id is: admorten My warm welcome and looking forward to future work within the project. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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ul.dump Problem
by Spencer Thomason 26 Nov '12

26 Nov '12

Hello, I'm attempting to dump the contents of the usrloc cache and I'm receiving the following error: sercmd> ul.dump error: 500 - Internal error creating aor struct sercmd> mi ul_dump error: 500 - Internal server error processing 's': buffer too small (overflow) (-2) It seems to work fine when there are only a small number of registrations (about 20 or so) but then fails when the number increases. It there a setting I need to increase somewhere? Thanks, Spencer

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problem with $time Pseudo-Variables on 3.3.2
by Uri Shacked 26 Nov '12

26 Nov '12

Hi, I upgraded to 3.3.2 On 3.2 i used the variable $time() to get the current time for some process. Every 2 minutes a RTIMER process started, checked the $time(hour) and $time(min) and on a certain time executed something. This process stopped working on 3.3.2 When i debug, i see that the $time() variable keeps on giving me the same hour and minute that was given on the first run. Meaning the $time() variable does not give the current time. I notice also that the $Ts that on 3.2 gave a current time stopped doing it and $TS does it now.... Any help here? Is it OK and i need to check time another way? Or something in the $time() has changed...? Thanks, Uri

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CRLF from Kamailio server
by Dmytro Bogovych 26 Nov '12

26 Nov '12

Greetings. May anyone give me advice? I have working Kamailio with TLS. It is used by iOS softphone - i developed own one based on resiprocate. All signalling is TLS. Almost everything is fine. But looks kamailio sends CRLF messages sometimes. If it happens when softphone is in background mode - it is killed by iOS (as iOS prohibits too often awakes from background mode). How I can disable these messages? I see natping_interval = 30 and ping_nated_only=0 Should I set natping_interval to zero ? Thank you :)

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Re: [SR-Users] Forwarding Relaying of REGISTER Messages
by kellon＠sis-grenada.com 25 Nov '12

25 Nov '12

Hi All Alex responded to my question saying the following: "You can relay SIP messages of any type just the same way as you do with INVITEs -- e.g. t_relay(), forward(). But, proxying registration messages can have complications." can anyone give me some pointers or a list of some of these complications so that I can further research to see how its is done. I will attempt using the t_relay to view the issues, just need some idea more specifically what to look for! Thanks

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Re: [SR-Users] Re-Directing of REGISTER Messages
by kellon＠sis-grenada.com 25 Nov '12

25 Nov '12

Hi Can anyone point to a good example of Kamailio performing the function of Re-direct server. We would like to forward messages onto another domain for authentication.

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Difficulties with with loose_route() when trying to handle non-compliant ACK
by Richard Brady 24 Nov '12

24 Nov '12

Hi guys I have a multihomed Kamailio proxy sitting between two B2BUAs on separate networks and record-routing all dialogs. The problem I have is that when one of these devices receives a 200 OK, it does not populate the RURI of the ACK correctly. Instead of taking it from the Contact header on the 200 OK, it uses the user part from the Contact header and sets the domain to the proxy IP. It then also populates the Route headers. The ACKs are below and the IPs are: 10.152.1.92:5060: UAC on outside 10.200.70.100:5060: proxy outside interface 192.168.242.100: proxy inside interface 192.168.242.102: UAS on inside This the ACK message going into and coming out of the proxy. # U 10.152.1.92:5060 -> 10.200.70.100:5060 ACK sip:natted_ua*9197**192.168.242.102*5080*udp@10.200.70.100 SIP/2.0. Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK-2280-1-5. Route: <sip:10.200.70.100;r2=on;lr=on;nat=yes>,<sip:192.168.242.100;r2=on;lr=on;nat=yes>. From: sipp <sip:sipp@127.0.1.1:5060>;tag=2280SIPpTag001. To: sut <sip:9197@10.200.70.100:5060>;tag=p0FaeQ1QUS9ae. Call-ID: 1-2280(a)127.0.1.1. CSeq: 1 ACK. Contact: sip:sipp@127.0.1.1:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 10.200.70.100:5060 -> 192.168.242.100:5060 ACK sip:192.168.242.100;r2=on;lr=on;nat=yes SIP/2.0. Via: SIP/2.0/UDP 10.200.70.100;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=10.152.1.92;branch=z9hG4bK-2280-1-5. Route: . From: sipp <sip:sipp@127.0.1.1:5060>;tag=2280SIPpTag001. To: sut <sip:9197@10.200.70.100:5060>;tag=p0FaeQ1QUS9ae. Call-ID: 1-2280(a)127.0.1.1. CSeq: 1 ACK. Contact: <sip:sipp@127.0.1.1:5060;alias=10.152.1.92~5060~1>. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. So I am trying to understand why it is trying to relay to itself (I have mhomed=1) and why it is rewriting the RURI as if it is a strict router. Currently decode_contact() is disabled but enabling doesn't seem to help. Unless there is a very specific place where it belongs. Any advice would be hugely appreciated. I can always paste logs / configs / traces. Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

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can not make calls with kamailio 3.3 and asterisk-10.7.0
by James zhu 24 Nov '12

24 Nov '12

hello, guys:i follow the linkhttp://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-a… can registered users, kamctl online shows those status. but i can not make calls. this is debug message when i make a call from 102 to SIP 103--------------------------------------------------------------- 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=486 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=3 n=return 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=489 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=636 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=601 a=25 n=has_totag 8(1286) DEBUG: siputils [checks.c:103]: no totag 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=501 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=26 n=is_method 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=496 a=25 n=t_check_trans 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [tm.c:1015]: lookup_original: t_lookupOriginalT returned: 0xb4d36320 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=497 a=25 n=t_relay 8(1286) DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=29 , global msg id=28 , T on entrance=(nil) 8(1286) DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=60763, isACK=0 8(1286) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(1286) DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found 8(1286) DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb4d3a698, callback type 1, id 0 entered 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [t_fwd.c:1243]: DEBUG: e2e_cancel: e2e cancel proceeding 8(1286) DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.1.103, 192.168.1.103, 0) 8(1286) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 8(1286) DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 8(1286) DEBUG: tm [t_reply.c:703]: DEBUG: reply sent out. buf=0xb721ee38: SIP/2.0 200 cancelin..., shmem=0xb4d38c08: SIP/2.0 200 cancelin 8(1286) DEBUG: tm [t_reply.c:713]: DEBUG: _reply_light: finished 8(1286) DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=3 n=exit 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list=================================anyone can give a hint for that? thanks! Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com

1 0

Using uac_req_send function
by Freddy C 22 Nov '12

22 Nov '12

Hello, I'm using uac_req_send () to send a request generated in Kamailio (eg Example 1.21. UAC module) $ uac_req (method) = "OPTIONS"; $ uac_req (ruri) = "sip: kamailio.org"; $ uac_req (furi) = "sip: kamailio.org"; $ uac_req (turi) = "sip: kamailio.org"; uac_req_send (); However, I need to capture the responses of this request with failure_route TM module. Is there a way to capture these responses when I'm using uac_req_send() function? I appreciate your comments. Freddy

2 1

Next IRC devel meeting
by Daniel-Constantin Mierla 22 Nov '12

22 Nov '12

Hello, I am thinking of having our next IRC devel meeting soon, to plan the next major release and review current stable releases and the environment around the project (e.g., if you can add anything else to the project to make life easier for developers and users). I'm proposing next week, Thursday, at 15:00GMT, on Freenode IRC server, channel #sip-router. Anyone able to join? If it is not a convenient date for you, just propose alternatives and we will select the one that meets the constraints of the most developers and users willing to participate. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 10

How does SIP transfer work?
by Grant Bagdasarian 22 Nov '12

22 Nov '12

Hello, I've been searching the internet to find an explanation on how SIP transfer works using Re-INVITE and/or UPDATE, but I can't seem to find a good source. >From what I understand(and this is the way we do it), the following happens: Bob=Caller Alice=Called John=Transfer party 1) Bob calls Alice. The usual INVITE,Trying,200 OK, ACK. 2) Alice transfers the call to John using Re-INVITE. a. Alice calls John. The usual INVITE,Trying,200 OK, ACK. b. Alice Re-INVITEs Bob using INVITE with adjusted SDP. 3) Bob is connected to John through Alice in some magical way. I'm guessing because the SDP has been changed and for some reason the RTP stream flows between Bob and John through Alice? Is this correct? If not, perhaps someone could explain it to me from scratch. Maybe useful to know that we are using Cisco equipment for call handling (VXML and TCL scripts). Thanks, Grant

3 2

Forward Register Message
by kellon＠sis-grenada.com 21 Nov '12

21 Nov '12

Hi All Can anyone point to documentation tutorial on howto forward statefully a REGISTER message to another SIP Server. I have looked at http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, However I'm getting errors with the uac_req commands even though I have loaded the uac.so module. .

2 1

Shared Call Appearances module
by Andrew Mortensen 21 Nov '12

21 Nov '12

I've been working on a Shared Call Appearances module for the past several months. It implements the Broadsoft SCA feature as laid out in Broadworks SIP Access Side Extensions Interface Specifications document. (Another implementation of the same feature was added to Freeswitch a few years ago, but we didn't want to use Freeswitch.) We've been testing and improving the module over the last month, and I think it's ready to share. The module is available at GitHub here: <https://github.com/fitterhappier/sca> To date, we've only tested the module with Polycom handsets. I'm happy to answer questions. I hope this will prove useful for others. Best, andrew

6 18

cost script
by Ruben Mejia 21 Nov '12

21 Nov '12

IPTEL We needhelp or cost for the development a script for kamailio which can do the following: Register kamailio to another gateway using UAC modules. route all traffic coming towards a private ip 10.x.x.x We will use a postgres database We have installed kamailio 3.1 with postgres support and UAC modules Rubén Mejía Telefonia Cabletamps

1 0

I can not make call from kamailio to asterisk by 3.3
by James zhu 21 Nov '12

21 Nov '12

hello, guys:i follow the linkhttp://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-a… can registered users, kamctl online shows those status. but i can not make calls. this is debug message when i make a call from 102 to SIP 103--------------------------------------------------------------- 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=486 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=3 n=return 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=489 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=636 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=601 a=25 n=has_totag 8(1286) DEBUG: siputils [checks.c:103]: no totag 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=501 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=26 n=is_method 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=496 a=25 n=t_check_trans 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [tm.c:1015]: lookup_original: t_lookupOriginalT returned: 0xb4d36320 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=497 a=25 n=t_relay 8(1286) DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=29 , global msg id=28 , T on entrance=(nil) 8(1286) DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=60763, isACK=0 8(1286) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(1286) DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found 8(1286) DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb4d3a698, callback type 1, id 0 entered 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [t_fwd.c:1243]: DEBUG: e2e_cancel: e2e cancel proceeding 8(1286) DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.1.103, 192.168.1.103, 0) 8(1286) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 8(1286) DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 8(1286) DEBUG: tm [t_reply.c:703]: DEBUG: reply sent out. buf=0xb721ee38: SIP/2.0 200 cancelin..., shmem=0xb4d38c08: SIP/2.0 200 cancelin 8(1286) DEBUG: tm [t_reply.c:713]: DEBUG: _reply_light: finished 8(1286) DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=3 n=exit 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list=================================anyone can give a hint for that? thanks! Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com > From: sr-users-request(a)lists.sip-router.org > Subject: sr-users Digest, Vol 90, Issue 72 > To: sr-users(a)lists.sip-router.org > Date: Wed, 21 Nov 2012 08:58:40 +0100 > > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Shared Call Appearances module (Daniel-Constantin Mierla) > 2. Re: Shared Call Appearances module (Andrew Mortensen) > 3. Re: LCR weight (Juha Heinanen) > 4. problem with $time Pseudo-Variables on 3.3.2 (Uri Shacked) > 5. Re: Dialog module not decrementing profile size > (Grant Bagdasarian) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 20 Nov 2012 22:56:10 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Shared Call Appearances module > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <50ABFC7A.2070602(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > On 11/20/12 10:41 PM, Ovidiu Sas wrote: > > On Tue, Nov 20, 2012 at 4:22 PM, Andrew Mortensen > > <admorten(a)isc.upenn.edu> wrote: > >> On Nov 20, 2012, at 3:16 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > >> > >>> On 11/20/12 8:43 PM, Andrew Mortensen wrote: > >>>> On Nov 20, 2012, at 4:08 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > >>>> > >>>>> Hello, > >>>>> > >>>>> thank for contribution! > >>>>> > >>>>> On 11/20/12 12:16 AM, Andrew Mortensen wrote: > >>>>>> On Nov 19, 2012, at 5:53 PM, David | StyleFlare <david(a)styleflare.com> wrote: > >>>>>> > >>>>>>> Also quick read at the readme, it looks like it does not support multi-domain setups? > >>>>>> Not yet, no. I don't think it would take much work, though. > >>>>> I see it binds to usrloc, what information needs from there? We have two such modules right now, it would be good to make it work with both, kamailio's one has more features in regard to GRUU and partial outbound extensions. > >>>> It's registering for contact expiration and deletion events. The callback terminates subscriptions for the expired/deleted contact. This should probably be configurable. > >>> Ok, so it is about when a contact record in removed from location. > >>> > >>> Just to avoid misunderstandings, do you consider to make configurable this termination of subscriptions based on location? > >> Yes. I can imagine a setup where the location data wouldn't be saved on the host handling SCA. > >> > >>> Also, you need only the AoR and contact address in a callback for location record removal event (un-registration or registration expiration), right? > >> Correct, although Ovidiu makes a good point about (well-behaved) clients unsubscribing at the time of unregistration. In registering for usrloc callbacks, I'm trying to handle a couple problems. First, I'd like to avoid unnecessarily sending and retransmitting NOTIFYs to offline clients. More importantly, I want to be able to hear about expired registrations so I can clear appearance state on the rest of the subscribers. > > For clearing appearance state, you should rely on the dialog module > > and register a callback on the dialog module. When a dialog expires, > > you can clear up the stale appearances. > > Also, by enabling the sst module, you can have a tight control over > > the duration of the dialog. > > This will make the implementation cleaner and more robust. > > > I will not jump to such conclusions. > > While dialog module is designed to track active calls, does not mean > everything has to be on top of it. Just for example, dispatcher module > has an embedded lightweight call tracking system. > > No idea here what the broadsoft extensions require, afaik, there were > some custom specs. It is no problem to have a dedicated dialog tracking > system that is better designed for a particular functionality. We do > have other duplicated systems, for example least cost routing, nat > traversal, a.s.o. > > Also, optimizations and reuse of other parts can be done later if found > to be better ways. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > > > > ------------------------------ > > Message: 2 > Date: Tue, 20 Nov 2012 17:17:38 -0500 > From: Andrew Mortensen <admorten(a)isc.upenn.edu> > Subject: Re: [SR-Users] Shared Call Appearances module > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <65EAE745-8966-4848-870E-6A50246E0AB7(a)isc.upenn.edu> > Content-Type: text/plain; charset=iso-8859-1 > > > On Nov 20, 2012, at 3:43 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: > > > Hello Andrew, > > > > First of all, thank you for sharing your work. > > I was following this thread and I have a couple of questions. Why do > > you need to bind to the usrloc module? The subscription itself should > > be sufficient because if a phone will unregister, it will also > > unsubscribe, > > Yes, that's good point. However, if the phone goes off-line instead of unregistering, the contact's registration will expire, and no unsubscribe will take place. As I mentioned in my reply to Daniel, I'm most concerned about catching this so I can release any appearances seized by the expired/deleted contact, and notify other members of the group. I'm certainly open to alternatives. > > > which leads me to the following question: why not add a > > new dedicated module for call-info presence and reuse the existing > > infrastructure from kamailio for handling subscriptions/presence. In > > this case, your module should just push PUBLISH events to the presence > > server and the server will automatically send out notifications for > > subscribers that subscribed to call-info events. > > We've worked extensively with the existing presence & pua modules, albeit primarily dialog;sla using OpenSIPS behind the proxy. (Thanks, Anca!) > > SCA's unusual entanglement with call processing made me hesitate about building on top of the existing presence module. For a first pass, I also felt working with an entirely distinct module gave me more control and transparency during development of a very loosely documented event package, especially as I became more familiar with the available API. > > I don't have any objection to revisiting design decisions, of course. I'm sure the module will continue to evolve, and it would be nice to eliminate redundancy if possible. > > > Based on your README file, you inspect SIP requests/replies and based > > on the presence of the call-info header, you create call-info events. > > This is great for sharing the appearances between phones, but how do > > you perform the retrieval of a call that was put on hold? Are you > > using a dedicated B2BUA behind kamailio? > > No B2BUAs are involved. > > The INVITE retrieving a call held by another member of an SCA group has a particular set of characteristics: RURI, To and From URIs are all the SCA AoR; new unconfirmed dialog (no to-tag); and a Call-Info header referring to the index of the held call. The module detects this type of INVITE, looks up the dialog associated with the information in the Call-Info header, and injects a Replaces header with the dialog of the held call before relaying it to the remote party. The remote party must support RFC3891. I've only worked with Polycoms and Cisco gateways to this point, and both do support that RFC. > > I'm very interested in hearing from users with Cisco (and Snom and Aastra?) SCA setups. I have no doubt they'll find bugs resulting from assumptions I've made in the code because I've only tested with Polycom, not least the dependency on RFC3891 support. > > Thanks for your feedback! > > Best, > andrew > > > > > On Tue, Nov 20, 2012 at 3:16 PM, Daniel-Constantin Mierla > > <miconda(a)gmail.com> wrote: > >> > >> On 11/20/12 8:43 PM, Andrew Mortensen wrote: > >>> > >>> On Nov 20, 2012, at 4:08 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> > >>> wrote: > >>> > >>>> Hello, > >>>> > >>>> thank for contribution! > >>>> > >>>> On 11/20/12 12:16 AM, Andrew Mortensen wrote: > >>>>> > >>>>> On Nov 19, 2012, at 5:53 PM, David | StyleFlare <david(a)styleflare.com> > >>>>> wrote: > >>>>> > >>>>>> Also quick read at the readme, it looks like it does not support > >>>>>> multi-domain setups? > >>>>> > >>>>> Not yet, no. I don't think it would take much work, though. > >>>> > >>>> I see it binds to usrloc, what information needs from there? We have two > >>>> such modules right now, it would be good to make it work with both, > >>>> kamailio's one has more features in regard to GRUU and partial outbound > >>>> extensions. > >>> > >>> It's registering for contact expiration and deletion events. The callback > >>> terminates subscriptions for the expired/deleted contact. This should > >>> probably be configurable. > >> > >> > >> Ok, so it is about when a contact record in removed from location. > >> > >> Just to avoid misunderstandings, do you consider to make configurable this > >> termination of subscriptions based on location? > >> > >> Also, you need only the AoR and contact address in a callback for location > >> record removal event (un-registration or registration expiration), right? > >> > >> > >>> > >>>> Would you consider adding it to main git repository and maintaining it > >>>> there? > >>> > >>> Yes, that would be great. > >> > >> > >> We will prepare write access for you. > >> > >> > >>> Would you prefer to have it in modules? I started work on it in > >>> modules_s, but there's no particular reason it has to stay there. > >> > >> To make it work with both variants of usrloc modules, I think it requires > >> some split of the code. IIRC, both modules still use same names for > >> structures, so including files from both of them will fail. > >> > >> One solutions I am thinking of is to add two small modules, one that binds > >> to usloc(k) and the other to usrloc(s). Each of these two modules will > >> export a function that allow registering callbacks for execution when a > >> location record expires, giving the aor, contact and event type. The > >> existing module will use the intermediate module instead of directly binding > >> to usrloc structures. > >> > >> Usrloc module from kamailio has more enhancements in regards to standard > >> extensions for location services, still it lacks the feature of being able > >> to store variables per contact record. It is the reason that kept back the > >> removal (upon merging) of usrloc module from ser. The solution with > >> intermediary modules should work until we have a merged usrloc module. > >> > >> What do you think about this proposal? > >> > >> Cheers, > >> Daniel > >> > >> > >> -- > >> Daniel-Constantin Mierla - http://www.asipto.com > >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > >> > > > > > ------------------------------ > > Message: 3 > Date: Wed, 21 Nov 2012 01:04:52 +0200 > From: Juha Heinanen <jh(a)tutpro.com> > Subject: Re: [SR-Users] LCR weight > To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <20652.3220.931000.738348(a)sip.test.fi> > Content-Type: text/plain; charset=us-ascii > > Camila Troncoso writes: > > > scripts are provided in lcr/utils directory that can be used to check > > the probabilities resulting from a given set of weight values. Same can > > be done with command 'kamctl eval_weights'. > > > > I really don't understand what is the probability finally assign to each > > gateway when they have the same priority. > > All I can see is that it doesn't correspond to percentage of capacity from > > the example above , but how can I determine what value gives the percent > > that I want? > > > > Please give an example like: > > > > Rule_id Gw_id Priority weight > > 36 68 1 60 > > 36 69 1 14 > > 36 70 1 13 > > 36 71 1 13 > > you have to figure that out yourself using the script referenced in > readme. > > -- juha > > > > ------------------------------ > > Message: 4 > Date: Wed, 21 Nov 2012 09:55:17 +0200 > From: Uri Shacked <ushacked(a)gmail.com> > Subject: [SR-Users] problem with $time Pseudo-Variables on 3.3.2 > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAMMbDhQn1r_ke72-u2is8MyRXUV6KkLnZqYwxL5eZG44yYG_2Q(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > I upgraded to 3.3.2 > > On 3.2 i used the variable $time() to get the current time for some process. > > Every 2 minutes a RTIMER process started, checked the $time(hour) and > $time(min) and on a certain time executed something. > > This process stopped working on 3.3.2 > > When i debug, i see that the $time() variable keeps on giving me the same > hour and minute that was given on the first run. > > Meaning the $time() variable does not give the current time. > > I notice also that the $Ts that on 3.2 gave a current time stopped doing it > and $TS does it now.... > > Any help here? > > Is it OK and i need to check time another way? Or something in the $time() > has changed...? > > Thanks, > > Uri >

1 0

dialog ka_timer and ka_interval problem
by Kristofer Signer 21 Nov '12

21 Nov '12

Hello all, first of all I would like to thank you all your help. I've tried to implement dialog keep-alive messages but cant make kamailio send more then just one SIP OPTION request to each end. The OPTION request is sent after ka_interval time but no more OPTION requests are sent during the call. How should this be implemented? and was is the difference and relation between ka_timer and ka_interval. tanks ./Kristofer

2 2

LCR weight
by Camila Troncoso 20 Nov '12

20 Nov '12

Hi, I have a question regarding LCR module and their weight variable. If I have 4 gateways for the same prefix, and with the same priority: Rule_id Gw_id Priority weight 36 68 1 60 36 69 1 14 36 70 1 13 36 71 1 13 The weight variable acts as a percentage of utility of that gateway? If for example I have 100 calls, the gateway with gw_id 68 will receive 60 calls , and the other 14, 13 and 13? Regards, Camila *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 3

Getting started with Kamailio
by Joel Smith | VOZELIA 20 Nov '12

20 Nov '12

Hi, First of all say that I'm new to Kamailio. I've been working with asterisk for a while now and it's time to get started with the sip router project. My idea for a start is to reduce the load of our asterisk servers and once we have good knowledge on how kamailio works, migrate some of our services. I've setup a server in our lab with kamailio 3.3.0 and siremis 3.3.0, created a user (subscriber) and registered with a softphone. So far OK. Now I'm trying to make a call and route it through one of our asterisk servers for testing. To do this I need LCR module (thanks Sébastien, I wasn't subscribed to this list yet) and configure it. After reading http://kamailio.org/docs/modules/stable/modules/lcr.html#id2553110 I've added in kamailio.cfg: #!define WITH_LCR #!ifdef WITH_LCR loadmodule "lcr.so" #!endif #!ifdef WITH_LCR # ----- lcr params ----- modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "tag_avp", "$avp(lcr_tag)") #!endif And now I see LCR module being loaded at startup. But, I don't understand what gw_uri_avp, ruri_user_avp or tag_avp mean. Where can I find some doc/manual to understand what AVP means and does? (For example the $avp(i:709), etc.) I've added a LCR GW, LCR TARGET and LCR RULE with the following data (I have no idea if it's completely wrong): mysql> select * from lcr_gw; +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | 1 | 0 | TEST | XX.XXX.XXX.X | test12345678.testing.com | 5060 | NULL | 1 | 1 | NULL | NULL | NULL | 0 | NULL | +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ 1 row in set (0.00 sec) mysql> select * from lcr_rule; +----+--------+--------+----------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+--------+----------+-------------+---------+---------+ | 1 | 1 | .* | | | 0 | 1 | +----+--------+--------+----------+-------------+---------+---------+ 1 row in set (0.00 sec) mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 1 | 1 | 1 | 1 | 1 | +----+--------+---------+-------+----------+--------+ 1 row in set (0.00 sec) mysql> Am I still missing basic stuff to be able to place a call ? (To get started I'd like to route all calls to our Asterisk server) I'm glad to read documentation so If all of this is already somewhere please guide me cause I haven't found too much... Thanks in advanced. Regards, Joel.

4 4

t_branch_timeout()
by Alex Balashov 20 Nov '12

20 Nov '12

I am using the latest pull of branch 3.3, and t_branch_timeout() doesn't work if the previous branch resulted in a reply (i.e. a negative final reply). That is to say, t_branch_timeout() == FALSE on a genuine timeout, as long as the preceding branch provided a result (such as 404). t_branch_timeout() == TRUE if this is a timeout, on the first branch. In other words, it appears that if t_any_replied() == TRUE, then t_branch_timeout() == FALSE, even for this branch, and even if the branch really did time out (no response at all from destination). That's not how the documentation says it should behave. Am I doing something wrong? If so, how do I find out if _this particular_ branch timed out, from the failure_route[]? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 3

temporary SIP user
by Bolang 20 Nov '12

20 Nov '12

Hi all, is there any standard mechanism to create a temporary user? i know i can create it and then delete it after some amount of time. but, i'm looking for an established/standard way of doing this. Ideally, the user will be automatically deleted after unregistered. thanks

2 2

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 20 Nov '12

20 Nov '12

I'd be curious to know if your performance might actually be higher, owing to improved CPUs since the study was done, hyperthreading, more frames per NIC interrupt, etc. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:Thank you, setting ulimits worked! And the performance is the same as stated in the document I mentioned! :) One more thing, I found these errors in syslog: Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:211]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5270]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location This is possibly something related to my sipp scenario, this is the INVITE sent from sipp <![CDATA[ INVITE sip:800@perftest.vm SIP/2.0 Via: SIP/2.0/[transport] 10.0.2.36:[local_port];branch=[branch] From: "700" <sip:500@10.0.2.36>;tag=[call_number] To: <sip:800@perftest.vm> Call-ID: [call_id] CSeq: 1 INVITE Contact: "700" <sip:700@10.0.2.36:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] 10.0.2.36 s=- c=IN IP[local_ip_type] 10.0.2.36 t=0 0 m=audio [auto_media_port] RTP/AVP 8 a=rtpmap:8 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 ]]> Do you see something that could cause this error ? Otherwise the call is initiated ok, but I really dont understand what is so strange to kamailio in this INVITE. On Thu, Sep 13, 2012 at 6:37 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >> You mean on the proxy side? I'm running rtpproxy as root, limits are >> still applied ? ulimit -s unlimited should do the trick ? > > Yes, they usually applied even for superuser, and yes - this should > help (if that's the issue). > > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

6 14

Clarification on is_known_dlg()
by David 20 Nov '12

20 Nov '12

Hello, I am running Kamailio 3.2.3. I am storing dialogs in the database table dialog. loadmodule "dialog.so" modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 1) later I have : if ( is_known_dlg() ) { } I would like a clarification on how the matching is done. Given that I have realtime table setup, does that mean that dialogs that are in the table will return true when I call is_known_dlg() ? Does that mean that if the dialog is NOT in the table, is_known_dlg will return false ? Thanks for the clarification ! David

2 1

UAC and realm
by Johan Wilfer 20 Nov '12

20 Nov '12

Hi, I've done some tests with the UAC module to authenticate to a remote proxy. I've based my config on this example: http://docs.huihoo.com/opensips/tutorials/uac/ar01s06.html (example 9) I have found that if I send a call from a asterisk via kamailio to my remote proxy the realm the remote proxy uses is the ip-address of the asterisk-server (aaa.bbb.ccc.ddd). Like this: WWW-Authenticate: Digest nonce="1353399363:8d2317487ee1521328d0e2237e444e2d",algorithm=MD5,realm="aaa.bbb.ccc.ddd",qop="auth",stale=false. If I have a param like this, the auth works: modparam("uac","credential","username:aaa.bbb.ccc.ddd:password") But as I would like to use more than one remote proxy (=more than one provider) per asterisk server this needs some more work. Is there a way to get UAC to ignore the realm from the remote proxy and send the auth anyway? Alternate paths I've examined: 1. Rewrite from-domain to get the provider to use another realm: modparam("uac","credential","username:example.com:password") uac_replace_from("sip:$fU@example.com"); Works, but I don't think this is a very good solution. 2. Use auth_username_avp / auth_password_avp / auth_realm_avp to feed the uac_auth()-function the right credentials. modparam("uac","auth_username_avp","$avp(s:uac_user)") modparam("uac","auth_password_avp","$avp(s:uac_pass)") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") in the failure route: $avp(s:uac_user) = "username"; $avp(s:uac_pass) = "password"; $avp(s:uac_realm) = "aaa.bbb.ccc.ddd"; This works as well, but I would like to either get the realm the remote proxy sends and use it or make uac ignore the realm. The realm is unimportant as we already know what proxy we are talking to. I've tried $hdr(WWW-Authenticate) but it evaluates to null. Any ideas how to do this? Thanks! -- Johan Wilfer

2 2

Re: [SR-Users] Returning problem
by Moacir Ferreira 19 Nov '12

19 Nov '12

I wish it were that simple... I have done some "reading" and found that Ubuntu has changed to upstart http://upstart.ubuntu.com/cookbook/. However, I could not find how I can start a non "upstart" style init script like Kamailio has before the upstart styled ones... I guess that I have to either to get the Kamailio community to create an upstart script or find some help from the Ubuntu community how to overcome the problem. Anyway, I think the Kamailio team should address it as it will become a problem for everyone using Kamailio on Ubuntu. Any help? Moacir From: 4lists(a)gmail.com To: sr-users(a)lists.sip-router.org Date: Mon, 12 Nov 2012 14:42:31 -0200 Subject: Re: [SR-Users] Returning problem This is more a system initialization problem, so, take a look in your /etc/rc<run-level>.d directory (or the correspondent one), specially in the S* files and find out the start sequence. There You'll find the answer. Edson. Em 09/11/2012 17:03, Moacir Ferreira escreveu: > My kamailio start script has the following BEGIN INIT INFO: > > # > ### BEGIN INIT INFO > # Provides: kamailio > # Required-Start: $syslog $network $local_fs $time mysql > # Required-Stop: $syslog $network $local_fs mysql > # Default-Start: 2 3 4 5 > # Default-Stop: 0 1 6 > # Should-Start: postgresql mysql radius > # Should-Stop: postgresql mysql radius > # Short-Description: Start the Kamailio SIP proxy server > # Description: Start the Kamailio SIP proxy server > ### END INIT INFO > > However, when I boot my server Kamailio will start before mysql. So, > after boot I can start Kamalio but at boot time it will fail. > > Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. > All the code has been compiled at the server. > > Cheers! > > Moacir > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

2 5

how to use uac_auth in module UAC
by Freddy C 19 Nov '12

19 Nov '12

Hello, I have a question about UAC and function module "uac_auth" for a REGISTER request. My idea is to use this when I get a 401 unauthorized response. This answer is captured from failure_route. Then generate the header Authorized and forward the request REGISTER to Asterisk. I could help with some examples?. Freddy Condori Ch Ingeniero Investigador Telefonía IP

1 0

Sync nonce between various servers
by Andreas Granig 19 Nov '12

19 Nov '12

Hi, There are lots of parameters controlling the creation of nonce values on a server, and I'm curious if there is a way to kind of "sync" them between servers. The use case would be to have a UA send for example its registration to Proxy1. Proxy1 would challenge it, UA will send the registration again, this time with credentials. Proxy1 would look up the user based on $au/$ar in the subscriber table, and if it's not found, will look up the responsible proxy from another table (with key being $au@$ar), forward it to Proxy2, which then would be able authenticate the user. The reason for this is that the auth credentials are unique across all servers and reliably identify a user, whereas for example From could be something else (e.g. in case of an IP-PBX sending a CLI in the From-userpart). Challenging the user on the second proxy again would theoretically be possible, but if the UA gets a 401 twice (once from Proxy1, once from Proxy2), it'll most likely pop up a password form for soft-clients, so I want to avoid that. Any ideas how to accomplish that? Andreas

4 9

Problem routing to voicemail
by Christophe ROY 19 Nov '12

19 Nov '12

Hi everyone I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT: - I had to adapt it because I use LDAP authentication with Kamailio - I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM - we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0" For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone(a)domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile: Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It seems to happen on the if (!t_relay()) line in ROUTE[RELAY] 192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk. What could be the problem? Thanks for your help Christophe Below is my kamailio.cfg: #!WITH_DEBUG #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # ###!define WITH_NAT # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ###!define WITH_ASTERISK ###!define WITH_VOICEMAIL #!define WITH_LDAP #!define WITH_AUTH #!define WITH_MYSQL ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openserrw:openserrw@192.168.14.28/openser" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://asterisk:asteriskpwd@192.168.14.28/asterisk" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL6 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="mydomain.corp" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=tcp:192.168.14.24:5060 #listen=tcp:192.168.14.25:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "192.168.14.32" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.14.32" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "192.168.14.24" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" # loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #chris loadmodule "ldap.so" modparam ("ldap", "config_file", "/etc/kamailio/ldap.cfg") #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_tmp") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # suppress the check for the CSEQ method # modparam("sanity", "default_checks", 967) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #edit asterisk #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 256) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif #chris commented out this part # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.14.25:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@teopad-toip.corp") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); xlog("L_INFO","apres REQINIT"); # NAT detection route(NATDETECT); xlog("L_INFO","apres NATDETECT"); # handle requests within SIP dialogs route(WITHINDLG); xlog("L_INFO","apres WITHINDLG"); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); xlog("L_INFO","apres t_check_trans"); # authentication route(AUTH); xlog("L_INFO","apres AUTH"); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); xlog("L_INFO","apres RECORD ROUTE"); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } xlog("L_INFO","apres INVITE"); # dispatch requests to foreign domains route(SIPOUT); xlog("L_INFO","apres SIPOUT"); ### requests for my local domains # handle presence related requests route(PRESENCE); xlog("L_INFO","apres PRESENCE"); # handle registrations route(REGISTRAR); xlog("L_INFO","apres REGISTRAR"); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); xlog("L_INFO","apres PSTN"); # user location service route(LOCATION); xlog("L_INFO","apres LOCATION"); route(RELAY); xlog("L_INFO","apres RELAY"); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. xlog("L_INFO","Dans route relay"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","avant manage branch"); t_on_branch("MANAGE_BRANCH"); xlog("L_INFO","avant manage reply"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { xlog("L_INFO","avant manage failure"); t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { xlog("L_INFO","reply error"); sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # if(!sanity_check("1511", "7")) # { # xlog("Malformed SIP message from $si:$sp\n"); # exit; # } } # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Dans WITHINDLG"); if (has_totag()) { xlog("L_INFO","dans has totag"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO","looseroute"); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_INFO","ack"); # ACK is forwarded statelessy route(NATMANAGE); } xlog("L_INFO","relay"); route(RELAY); } else { xlog("L_INFO","else"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests xlog("L_INFO","subscribe avant presence"); route(PRESENCE); xlog ("L_INFO","apres presence"); exit; } if ( is_method("ACK") ) { xlog("L_INFO","else ack"); if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server xlog("L_INFO","else ack avant relay"); t_relay(); xlog("L_INFO","else ack apres relay"); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO","else final"); exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #edit asterisk #!ifdef WITH_ASTERISK xlog ("L_INFO","avant regfwd dans registrar"); route(REGFWD); xlog ("L_INFO","apres regfwd dans registrar"); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #edit asterisk #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { #if new call from out there - send to Asterisk # - non-INVITE requests are routed directly by Kamailio # - traffic from Asterisk is router also directly by Kamailio xlog ("L_INFO", "avant toasterisk dans location"); route(TOASTERISK); xlog ("L_INFO", "apres toasterisk dans location"); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #edit asterisk #!ifdef WITH_ASTERISK #do not auth traffic from Asterisk: trusted! xlog ("L_INFO", "avant if route fromasterisk"); if(route(FROMASTERISK)) return; #!endif if (is_method("REGISTER")) # { # # authenticate the REGISTER requests (uncomment to enable auth) # if (!www_authorize("$td", "subscriber")) # { # www_challenge("$td", "0"); # exit; # } # # if ($au!=$tU) # { # sl_send_reply("403","Forbidden auth ID"); # exit; # } { #edit asterisk ##!ifdef WITH_ASTERISK # xlog ("L_INFO", "dans auth / authcheck sipusers"); # if (!auth_check("$fd","sipusers","1")) ##!else if(is_present_hf("Authorization")) ##!endif { # ldap search if (!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?teopad-Sip-Username,teopadSipPassword?one?(teopad-Sip-Username=$fU)")) # if (!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?sAMAccountName,?one?(sAMAccountName=$fU)")) { switch ($retcode) { case -1: # no LDAP entry found sl_send_reply("404", "User Not Found"); xlog("L_INFO", "ldap_search: NO found [$retcode] entries for (uid=$fU)"); exit; case -2: # internal error sl_send_reply("500", "Internal server error"); exit; default: exit; } } ldap_result("teopad-Sip-Username/$avp(s:username)"); ldap_result("teopadSipPassword/$avp(s:password)"); xlog("L_INFO", "ldap_search: found [$retcode] entries for (uid=$fU)"); if(!pv_www_authenticate("$td", "$avp(s:password)", "0")) { xlog ("L_INFO", "ldap pv_authenticate failed") ; www_challenge("$td","1"); exit; } save("location"); sl_send_reply("200", "ok"); xlog ("L_INFO", "ldap pv_authenticate ok") ; exit; } else { www_challenge("$td","1"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # # authenticate if from local subscriber if (from_uri==myself) { # if (!proxy_authorize("$fd", "subscriber")) { # proxy_challenge("$fd", "0"); # exit; # } if (is_method("PUBLISH")) { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; xlog ("L_INFO", "tU = $tU") ; xlog ("L_INFO", "fd = $fd") ; xlog ("L_INFO", "rd = $rd") ; if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID au!=fu ou au!=tu"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; # if ($au!=$fU) { # sl_send_reply("403","Forbidden auth ID au!=fu"); # exit; # } } #consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); #rtpproxy_manage("co","82.127.95.167"); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "tcp:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); xlog("L_INFO","tovoicemail ru: $ru"); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_INFO","dans managereply"); xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]"){ xlog("L_INFO","avant route natmanage"); route(NATMANAGE); } } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #edit asterisk #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { xlog ("L_INFO", "Dans FROMASTERISK? $si / $sp"); if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); xlog ("L_INFO", "Dans TOASTERISK $du") ; xlog ("L_INFO", "Juste avant route relay"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { xlog("L_INFO", "Dans REGFWD"); if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; xlog("L_INFO","avant if dans regfwd"); if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif

2 3

Problem with include_file quotes
by Pirjo Ahvenainen 19 Nov '12

19 Nov '12

Hi All, I've been away from Kamailio for a few months, but recently I started working with it again. I could use a hand in a small problem: I'm using include_file statement to give some structure to my config file, but when checking the config I keep getting error message: 0(4290) : <core> [cfg.lex:1554]: included file name must be between quotes: "defined_modules.cfg" I'm using Kamailio 3.3.2 (version: kamailio 3.3.2 (x86_64/linux) 78d4b1) on Ubuntu. And here's the beginnign of the config file: #!KAMAILIO # include_file "defined_modules.cfg" include_file "defined_values.cfg" include_file "global_parameters.cfg" Those look like quotes to me, but could the problem be something like encoding of the config file itself, something in the included files, or is there something else I can check? I don't have extra whitespaces etc, perhaps a typo I just don't see? Cheers, Pirjo

2 3

How to get dialog variable externally?
by Mino Haluz 19 Nov '12

19 Nov '12

Hi, I'm using dialog module and I set some dlg_var for every dialog. Is it possible to get this variable value with MI command somehow ? I know to get list of dialogs using dlg_list only. Thanks! Mino

2 1

db_sqlite bugs
by Pedro Antonio Vico Solano 19 Nov '12

19 Nov '12

5 11

Planning for memory
by Moacir Ferreira 19 Nov '12

19 Nov '12

Hi, I am trying to figure out how much memory a Kamailio server should have to support an X number of users, considering that I am running Kamailio, MySQL and RTPproxy on the same machine. I am planning to use Kamailio at medium/large enterprises, not at service providers but I got the following questions: - Will Kamailio dynamically allocate the system memory it needs or I have to tune up the memory it takes? I.e.: If I use 8G memory, will it take what it needs or I need to adjust it somewhere? - How much memory is reasonable for 10K, 50K and 100K users in such "enterprise" scenarios? - Can you reference any documentation from where I can get this information? Thanks, Moacir

2 1

Kamailio cfg file doubt
by Kamal Palei 17 Nov '12

17 Nov '12

Dear All When I get INVITE (received in UDP), I send it to another proxy in TLS. For this purpose I have modified kamailio.cfg route section , please refer the lines in red. Just before sending in TLS, I modify the IP address in SDP, so that media packets will pass through RTP proxy running When INVITE is reached the next proxy, I see the IP address is changed . So far so good. Next I am trying to modify the IP address in 200 OK to INVITE. However it is not taking effect. I hope we need to modify in onreply_route function. Something is wrong here. I am receiving 200 OK for INVITE in TLS. Next I am sending 200 OK for INVITE to endpoints using UDP. Not sure if onreply_route is being executed or not. Can somebody give me pointer whats wrong here. *# Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy("r"); } if (isbflagset(FLB_NATB)) { fix_nated_contact(); } #!endif } * * # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); setflag(FLT_NATS); # NAT detection route(NAT); if(is_method("PUBLISH")) { loose_route(); t_relay(); exit; } if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:443"); exit(); } if(is_method("INVITE|BYE|CANCEL|SUBSCRIBE|REFER|NOTIFY")) { xdbg("incoming request\n"); route(RTPPROXY); t_relay_to("tls:115.114.48.75:443"); exit(); } *Thannks Kamal

2 2

segfault when phone registers on TLS
by Denis 15 Nov '12

15 Nov '12

Hi, I am using kamailio-3.3.2 on Ubuntu 11.04 x64 and when trying to connect via TLS from my Yealink phone kamailio crashes with the following backtrace: Loaded symbols for /lib/x86_64-linux-gnu/libnss_files.so.2 Core was generated by `/opt/kamailio/sbin/kamailio'. Program terminated with signal 11, Segmentation fault. #0 handle_ser_child (p=0x7fcbaea2eba0, fd_i=-1) at tcp_main.c:3575 3575 if (unlikely(p->unix_sock<=0)){ (gdb) bt #0 handle_ser_child (p=0x7fcbaea2eba0, fd_i=-1) at tcp_main.c:3575 #1 0x000000000051e5bc in send2child (tcpconn=0x7f8c001597d8, ev=<value optimized out>, fd_i=-1) at tcp_main.c:3975 #2 handle_tcpconn_ev (tcpconn=0x7f8c001597d8, ev=<value optimized out>, fd_i=-1) at tcp_main.c:4310 #3 0x0000000000527dbc in io_wait_loop_epoll () at io_wait.h:1092 #4 tcp_main_loop () at tcp_main.c:4656 #5 0x00000000004726f5 in main_loop () at main.c:1727 #6 0x000000000047402e in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2546 (gdb) The manual I am following is http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates Thanks, Den

3 10

Kamailio as SMTP server
by Krishna Kurapati 15 Nov '12

15 Nov '12

Hi, Can Kamailio act as SMTP server to receive emails and forward those to SIP users as MESSAGEs? It would be nice if postfix or some other rich SMTP server could function as module of kamailio using all the services provided by kamailio scripting. Krish Kura

2 1

Problem routing to voicemail
by Christophe ROY 15 Nov '12

15 Nov '12

Hi everyone I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT: - I had to adapt it because I use LDAP authentication with Kamailio - I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM - we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0" For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone(a)domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile: Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It seems to happen on the if (!t_relay()) line in ROUTE[RELAY] 192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk. My kamailio.cfg is available here as it's too big for the list: http://pastebin.com/zhncJ6j1 What could be the problem? Thank you very much for your help Christophe

1 0

Replicate dialog state
by Kristofer Signer 15 Nov '12

15 Nov '12

In my current setup I've register messages replicated to a second server. Is it possible to also replicate other messages so I can hold dialog state on both servers? I've tried to read the documentation and also google but can't find any examples. Thanks ./Kristofer

2 1

RPC, XMLRPC or other
by Kristofer Signer 14 Nov '12

14 Nov '12

Hello, I'm trying to dig in to the kamailio XMLRPC interfaces and the limitations I read in http://www.kamailio.org/docs/modules/stable/modules/xmlrpc.html#xmlrpc.impl… is really a no go for us. For example, when to retrieve user locations and list of dialogs we get a xml document which is not so well formatted and we can not successfully parse it in an xmlrpc-parser. Is there other options for xmlrpc module? thanks ./Kristofer

4 11

check if the sent reply was local
by Uri Shacked 14 Nov '12

14 Nov '12

Hi, what is the command to check if the reply was recieved or localy generated? Uri

3 3

Re: [SR-Users] sr-users Digest, Vol 90, Issue 39
by Yufei Tao 14 Nov '12

14 Nov '12

What I did to overcome this problem was writing a bash script, that checks if mysql is running, if so, start Kamailio, if not, start mysql first, then start Kamailio, and retries if things fail. And in /etc/rc.local, call this script. Seems to work fine on Ubuntu 10.04. Not tried 12.04 yet. Yufei > > > ------------------------------ > > Message: 2 > Date: Tue, 13 Nov 2012 18:15:05 -0500 > From: Moacir Ferreira <moacirferreira(a)hotmail.com> > Subject: Re: [SR-Users] Returning problem > To: Kamailio Mail List <sr-users(a)lists.sip-router.org> > Message-ID: <COL125-W3986EA29065F0966C8AE6C86C0(a)phx.gbl> > Content-Type: text/plain; charset="iso-8859-1" > > > I wish it were that simple... I have done some "reading" and found that Ubuntu has changed to upstart http://upstart.ubuntu.com/cookbook/. However, I could not find how I can start a non "upstart" style init script like Kamailio has before the upstart styled ones... I guess that I have to either to get the Kamailio community to create an upstart script or find some help from the Ubuntu community how to overcome the problem. Anyway, I think the Kamailio team should address it as it will become a problem for everyone using Kamailio on Ubuntu. Any help? Moacir From: 4lists(a)gmail.com > To: sr-users(a)lists.sip-router.org > Date: Mon, 12 Nov 2012 14:42:31 -0200 > Subject: Re: [SR-Users] Returning problem > > This is more a system initialization problem, so, take a look in your > /etc/rc<run-level>.d directory (or the correspondent one), specially in > the S* files and find out the start sequence. There You'll find the answer. > > Edson. > > Em 09/11/2012 17:03, Moacir Ferreira escreveu: >> My kamailio start script has the following BEGIN INIT INFO: >> >> # >> ### BEGIN INIT INFO >> # Provides: kamailio >> # Required-Start: $syslog $network $local_fs $time mysql >> # Required-Stop: $syslog $network $local_fs mysql >> # Default-Start: 2 3 4 5 >> # Default-Stop: 0 1 6 >> # Should-Start: postgresql mysql radius >> # Should-Stop: postgresql mysql radius >> # Short-Description: Start the Kamailio SIP proxy server >> # Description: Start the Kamailio SIP proxy server >> ### END INIT INFO >> >> However, when I boot my server Kamailio will start before mysql. So, >> after boot I can start Kamalio but at boot time it will fail. >> >> Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. >> All the code has been compiled at the server. >> >> Cheers! >> >> Moacir >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > > > >

1 0

update the to_tag on a local generated reply
by Uri Shacked 14 Nov '12

14 Nov '12

Hi, When i set a timer to "request time out" kamilio generates a 408 with now to_tag. I saved the to_tag from the 183 and try to update the acc table with the to_tag (i need that for my CDR to work) My question is (understanding there is no option to set the to_tag on the 408) is: How to i get the right timing to update using sqlops? i must update after the acc inserted the row to the acc table.... Any ideas? Thanks, Uri

2 3

Presence Dialog Info synchronisation issue
by David 12 Nov '12

12 Nov '12

Hello, I have two Kamailio boxes running. First one routes calls between my phones and asterisk. Here are the extra lines compared to the default config that came with my installation on the first server. Let's call this server kamailio : loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_dialoginfo.so" modparam("pua_dialoginfo", "include_localremote", 0) modparam("pua_dialoginfo", "include_tags", 0) modparam("pua_dialoginfo", "include_callid", 0) modparam("pua_dialoginfo", "caller_confirmed", 1) modparam("pua", "outbound_proxy", "sip:kamailio_presence_ip:5060") if ( is_method("SUBSCRIBE") ) { t_relay("kamailio-presence-ip", "5060"); } The second one handles presence stuff ( SUBSCRIBE and PUBLISH ), let's call it presence : After the standard sanity checks : if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; exit; The problem is that I have two phones connected to the same username on the server. Here is the sequence of events of a correctly functionning call : 1. Kamailio receives an INVITE 2. It sends the INVITE to each of the two phones 3. The first phone replies 180 Ringing 4. Kamailio sends a PUBLISH to presence 5. Presence replies with an OK, which includes a SIP-ETag header 6. Kamailio recieves the OK 7. Kamailio receives the 180 Ringing from the second phone 8. Kamailio sends a new PUBLISH with the SIP-If-Match using the value from the SIP-ETag header 9. Presence receives it, updates the presence table and sends an OK. This is ok, the trouble is when the second 180 Ringing reaches Kamailio before the OK to the publish with the SIP-ETag. 1. Kamailio receives an INVITE 2. It sends the INVITE to each of the two phones 3. The first phone replies 180 Ringing 4. Kamailio sends a PUBLISH to presence 5. Kamailio receives the 180 Ringing from the second phone 6. Kamailio sends a PUBLISH to presence 7. Presence replies with an OK, which includes a SIP-ETag header 8. Presence replies with an OK, which includes a second SIP-ETag header 9. Kamailio recieves the OK 10. Kamailio recieves the OK The presence table now has two seperate dialogs for the call because Kamailio didn't send the SIP-If-Match header as it did not yet receive the reply from presence. When the two phones start ringing at the same time, the time between the two packets arriving at the server is shorter than a round trip to the presence server. In the second scenario, Kamailio never updates the presence information using the first SIP-ETag, so the light on my phone stays stuck at "ringing". Where do I go from here ? Thanks, David

1 0

Returning problem
by Moacir Ferreira 12 Nov '12

12 Nov '12

My kamailio start script has the following BEGIN INIT INFO: # ### BEGIN INIT INFO # Provides: kamailio # Required-Start: $syslog $network $local_fs $time mysql # Required-Stop: $syslog $network $local_fs mysql # Default-Start: 2 3 4 5 # Default-Stop: 0 1 6 # Should-Start: postgresql mysql radius # Should-Stop: postgresql mysql radius # Short-Description: Start the Kamailio SIP proxy server # Description: Start the Kamailio SIP proxy server ### END INIT INFO However, when I boot my server Kamailio will start before mysql. So, after boot I can start Kamalio but at boot time it will fail. Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. All the code has been compiled at the server. Cheers! Moacir

2 1

LUA 5.2 - Support
by David | StyleFlare 12 Nov '12

12 Nov '12

Curious to know if there is any plans to support LUA 5.2. Seems like this is a newer release, I tried to build kamailio lua modules using 5.2 but it was not supported. Seems like some minor changes in the way LUA handles some things. Any ideas? Thanks.

2 1

xcap: no more space for spec value
by Juha Heinanen 12 Nov '12

12 Nov '12

i got the these error messages when my xcap server tried to serve 'GET xcap-caps' request: Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: INFO: XCAP service GET <xcap-caps> Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [pvapi.c:1305]: no more space for spec value Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [pvapi.c:1314]: buffer overflow -- increase the buffer size... Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [sr_module.c:1665]: Could not convert the PV-formated string to str Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: xhttp [xhttp_mod.c:461]: unable to get body config code where this comes is this: xlog("L_INFO", "XCAP service $rm <$xcapuri(u=>auid)>\n"); $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.pres-content</auid> <auid>org.openmobilealliance.search</auid> <auid>org.openmobilealliance.xcap-directory</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:prs:pres-content</namespace> <namespace>urn:oma:xml:xdm:search</namespace> <namespace>urn:oma:xml:xdm:xcap-directory</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "OK", "application/xcap-caps+xml", "$var(xbody)"); i have not checked yet, which buffer should be increased, but would it be a good idea to make the default bigger? -- juha

2 2

Use different flags in rtpproxy_manage() based on from which route its called
by Aft nix 12 Nov '12

12 Nov '12

Hi, rtpproxy_manage() combines both rtpproxy_offer() and rtpproxy_answer(). Now if i want different flags set for rtpproxy_offer() and rtpproxy_answer() then how should i use rtpproxy_manage()? To specific, what i want : route[REQUEST]{ ..... rtpproxy_offer("z20"); ... } onreply_route[]{ .... rtpproxy_answer("z120"); .... } Now in kamailio.cfg there is one call rtpproxy_manage() in route[NATMANAGE]. route[NATMANAGE] is called from both route[REQUEST] and onreply_route[]. So i can't emulate the functionality described above. My question is using only rtpproxy_manage() is it possible to emulate the above functionality? Thanks in advance. -- -aft

3 3

Homer multiple domains
by Reda Aouad 11 Nov '12

11 Nov '12

Hi I am using homer and managing multiple SIP domains. My problem is that in the sip_capture table, the FROM and TO domains are not logged in the FROM and TO fields. The MSG field contains the domains, but the queries involving domains become tricky and heavy to execute. Is there a way I can have the domains in their respective FROM/TO columns? Thanks Reda

1 0

how to use "t_drop_replies" but keep the last reply for acc
by Uri Shacked 11 Nov '12

11 Nov '12

Hi, I am trying to make an option of "route when no answer" or " route when busy". What I am doing is checking the reply and if "busy", for example, I use "t_drop_replies". Then, I set the new number and route[relay] again. On the accdb table, I get the first invite with 183 and after that the second invite with 183 and with 200. I would like to do exactly what i do, but would like to see on the accdb the 486 reply from the first invite. how do i do it? BR, Uri

4 11

Kamailio language definitions- dictionary
by kellon＠sis-grenada.com 11 Nov '12

11 Nov '12

Hi All Can anyone direct me to some go literature/documentation on the language/functions used in the kamailio config file. E.g cfg_get(), strempty(), proxy_authorize(), sl_send_reply() etc.... Thanks Kellon

2 1

No Debug Log message
by kellon＠sis-grenada.com 10 Nov '12

10 Nov '12

Hi all I am currently having to modify the kamailio config file without the benefit of log messages in /var/log/message. I am using the default configuration parameter settings of: #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 Has anyone experience this issue before or have any suggestions on howto fix. Making changes to the config file line by line and restarting the daemon is killing me!!!

2 1

Kamailio CDR logging to RADIUS
by Owen Lynch 10 Nov '12

10 Nov '12

Hi, I have transaction accounting to RADIUS setup and get the start and stop events logged to file when a call is made. I want to extend this to also get CDR logs - call duration etc. To do this I understand I need to configure the acc module and load and configure the dialog module: #!define FLD_DLG 4 modparam("acc", "cdr_enable", 1) modparam("dialog", "dlg_flag", FLD_DLG) I was expecting extra entries in the RADIUS file but they don't appear. I think this is because I need to set and clear the dialog flag at appropriate places in the route script (e.g. setflag(FLD_DLG)) but I'm not sure where. Should this be done in the INVITE and BYE method handling as is currently done for the transaction accounting, e.g.: # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } Thanks, Owen Lynch

3 4

"branch" tag in the "Via" header of the ACK message for the re-INVITE
by Alex Solt 10 Nov '12

10 Nov '12

Hi, We have a problem with the "branch" tag in the "Via" header of the ACK message for the re-INVITE. The "branch" tag value is not a unique number so the end point thinks the ACK is belong to the previous call. This causes the active calls fail after 15 min (right after the ACK message for re-INVITE passed by kamailio proxy). Is there a way to set the "branch" tag in the "Via" header of the ACK message for the re-INVITE to be a unique number? PS1: We are running kamailio v3.3.PS2: we did not have this issue with the old version of openser. Thanks,AS

2 3

Dispatcher question
by Mino Haluz 10 Nov '12

10 Nov '12

Hi, I have simple dispatcher that is dispatching initial invites to the core router. The question is, how can I tell kamailio to not add the Via header in the INVITE. I mean, I want that this dispatcher will be completely transparent to the core router, so any responses will go directly to the original sender, not to the dispatcher. This is my request route: request_route { if (is_method("INVITE")) { if (!ds_select_domain(1, 0)) { xlog("L_INFO", "[MAIN] ERROR: Proxy1 failed"); if (!ds_select_domain(2, 0)) { xlog("L_INFO", "[MAIN] FATAL: Proxy2 failed"); exit; } } xlog("L_INFO","[MAIN] Dispatching INVITE to $rd"); forward(); } } 1 and 2 are defined in dispatcher.list. If it can be done in kamailio, please how. Thank you! Mino

3 5

presence module makes excessive cleanups when subs_db_mode=3
by Andrew Pogrebennyk 09 Nov '12

09 Nov '12

Hi, when presence module is running with subs_db_mode=3 it makes an excessive number of SQL select queries, litelly dozens per second: > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=64 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=116 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=116 AND event='presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=64 AND event='presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=14 AND event<>'presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=14 AND event='presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=118 AND event<>'presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=66 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=118 AND event='presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=66 AND event='presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=16 AND event<>'presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=16 AND event='presence.winfo' Steps to reproduce: registers two subscribers in jitsi and add them to contact lists of each other. The kamailio version is 3.3.2. I'm not doing anything special configuration-wise: loadmodule "presence.so" modparam("presence", "db_url", "mysql://kamailio:snbF93@localhost/kamailio") # in 3.3 the fallback2db change to subs_db_mode modparam("presence", "subs_db_mode", 3) modparam("presence", "notifier_processes", 3) loadmodule "presence_xml.so" modparam("presence_xml", "db_url", "mysql://kamailio:snbF93@localhost/kamailio") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) # retry-after 5 minutes modparam("presence_xml", "xcapauth_userdel_reason", "probation;retry-after=300") Q: it is a bug of a feature? :) Thanks. Andrew

2 2

xmlrpc dlg.end_dlg problem
by Kristofer Signer 09 Nov '12

09 Nov '12

Hi all, I have an kamailio 3.3.2 server deployed with dialog awareness. When call dlg.end_dlg over xmlrpc I get an 200 OK response but the dialog is not ended. What I can do; call dlg.list (xmlrpc) and get the active dialog list, xmlrpc interface working. call dlg.dlg_list (xmlrpc) and get description of one dialog, xmlrpx method call with parameters working. call dlg_end_dlg (kamctl) and successfully end the dialog. so my questions are... Is the xmlrpc method dlg.end_dlg known to work? and if, what have I missed? thanks ./Kristofer Signer $ /usr/local/kamailio-3.3/sbin/kamailio -V version: kamailio 3.3.2 (x86_64/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, select, kqueue. id: unknown compiled on 09:53:26 Nov 2 2012 with gcc 4.2.1

2 3

dispatcher: What's "first destination" in a database
by Olle E. Johansson 08 Nov '12

08 Nov '12

Friends, Dispatcher has a mode 8 - "use first destination" which makes sense for a textfile driven dispatcher configuration. Now if I use a database - will dispatcher first sort on priority, then take the first? It seems so from the documentation of the text file: priority: sets the priority in destination list (based on it is done the initial ordering inside the set) Which one is top priority - lowest or highest value? I can update the docs to clarify if someone helps me with the answer :-) /O

2 1

Another NAT question
by Moacir Ferreira 08 Nov '12

08 Nov '12

Hi, I have a scenario where I got a firewall with 3 interfaces: internal, DMZ and external. All the traffic from internal going to external is NATed. However, the traffic between internal and DMZ is NOT NATed. The external and DMZ are using public IP addresses. On the DMZ I have a Kamailio server running with RTPProxy + NAT Helper. Everything works fine when public (from the internet) users (UAs) are behind NAT as Kamailio will force the RTP to go via RTPProxy. However, when the public user has a public IP, it will fail to establish the RTP to a user who registered on Kamailio coming from the internal firewall interface. The reason is that, as I don't do NAT between internal and DMZ firewall interfaces, Kamailio will not RTPRroxy in between the UAs because, from the way Kamailio detects NAT, they are not behind NAT. So the public user UA tries to reach a private IP address. If the "inside" user tries to call a public user (a user on the Internet with a public IP), it works. Yes, I could do NAT in between the internal and DMZ firewall interfaces. However, this would force all RTP traffic of my UAs at the LAN go to Kamailio RTPProxy, an bad effect that I would like to avoid. So my question is: what would be the best approach to solve this issue using Kamailio and RTPProxy in such scenario? Cheers! Moacir

2 4

siptrace buffer
by Martin Formanko 08 Nov '12

08 Nov '12

Hi, do you know what happens if the database for siptrace is not reachable ? I assume siptrace is non-blocking in new versions of kamailio, but is there any buffer? Or insert queries are discarded when timed out ? Mino

2 1

Gateway failover on slow call setup
by Spencer Thomason 08 Nov '12

08 Nov '12

Hello, I'm trying to address a problem where sometimes our upstream providers have slow call setup times. We use drouting with multiple providers for each route to handle failover in the event of an outage but in this case, we get a 100 Trying but sometimes a 18[03] response takes 4-5 seconds or we never get it because the caller cancels the request. What would be a "safe" amount of time to allow a PSTN gateway to reply with a 180/183? What I'm trying to avoid is a scenario where the PSTN legitimately takes a long time to setup (i.e. an international call to some cell phone providers) and we needlessly failover which will actually increase the post dial delay for the user. My thought is this for example: route[PSTN] { t_set_fr(1000); # set fr timer to 1 second so we failover quickly if no provisional response t_on_reply("PSTN_REPLY"); # Routing logic } onreply_route[PSTN_REPLY] { if (status=~"1[0-9][0-9]") { t_set_fr(4000); # set fr timer to 4 seconds } if (status=~"18[03]") { t_reset_fr(); # reset timer if we get a 180 or 183 } } Also, if we failover because of slow call setup for a particular route, I'd like to temporarily avoid that gateway for that route (but only that route). I'm curious if anyone has any pointers or how they solved this. Thanks, Spencer

2 1

Adding Users not excepting email option
by kellon＠sis-grenada.com 08 Nov '12

08 Nov '12

I have attempted to add users as per documentation. kamctl add <username> <password> <email> However, Kamailio responds with the following: -- command 'add|passwd|rm' - manage subscribers add <username> <password> .......... add a new subscriber (*) passwd <username> <passwd> ......... change user's password (*) rm <username> ...................... delete a user (*) Is the option to add user with email address etc still available in the current release of 3.3 Thanks Kellon

2 1

Kamailio not responding
by kellon＠sis-grenada.com 07 Nov '12

07 Nov '12

Thanks Anton, Fred, Jeremy thanks for the suggestions. The problem was the firewall. I don't know why I didn't think of it. I learnt one other thing also the use of kamctl monitor. Just two other problems at the moment. I'll start a new thread for them

1 0

TLS Certificate Verification Issue
by Kamal Palei 07 Nov '12

07 Nov '12

Dear All I have modified kamailio,cfg and compiled all the modules with TLS enabled, and able to bring up the kamailio proxy properly. Kamailio proxy will receive the REGISTER message from endpoints in UDP , and want to send this REGISTER message to another intermediate proxy in TLS. For this purpose, I have added few lines in kamailio.cfg file as below. I have created the certificates, private keys as explained by README file in kamailio-3.1.5/modules/tls/ path. if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:443"); exit(); } Looks like this is taking effect. When Kamailio receives REGISTER message it is trying to do handshake with intermediate proxy. I used wireshark to see the handshake messages. 1. From Kamailio proxy, a TCP SYNC message is going to intermediate proxy. 2. intermediate proxy sends SYNC + ACK 3. Kamailio sends CLIENT HELLO 4. intermediate proxy sends SERVER HELLO, CERTIFICATE and SERVER HELLO DONE 5. The Kamailio sends ALERT (Level: Fatal, Description: Unknown CA) ---> IS something going wrong here.............. 6. Then Kamailio sends FIN + ACK Can somebody please let me know why the certificate verification fails (I get this log in console). How can I put a work around to avoid certification verification failure. Best Regards kamal

2 10

Kamailio not responding
by kellon＠sis-grenada.com 07 Nov '12

07 Nov '12

I have recently installed kamailio on CentOS 6.3 and configured a couple of SIP phone Linksys and Grandstream to test inercommunication. The SIP-Server/ Kamailio server ha started and is listen on port 5060. The UA are sending registered message to the SIP Server but there is no response from the SIP server. I left the syslog at default but not seeing any messages on the /var/log/messages. Can someone tell me if I have missed some configuration information. Note: I registered two users 101 and 202 using the command kamctl add 101 101. Is it possible to check whether the Users added have been added to the system??. and does anyone have any idea how to to troubleshoot a response from the Server. I am using Wireshark to monitor communication between between UA and SIP Server

4 3

Problem with my kamailio installation
by Ramazan Yilmaz 06 Nov '12

06 Nov '12

Hi, I installed kamailio 3.2.x a few times before and they were really fine. I decided to install a fresh one and followed the instructions on http://www.kamailio.org/wiki/install/3.3.x/git. However there is a problem: clients cannot register to kamailio. Each registration attempt produces the following error in the syslog: Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [tcp_main.c:3578]: BUG: handle_ser_child: fd -1 for 0 (pid 32414) Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [pass_fd.c:103]: ERROR: recv_all: 1st recv on 56 failed: Bad file descriptor Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [tcp_main.c:3363]: ERROR: handle_tcp_child: read from tcp child 0 (pid 0, no 0) Bad file descriptor [9] Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 17 on 56: Bad file descriptor (9) Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: ERROR: <core> [tcp_main.c:4006]: ERROR: send2child: send_fd failed for 0x7f510bc41588 (flags 0x4018), fd 17 What could be the problem? Any suggestion? In case you need, the server is ubuntu server 11.10, and I installed including mysql and tls modules.

2 15

ERROR: presence not working correctly with Kamailio 3.3.2-1.1 and Postgresql 9.1
by Robert Verspuy 05 Nov '12

05 Nov '12

Hi All, I've got a problem with Kamailio 3.3.2-1.1 on CentOS 6.3 with locally postgresql 8.4.13-1.el6_3 libraries, connected to a remote postgresql91-9.1.6-1PGDP.rhel6 server The presence module tries to insert a record in the presentity table, with some xml in the body column. But the body value in the postgresql table is not represented as a string '<?xml ...etc' But encodes as '\x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462...etc' This gives problems when the presense module is trying to send out a notify based on the value in the database. With debugging, this gives the error: Entity: line 1: parser error : Start tag expected, '<' not found x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462d38273f3e3c The \x in front of the hex encodes string is not properly interpreted by postgresql or kamailio. The body column of the table presentity is a bytea column, and I believe postgresql 9 outputs these bytea columns a bit differently then postgresql 8. When I try to run the sql statement: insert into presentity (domain,username,event,etag,sender,body,received_time,expires ) values ('newsip.lifexs.nl','00086','presence','a.1352107949.18632.27.0','','<?xml version=''1.0'' encoding=''UTF-',1343534532,1345213723) On the postgresql 9.1 server locally, I still get the '\x3c3f786d6c... etc' value in the database. The bytea_output setting in postgresql 9 is now standard set to 'hex' in stead of escape. I've changed the bytea_output setting in postgresql.conf to 'escape', and then the presence is working without any issues. So maybe the database module of postgresql has to be changed to pick up these bytea encodings properly? Because i think kamailio should work correctly with the default postgresql settings. With kind regards, Robert Verspuy -- *Exa-Omicron* Eenspan 8-K 3897 AL Zeewolde http://exa.nl

2 1

Kamailio sending re invite to the next proxy
by Maurice Vistan 05 Nov '12

05 Nov '12

Hi All, I am using kamailio as dispatcher and set up kamailio just to send 200 okay back for the method register, I would like check if its normal for the kamailio send another invite on the next proxy listed on my dispatcher.lst after the first proxy process the invite and auth. Below is how the call is being process. user kamailio 1proxy 2proxy user2 invite |----------> | 100 | <---------- | invite | | ---------> | 100 |<--------- | auth |<-----------| auth | <------------------------| ack |------------->| ---------->| Invite |-------------->| --------------------------> | then process Is there any I can limit this? or it is realy the behavior? Thanks a lot! Maurice

2 2

ERROR: presence not working correctly with Kamailio 3.3.2-1.1 and Postgresql 9.1
by Robert Verspuy 05 Nov '12

05 Nov '12

Hi All, I've got a problem with Kamailio 3.3.2-1.1 on CentOS 6.3 with locally postgresql 8.4.13-1.el6_3 libraries, connected to a remote postgresql91-9.1.6-1PGDP.rhel6 server The presence module tries to insert a record in the presentity table, with some xml in the body column. But the body value in the postgresql table is not represented as a string '<?xml ...etc' But encodes as '\x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462...etc' This gives problems when the presense module is trying to send out a notify based on the value in the database. With debugging, this gives the error: Entity: line 1: parser error : Start tag expected, '<' not found x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462d38273f3e3c The \x in front of the hex encodes string is not properly interpreted by postgresql or kamailio. The body column of the table presentity is a bytea column, and I believe postgresql 9 outputs these bytea columns a bit differently then postgresql 8. When I try to run the sql statement: insert into presentity (domain,username,event,etag,sender,body,received_time,expires ) values ('newsip.lifexs.nl','00086','presence','a.1352107949.18632.27.0','','<?xml version=''1.0'' encoding=''UTF-',1343534532,1345213723) On the postgresql 9.1 server locally, I still get the '\x3c3f786d6c... etc' value in the database. The bytea_output setting in postgresql 9 is now standard set to 'hex' in stead of escape. I've changed the bytea_output setting in postgresql.conf to 'escape', and then the presence is working without any issues. So maybe the database module of postgresql has to be changed to pick up these bytea encodings properly? Because i think kamailio should work correctly with the default postgresql settings. With kind regards, Robert Verspuy -- *Exa-Omicron* Eenspan 8-K 3897 AL Zeewolde http://exa.nl

1 0

Entries in PUA List and RLS Watchers List
by Sangeeta Shah 03 Nov '12

03 Nov '12

Hello All, I am hoping the authors of the RLS/PUA modules can answer this question since I am not sure what's going on. I have the rls module enabled, with the following config spec: # ------rls module params ------ modparam("rls", "db_url", DBURL) modparam("rls", "db_mode", 2) modparam("rls", "integrated_xcap_server", 1) modparam("rls", "to_presence_code" ,10) modparam("rls", "server_address", "sip:rls@ip:5060") #!endif and modparam("pua_reginfo", "default_domain", "ip") modparam("pua_reginfo", "server_address", "sip:reginfo@ip") modparam("pua", "db_mode", 2) So for both I have DB_MODE set to DB ONLY. For whatever reason, even though all I am doing is bringing up my client I see over 250+ entries in the rls_watchers table for the same watcher and over 100 entries in the pua list table for the same presentity? Anyone have any idea what would be triggering this. There is definitely not anymore more than the normal messaging going on between the client and the server. Only error I see in the syslog: Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: db_mysql [km_dbase.c:122]: driver error on query: Duplicate entry '1350937406' for key 'expires_idx' Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11744]: DEBUG: presence [subscribe.c:1216]: subs->contact= sip:rls@ip:5060 - len = 25 Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11761]: DEBUG: <core> [db_val.c:117]: converting STRING [8475551001] Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11746]: DEBUG: <core> [parser/msg_parser.c:626]: method: <SUBSCRIBE> Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: <core> [db_query.c:210]: error while submitting query Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11744]: DEBUG: rls [rls.h:241]: did= 41826031568472f9-11752@127.0.0.1;533cb9e91f4b999cf76861cbb9ed54ed-93a3;a6a1c5f60faecf035a1ae5b6e96e979a-464d Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11761]: DEBUG: <core> [db_val.c:117]: converting STRING [10.50.251.12] Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11746]: DEBUG: <core> [parser/msg_parser.c:628]: uri: <sip:8475551001@ip> Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: pua [pua_db.c:1149]: DB insert failed -- Sangeeta Shah

5 18

Error while "Make All"
by Techie Sup 02 Nov '12

02 Nov '12

Hello, I followed the instruction mentioned in Kamailio website. When I do a make all, I get the error. Please let me know how to fix this. I am new to Kamailio. ------------------- Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found CC (gcc) [M app_mono.so] app_mono_api.o In file included from app_mono_api.c:37: app_mono_api.h:27:26: warning: mono/jit/jit.h: No such file or directory app_mono_api.h:28:39: warning: mono/metadata/environment.h: No such file or directory In file included from app_mono_api.c:37: app_mono_api.h:35: error: expected specifier-qualifier-list before ‘MonoDomain’ app_mono_api.c:39:39: warning: mono/metadata/mono-config.h: No such file or directory app_mono_api.c:49: error: expected specifier-qualifier-list before ‘MonoDomain’ app_mono_api.c: In function ‘sr_mono_load_script’: app_mono_api.c:93: error: ‘sr_mono_load_t’ has no member named ‘next’ app_mono_api.c: In function ‘mono_sr_init_load’: app_mono_api.c:139: warning: implicit declaration of function ‘mono_config_parse’ app_mono_api.c:141: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:148: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:148: warning: implicit declaration of function ‘mono_jit_init’ app_mono_api.c:149: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:157: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:157: warning: implicit declaration of function ‘mono_domain_assembly_open’ app_mono_api.c:157: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:158: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:162: error: ‘sr_mono_load_t’ has no member named ‘next’ app_mono_api.c: In function ‘mono_sr_initialized’: app_mono_api.c:208: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c: In function ‘app_mono_exec’: app_mono_api.c:230: error: ‘sr_mono_env_t’ has no member named ‘msg’ app_mono_api.c:237: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:246: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:246: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:247: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:256: warning: implicit declaration of function ‘mono_jit_exec’ app_mono_api.c:256: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:256: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:257: warning: implicit declaration of function ‘mono_environment_exitcode_get’ app_mono_api.c:261: warning: implicit declaration of function ‘mono_jit_cleanup’ app_mono_api.c:261: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c: In function ‘app_mono_run’: app_mono_api.c:282: error: ‘sr_mono_env_t’ has no member named ‘msg’ app_mono_api.c:284: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:284: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:285: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:285: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:286: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:291: warning: implicit declaration of function ‘mono_domain_set’ app_mono_api.c:291: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:298: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:298: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c: At top level: app_mono_api.c:310: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ token app_mono_api.c:314: error: expected declaration specifiers or ‘...’ before ‘MonoString’ app_mono_api.c: In function ‘sr_mono_log’: app_mono_api.c:316: warning: implicit declaration of function ‘mono_string_to_utf8’ app_mono_api.c:316: error: ‘text’ undeclared (first use in this function) app_mono_api.c:316: error: (Each undeclared identifier is reported only once app_mono_api.c:316: error: for each function it appears in.) app_mono_api.c:316: warning: assignment makes pointer from integer without a cast app_mono_api.c:318: warning: implicit declaration of function ‘mono_free’ app_mono_api.c: At top level: app_mono_api.c:321: error: expected ‘)’ before ‘*’ token app_mono_api.c:328: error: expected ‘)’ before ‘*’ token app_mono_api.c:338: error: expected ‘)’ before ‘*’ token app_mono_api.c:408: error: ‘sr_mono_api_version’ undeclared here (not in a function) app_mono_api.c:410: error: ‘sr_mono_err’ undeclared here (not in a function) app_mono_api.c:411: error: ‘sr_mono_dbg’ undeclared here (not in a function) app_mono_api.c:412: error: ‘sr_mono_modf’ undeclared here (not in a function) app_mono_api.c: In function ‘sr_mono_load_class_core’: app_mono_api.c:420: warning: implicit declaration of function ‘mono_add_internal_call’ app_mono_api.c: At top level: app_mono_api.c:427: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ token app_mono_api.c:478: error: expected ‘)’ before ‘*’ token app_mono_api.c:530: error: expected ‘)’ before ‘*’ token app_mono_api.c:579: error: expected ‘)’ before ‘*’ token app_mono_api.c:632: error: expected ‘)’ before ‘*’ token app_mono_api.c:679: error: expected ‘)’ before ‘*’ token app_mono_api.c:729: error: ‘sr_mono_pv_gets’ undeclared here (not in a function) app_mono_api.c:729: error: initializer element is not constant app_mono_api.c:729: error: (near initialization for ‘_sr_M_export_pv[0].method’) app_mono_api.c:730: error: ‘sr_mono_pv_geti’ undeclared here (not in a function) app_mono_api.c:730: error: initializer element is not constant app_mono_api.c:730: error: (near initialization for ‘_sr_M_export_pv[1].method’) app_mono_api.c:731: error: ‘sr_mono_pv_seti’ undeclared here (not in a function) app_mono_api.c:731: error: initializer element is not constant app_mono_api.c:731: error: (near initialization for ‘_sr_M_export_pv[2].method’) app_mono_api.c:732: error: ‘sr_mono_pv_sets’ undeclared here (not in a function) app_mono_api.c:732: error: initializer element is not constant app_mono_api.c:732: error: (near initialization for ‘_sr_M_export_pv[3].method’) app_mono_api.c:733: error: ‘sr_mono_pv_unset’ undeclared here (not in a function) app_mono_api.c:733: error: initializer element is not constant app_mono_api.c:733: error: (near initialization for ‘_sr_M_export_pv[4].method’) app_mono_api.c:734: error: ‘sr_mono_pv_is_null’ undeclared here (not in a function) app_mono_api.c:734: error: initializer element is not constant app_mono_api.c:734: error: (near initialization for ‘_sr_M_export_pv[5].method’) app_mono_api.c:750: error: expected ‘)’ before ‘*’ token app_mono_api.c:801: error: expected ‘)’ before ‘*’ token app_mono_api.c:850: error: expected ‘)’ before ‘*’ token app_mono_api.c:895: error: expected ‘)’ before ‘*’ token app_mono_api.c:930: error: ‘sr_mono_hdr_append’ undeclared here (not in a function) app_mono_api.c:930: error: initializer element is not constant app_mono_api.c:930: error: (near initialization for ‘_sr_M_export_hdr[0].method’) app_mono_api.c:931: error: ‘sr_mono_hdr_remove’ undeclared here (not in a function) app_mono_api.c:931: error: initializer element is not constant app_mono_api.c:931: error: (near initialization for ‘_sr_M_export_hdr[1].method’) app_mono_api.c:932: error: ‘sr_mono_hdr_insert’ undeclared here (not in a function) app_mono_api.c:932: error: initializer element is not constant app_mono_api.c:932: error: (near initialization for ‘_sr_M_export_hdr[2].method’) app_mono_api.c:933: error: ‘sr_mono_hdr_append_to_reply’ undeclared here (not in a function) app_mono_api.c:933: error: initializer element is not constant app_mono_api.c:933: error: (near initialization for ‘_sr_M_export_hdr[3].method’) make[1]: *** [app_mono_api.o] Error 1 make: *** [modules] Error 1 Regards Selvam

3 4

Kamailio WebSocket support
by Owen Lynch 02 Nov '12

02 Nov '12

Hi, I see Peter Dunkley's websocket work is not in the latest 3.3 release (3.3.2). What is the road map for this? Thanks, Owen Lynch

2 1

Re: [SR-Users] Kamailio and sylkserver integration
by Vijay Thakur 01 Nov '12

01 Nov '12

Thanks for your reply. Will you please explain that how can i forward calls from kamailio to sylkserver. Regards, VJ++ Message: 7 Date: Wed, 31 Oct 2012 08:26:50 +0100 From: "Olle E. Johansson"<oej(a)edvina.net> Subject: Re: [SR-Users] Kamailio and sylkserver integration To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List"<sr-users(a)lists.sip-router.org> Message-ID:<D7F883EA-4231-4D21-A79F-BB3449F972B1(a)edvina.net> Content-Type: text/plain; charset=us-ascii 31 okt 2012 kl. 06:49 skrev Vijay Thakur<vijay.thakur(a)loopmethods.com>: > Dear Sir, > > Is there any one in the list who have set up (integrated) Kamailio with sylkserver for conference. There is very less available information on the internet on this topic. All the required changes are made in sylkserver config files. > Kindly advise for required changes in Kamailio. As far as I understand, Sylkserver is just a SIP server. Forward calls from kamailio to sylkserver using any of the forwarding commands. /O -- * Olle E. Johansson -oej(a)edvina.net * Kamailio & SIP Masterclass Miami FL December 2012 *http://edvina.net/training/

2 1

Two kamailio servers using same DB but different version table
by Uri Shacked 31 Oct '12

31 Oct '12

Hi, can two kamailio servers, one running 3.2 and the other 3.3 run using the same DB? one will turn to version table X and the other to version table Y (i am not using the modules that the DB table changed... BR, Uri

2 1

statefull proxy Q
by Alex Solt 31 Oct '12

31 Oct '12

Hi, Is there any function in kamailio that could differentiates between a new incoming INVITE message and a re-INVITE message for an existing active call? Thanks, AS

3 6

2600Hz selects Kamailio
by Olle E. Johansson 31 Oct '12

31 Oct '12

Friends, The 2600Hz project that builds an API-driven telephony service have selected to switch to Kamailio as a SIP proxy for their SBC. They blog about the switch, comparing Kamailio with their previous choice and coming to a conclusion: "To our minds, a more active development community means that other users are likely to help solve bottlenecks for us, so we spend less time worrying about our SBC, and more time writing fun new applications! Conclusion: We’re switching to Kamailio because: • Many of our Major Clients already use it • The Naming Conventions are more sensible • The Pace of Innovation on Kamailio is faster" Join me in welcoming 2600Hz to the Kamailio project! /O Links: * http://www.kamailio.org/w/2012/10/2600hz-selects-kamailio/ * http://blog.2600hz.com/post/34639934434/our-other-border-brother-kamailio

1 0

Kamailio and sylkserver integration
by Vijay Thakur 31 Oct '12

31 Oct '12

Dear Sir, Is there any one in the list who have set up (integrated) Kamailio with sylkserver for conference. There is very less available information on the internet on this topic. All the required changes are made in sylkserver config files. Kindly advise for required changes in Kamailio. With Regards, Vijay Thakur

2 1

additional user data for SIP gateway usage?
by Jeremy Ardley 31 Oct '12

31 Oct '12

Hi, I'm after advice. I'm developing a script to provide a degree of PBX functionality for a small to medium office using Kamailio & SEMS. This includes mapping internal extensions to external PSTN numbers via the SIP trunk provider. The incoming mapping is easy using the db_alias functionality. The outgoing mapping is more difficult as the SIP trunk provider requires both the Full National Number and an account number matching the FNN in the outgoing From header. I'd appreciate suggestions on the best way to store, retrieve, and manage the additional information required for outgoing PSTN calls. I think avpops will work but is there some other way to do this? Thanks,

2 1

SIP connection tracking
by Daniil Baturin 31 Oct '12

31 Oct '12

Hi all, We have two softswitch hosts that don't support TLS. We have a peer that requires TLS, they terminate all our connections on a single address. We want to use kamailio as a proxy for it. Could you point me to the documentation and/or working examples that describe how to implement connection tracking to ensure replies to connections originated from host A are relayed back to host A, and connections from host B get back to host B? Thanks in advance. -- #!/usr/bin/env perl @a=split(//, "daniil @ baturin . org" );# Daniil Baturin @b=split(//,q/Px%!+o0Q6lh*7dp$.@8#%|y{/);while($i<24){$_.= chr((ord(@b[$i])-ord(@a[$i])+62)%94+32);$i++};print"$_\n"#

2 1

Re: [SR-Users] Blacklist initialization
by Javi Gallart 30 Oct '12

30 Oct '12

Hi Olle What about doing the init stuff in event-route[htable:mod-init] ? Regards Javi ------Original Message------ From: Olle E. Johansson Sender: sr-users-bounces(a)lists.sip-router.org To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List ReplyTo: SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List Subject: [SR-Users] Blacklist initialization Sent: Oct 30, 2012 21:58 Hi! I am looking for a way to add addresses to the IP blacklist in Kamailio at startup. Is that possible? I know I can do it with sercmd, but I fail to find a documented way to do it in the configuration... Must be missing something simple. Thank you for your help! /O _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Blacklist initialization
by Olle E. Johansson 30 Oct '12

30 Oct '12

Hi! I am looking for a way to add addresses to the IP blacklist in Kamailio at startup. Is that possible? I know I can do it with sercmd, but I fail to find a documented way to do it in the configuration... Must be missing something simple. Thank you for your help! /O

1 0

Default database named now kamailio
by Daniel-Constantin Mierla 30 Oct '12

30 Oct '12

Hello, just short note to let everyone know that the default database for Kamailio is now named 'kamailio' - the change happened over the weekend as part of migrating from last bits from openser to kamailio namespace. Also, the default usernames and passwords to access the database are now: - kamailio with password kamailiorw for read-write operations - kamailioro with password kamailioro for read-only operations If you set db_url parameters in the config file, there should be no impact on existing configurations. You also have to make sure you set the right value in kamctlrc file if you update kamctl. This change impacts development version (to be released as 3.4.0). Two other components are in the to-do list for this change: perl and snmpstats Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Question about acc module
by Thorsten Willand 30 Oct '12

30 Oct '12

3 2

Unknown status in presence_dialog
by David 29 Oct '12

29 Oct '12

Hello, I am using the presence_dialoginfo module on a stand alone presence server. I have this : if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; If the server does not have any status information, it will send out a blank notify with a 0 content length. Is there anyway to tell Kamailio to send an xml doc with status "terminated" if the server does not know the status that was requested by the SUBSCRIBE? Thanks, David Server : version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 14:19:23 Apr 19 2012 with gcc 4.4.3

1 0

siptrace vs. sipcapture
by Mino Haluz 29 Oct '12

29 Oct '12

Hi, we are thinking about switching from siptrace to sipcapture + HEP encapsulation. Do you know some reasons why we should switch to sipcapture or why we should not ? ;) We have multiple SIP proxies and Cisco gateways. In earlier versions of kamailio was siptrace blocking, that's why we started to looking for some other capturing mechanism. I think now siptrace is buffered, so we are hesitating .. Mino

2 1

rtpproxy ipv4/ipv6 bridging question
by Juha Heinanen 29 Oct '12

29 Oct '12

is it possible to make rtpproxy to include both ipv4 and ipv6 address in sdp? if not, i don't see how ipv4/ipv6 bridging can be made to work when $rd is domain name. -- juha

1 0

NEW: new module: xhttp_pi
by Ovidiu Sas 28 Oct '12

28 Oct '12

Hello all, A new module is available in trunk. It provides a web provisioning interface: http://kamailio.org/docs/modules/devel/modules/xhttp_pi.html A sample config example is provided in the source tree: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex… An additional xml file is required for web config framework (what to configure, where and how). A sample xml file is provided in the source tree for dispatcher and dialplan tables: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex… When kamailio is up and running, a web provisioning interface for managing db records will be available. No supplementary web server needs to be installed in order to manage the db records. Also, it provides a tight control over what can be modified and how. Testing and feedback is appreciated. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

2 1

web homer not displaying data
by Christopher Holland 26 Oct '12

26 Oct '12

I have sipcapture and siptrace setup on the same server using kamailio 3.4.0-dev4 and webhomer 3.3.0 setup to capture sip traffic from a mirriored port on the LAN. Interface ETH1 is to capture and ETH0 is for managment. The sipcapture database is getting data put into it. I can log into webhomer, but see no data in the graphes nor can I search. I have double checked my database permissions and they seem to be set up correctly. webserver is set up with: Apache/2.2.16 (Debian) PHP5 MYSQL Json phpmyadmin Here is a copy of my kamailio.cfg and configuration.php rev_dns=no auto_aliases=no debug=1 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=5 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* IP and port for HEP capturing) */ listen=udp:127.0.0.1:9060 mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" loadmodule "pv.so" loadmodule "db_mysql.so" loadmodule "sipcapture.so" loadmodule "textops.so" loadmodule "siptrace.so" # ----- mi_fifo params ----- ####### Routing Logic ######## modparam("sipcapture", "db_url", "mysql://openser:******@localhost/homerdb") modparam("sipcapture", "capture_on", 1) /* activate HEP capturing */ modparam("sipcapture", "hep_capture_on", 1) /* IP to listen. Port/Portrange apply only on mirroring port capturing */ modparam("sipcapture", "raw_socket_listen", "10.0.0.0:5060-5080") /* Name of interface to bind on raw socket */ modparam("sipcapture", "raw_interface", "eth1") /* activate IPIP capturing */ #modparam("sipcapture", "raw_ipip_capture_on", 1) /* My table name*/ modparam("sipcapture", "table_name", "sip_capture") /* children for raw socket */ modparam("sipcapture", "raw_sock_children", 4) /* insert delayed */ #modparam("sipcapture", "db_insert_mode", 1) /* activate monitoring/mirroring port capturing. Linux only */ modparam("sipcapture", "raw_moni_capture_on", 1) /* Promiscious mode RAW socket. Mirroring port. Linux only */ modparam("sipcapture", "promiscious_on", 1) /* activate Linux Socket Filter (LSF/BPF) on mirroring interface. Linux only */ #modparam("sipcapture", "raw_moni_bpf_on", 1) /* sipcapture node */ modparam("sipcapture", "capture_node", "homer01") # check IP and port of your capture node modparam("siptrace", "duplicate_uri","sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag",22) modparam("siptrace", "trace_on", 1) # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { sip_capture(); drop; } onreply_route { sip_capture(); drop; } ################################## configuration.php <?php /*********************************************************************************/ /* Access db of homer */ define('HOST', "localhost"); define('PORT', 3306); define('USER', "webhomer"); define('PW', "******"); define('DB', "web_homer"); /* Homer connection * this user must have the same password for all Homer nodes * please define all your nodes in homer_nodes table */ define('HOMER_HOST', "localhost"); /* DEFAULT. Don't forget insert this host to your DB nodes table */ define('HOMER_PORT', 3306); define('HOMER_USER', "openser"); define('HOMER_PW', "******"); define('HOMER_DB', "homerdb"); define('HOMER_TABLE', "sip_capture"); /*********************************************************************************/ /* webHomer Settings * Adjust to reflect your system preferences */ define('PCAPDIR',"/var/www/webhomer/tmp/"); define('WEBPCAPLOC',"/webhomer/tmp/"); define('APIURL',"http://localhost"); define('APILOC',"/webhomer/api/"); /* INCLUDE preferences */ include("preferences.php"); ?>

2 2

Message Routing To/From Gateway
by Ed Brady 26 Oct '12

26 Oct '12

Hi, I am trying to route SIP messages to a gateway. I am using SIP MESSAGE over UDP as well as the MSILO module for storing messages for users that aren't currently logged in. I have read through the Dynamic Routing module info but am not able to determine the best way to accomplish the routing I require. Our scenario is outlined below. User A sends a message addressed to "sip:12345678900@mydomain.com" Kamailio at mydomain.com checks if 12345678900 is a registered user in the system. If not, the message header is translated to "sip:12345678900@gatewaydomain..com" and forwarded. Incoming messages need to be accepted from their IP as well. It would also be useful if the address came as "sip:2345678900@mydomain.com" could be translated to "sip:*1*2345678900@gatewaydomain.com". Is there any built in functionality I can configure for this, or will this need to be scripted? Thanks, Ed

2 9

[OT] the role of SBCs
by Daniel-Constantin Mierla 26 Oct '12

26 Oct '12

Hello, based on the outcome of the discussion carried in the thread: http://lists.sip-router.org/pipermail/sr-users/2012-August/074480.html I am looking again the clarify some aspects out there in the VoIP world. So, as mainly dealing with proxy/sip signaling deployments, it's very often to be the first one hit by support issues claiming things don't work. Then you investigate and end up in conclusions like in the above thread: "The problem was at the SBC, __where I did not expect it__." The underlined part heats me up a bit, because I never understood from where it comes this perception that SBC is a MUST-TO-HAVE and the PERFECT (never mistaken or breaking things) node in a VoIP networks. To some extent, the SBC is just a very costly SIP ALG, and a SIP ALG is there to break the things. I don't want to start like a flame war, but is it something that I am obviously missing in regards to what benefits a SBC can bring? I see only inconveniences: - another point of failure - it is a b2bua, therefore very unlikely to offer the same performances of a proxy - if transcoding is needed, a media server can be used behind the proxy, properly protected of attacks by the proxy and eventually deployed as a farm load balanced by the proxy - if topology hinding is wanted in the b2bua fashion (not the proxy fashion with encoding headers), then the b2bua can be behind the proxy, properly protected of attacks by the proxy and eventually deployed as a farm load balanced by the proxy - nat traversal was solved long time ago in proxy environment, being scalable by deploying a farm of rtp proxy I don't want to go to other features, including the transport layer, it's a clear win of the proxy in my experience (ok, being deep involved in this project). Then, what makes the SBC so desirable in many companies/voip deployments? If any SBC user here that can share, what was the reason to buy such a device? Any conceptual functionality that cannot be achieved with the proxy as the first hop in front of the (wild) clients? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

10 14

MSRP: Generation of REPORTs on the relay
by Peter Dunkley 26 Oct '12

26 Oct '12

Hello, One of my colleagues at Crocodile has been doing some testing with the msrp module. With the auth change that was made yesterday, and the updated configuration I put in the README example it seems to mostly work. However, there is one set of scenarios relating to failure REPORTs that don't seem to be possible with the current module. RFC 4976 section 6.4.1 paragraph 3 states: If the Failure-Report header is "yes" or "partial", and if there is a problem processing the SEND request or if an error response is received for that SEND request, then the relay MUST respond with an appropriate error response in a REPORT back to the original source of the message. This means that when a SEND request that contains a Failure-Report: header with value of "yes" or "partial" is relayed by Kamailio we need to maintain some state so that when the TCP send fails, or we receive a failure response from the next hop, Kamailio can generate and send a REPORT. At the moment I don't think the msrp module has these capabilities. If the module is getting some re-work before the next release it would be good to get this in there. I am quite happy to help out in any way I can, but I am not sure of the best way to proceed with this. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Kamailio on Twitter: @kamailioproject
by Olle E. Johansson 26 Oct '12

26 Oct '12

The marketing arm of the Kamailio project has without further discussion opened up a Twitter account for Kamailio, to support our web site news, G+ and Facebook pages. Feel free to contribute with case stories and news we can publish on the web site and follow up in the social media flow. The twitter account is @kamailioproject - please follow it. And yes, Kamailio is THE best thing since the invention of sliced bread! This week's EXTRAORDINARY offer: Download TWO licenses of Kamailio for the same licensing cost as we've been offering for ONE license! And get an unlimited number of user licenses for your Kamailio SIP accounts! Kamailio greetings! Have a great SIP weekend! /Marketing

1 0

MSRP relay authentication
by Peter Dunkley 25 Oct '12

25 Oct '12

Hi, MSRP AUTH requests must be authenticated using HTTP Digest authentication. Part of the concatenated data when doing this authentication is the method name. Because of the way MSRP requests are formatted, the part of the request-line that would contain the method name in HTTP or SIP requests is always MSRP. The MSRP method name (AUTH in this case) is at the end of the request-line. When Kamailio converts an MSRP request to SIP it has a method name of MSRP. This means that when Kamailio authenticates an MSRP request it uses "MSRP" as the method name, not the actual MSRP method name (AUTH). Is this correct? Should MSRP requests be authenticated with a method name of "MSRP" (which kind-of makes sense as this is the string at the start of the MSRP request line - which is where the method names are in HTTP and SIP) or should the MSRP method name of "AUTH" be used? If it is the later, does anyone know of an easy way to add this to Kamailio? Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 4

dns queries on ipv6 addresses
by Juha Heinanen 25 Oct '12

25 Oct '12

from the mailing list archives i have learned that dns_try_ipv6=yes in config file is supposed to prevent dns query on ipv6 address. if i have it "no", i get to syslog: Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: INFO: Request URI is <sip:+35832345679@[2002:c062:670a::10]:4050;transport=udp> Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: ERROR: tm [ut.h:314]: ERROR: uri2dst: failed to resolve "[2002:c062:670a::10]" :unresolvable A or AAAA request (-7) Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches and sip proxy naturally does not send anything out. if i have it "yes", request is sent out properly, but still sip proxy makes dns queries on the ipv6 address: Oct 24 14:08:35 siika /usr/sbin/sip-proxy[18196]: INFO: Request URI is <sip:+35832345679@[2002:c062:670a::10]:4050;transport=udp> Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail why is that? how to completely prevent dns queries on ipv6 addresses? -- juha

6 16

rtpproxy delete fails
by Juha Heinanen 25 Oct '12

25 Oct '12

i made rtpproxy test in setup where two sip phones have registered the same AoR test(a)test.fi. one is behind nat and the other is not. when i call this AoR, my sip proxy executes rtpproxy_manage("FROW3"); in branch route of the branch that is behind nat. syslog shows: Oct 19 15:49:59 siika /usr/sbin/sip-proxy[18596]: INFO: Calling rtpproxy_manage on INVITE <sip:test@192.168.0.31:5074;transport=tcp> Oct 19 15:49:59 siika mediaproxy-ng[12832]: Got valid command from udp:127.0.0.1:48714: 18596_16 US ptonjivixvatrhz(a)siika.tutpro.com;z9hG4bKqxklorkm 192.98.103.10 8000 wnzdf;1 Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com] Creating new call Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Opened ports 50104/50105 for RTP Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Opened ports 50106/50107 for RTP Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Returning to SIP proxy: 18596_16 50104 192.98.103.10 4 then the uas hehind nat replies with 480 and sip proxy executes in onreply route rtpproxy_manage("FROW3"); that fails and i get to syslog: Oct 19 15:50:15 siika mediaproxy-ng[12832]: Got valid command from udp:127.0.0.1:56183: 18594_12 D ptonjivixvatrhz(a)siika.tutpro.com;z9hG4bKqxklorkm wnzdf shsqn Oct 19 15:50:15 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com] Tags didn't match for delete message, ignoring looks like this mismatch is due to the missing ";1" in from tag param that was there when rtpproxy_manage was executed on invite. if my theory is correct, why is ";1" missing? if not correct, what did i do wrong? -- juha

4 17

udp transport in rtpproxy sockets
by Juha Heinanen 25 Oct '12

25 Oct '12

rtpproxy module readme gives impression that udp transport is the only possibility: 4.1. rtpproxy_sock (string) Definition of socket(s) used to connect to (a set) RTPProxy. It may specify a UNIX socket or an IPv4/IPv6 UDP socket. if that is the case, what is the justification for it, i.e, why is tcp (and better yet tls) transport not supported? -- juha

3 3

rtpproxy module flags 1 and 2 question
by Juha Heinanen 25 Oct '12

25 Oct '12

i'm new to rtpproxy (have been using mediaproxy for years) and now try to understand if it would make sense to switch over. rtpproxy readme has this on flags value 1: + 1 - append first Via branch to Call-ID when sending command to rtpproxy. ... This is especially useful if you have serially forked call scenarios where rtpproxy gets an "update" command for a new branch, and then a "delete" command for the previous branch, which would otherwise delete the full call, breaking the subsequent "lookup" for the new branch. how it would be possible in SERIAL forked scenario to get anything for a new branch before previous branch has first failed and gone? -- juha

4 7

Manually add Via Header
by Vassilis Radis 25 Oct '12

25 Oct '12

I have a situation where a far end SIP provider doesn't behave properly when sending 487 replies. The scenario is this: I have a registered user calling into my kamailio which ,using lcr module, routes the call to a SIP provider. When the caller Cancels the call, my kamailio forwards the cancel message to the provider (along with a 200 Cancelling message to the caller). The problem is with the 487 sent by the provider to my kamailio: It does not contain all the two Via headers. It only contains the via header that names my kamailio. So when kamailio gets the 487, it removes its Via header and sends the 487 back to the caller, which now doesnt find any via header. Note that the far end provider, correctly sends all the Via Headers in other replies (183 etc). So now I am trying to "intercept" the 487 from this specific provider in the onreply route, and patch it with the header that the provider should have included. How can I do it? The problem is twofold: 1.I need to detect which header I should manually add (maybe store the Via Header set from the initial invites or the 183 replies, and get it from there? .I dont know how to do that) 2.I need to add it. (How can I manually add a Via header? .This seems easier Can anyone help me, maybe with another solution?

4 5

ip address fields
by Juha Heinanen 25 Oct '12

25 Oct '12

lcr_gw table has: ip_addr varchar(47) trusted table has: src_ip varchar(50) address table has: ip_addr varchar(48) why is it that they are not all the same? also, why the common length is not 8 * 4 + 7 = 39 (http://en.wikipedia.org/wiki/IPv6#Address_format)? -- juha

4 19

Kamailio & OMA Presence Compliance
by Thomas Hoffman 24 Oct '12

24 Oct '12

My colleague and I have been trying to test Kamailio configured as a Presence Server using OMA standards/messages. It appears that Kamailio does not support the OMA standards as it pertains to Presence. This is primarily in the area of the XCAP messages. Can someone please confirm if this is the case? If this is the case, it is on the roadmap to support it? If not, we'd like to know that too and we'll continue trying to get it to work. Thank you, Tom ______________________________________________________________________________________________________________________________ This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify so to the sender by e-mail and delete the original message. In such cases, please notify us immediately at sanchar-sadhan(a)infinite.com . Further, you are not to copy, disclose, or distribute this e-mail or its contents to any unauthorized person(s) .Any such actions are considered unlawful. This e-mail may contain viruses. Infinite has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachments. Infinite reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infinite e-mail system. ***INFINITE******** End of Disclaimer********INFINITE********

1 0

RFC5626 SIP Outbound Support of Flow-Timer Header
by Jonathan Hunter 24 Oct '12

24 Oct '12

Hi, Testing with RFC5626, can you confirm that the Flow-Timer header field is supported and configurable with Kamailio 3.3 ? Thanks Jon

2 3

sercmd compiles and installs as kamcmd for kamailio
by Daniel-Constantin Mierla 24 Oct '12

24 Oct '12

Hello, short note to let everyone know that sercmd tool compiles and installs as kamcmd for flavour kamailio as of today in the master git branch. There was a conflict when trying to install both flavours from packages, because of overlapping sercmd file. I added a basic manpage for the tool, a recommendation for each binary app in many linux distros. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Registering Phones
by Re Alvarez 24 Oct '12

24 Oct '12

Hi I am setting Kamailio as SBC in my organization using Freeswitch as PBX. The integration between the two systems is working perfectly and now I am trying to add some CISCO 7970 phones to the mix. Here's a summary of the setup 1. Kamailio has 2 NICs (1 public, 1 private), RTPproxy is running 2. Freeswitch is on private network. 3. All the CISCO phones are also on the private network. We will never have a SIP phone outside our network, My question, in this case. Should I register my phones to Kamailio or Freeswitch. I am new to Kamailio and still getting down with the SIP concepts. I would prefer to register the phones with Freeswitch and only want Kamailio to get involved when calls are leaving/entering the network. If two users, registered to the same freeswitch server call each other, Kamailio should not have to do anything (ideally, not sure if its possible with SBC setup). Thanks Re

1 0

getting siptrace error on start up
by Christopher Holland 23 Oct '12

23 Oct '12

getting this error: kamailio: ERROR: <core> [modparam.c:151]: set_mod_param_regex: parameter <trace_to_database> not found in module <siptrace> parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 63, column 46: Can't set module parameter I have set this to modparam("siptrace", "trace_to_database", "0") in my .cfg /I didn't compile siptrace when I compiled kamailio. I don't know how to add a module after compiling and installing kamailio. I haven't been able to find any documentaion on how to add a module after installing only compiling before installing. Thanks all for your help Christopher /

3 3

Kamailio Hosted Solutions
by Satyajeet Thokchom 23 Oct '12

23 Oct '12

I am looking for hosted Kamailio Solutions for my VOIP calling app in android and iphone. I have the app ready and have even setup kamailio server in our rackspace cloud but there is many problems like one way calling, noisy calls, no call connection etc. (application VOIP library , we are using IMSdroid and Idoubs , based on doubango SIP) I am looking for a hosted solutions affordable for me - with a support for 300 concurrent calls at a time. Best Regards, Satyajeet Thokchom Business Manager, NGBM Solutions Ltd Phone: India:+91-11-25541320 Mobile: 91 7428565264 Website: www.ngbm.net Web Support: www.ngbmsolutions.com Note: "This e-mail is confidential and may also be privileged. If you are not the intended recipient, please notify us immediately; you should not copy or use it for any purpose, nor disclose its contents to any other person"

1 0

turning off rev. dns
by Christopher Holland 22 Oct '12

22 Oct '12

I'm setting up Homer with kamailio for sip capturing. I've got it all set up and am getting this error. I have one interface for packet capture on a mirroired port and one interface for managment. I'm thinking that I have to only have kamailio listen on the packet capture interface and not the managment interface. Would that resolve this issue? loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(1832) WARNING: <core> [socket_info.c:1406]: WARNING: fix_hostname: could not rev. resolve 192.168.111.88 Listening on udp: 127.0.0.1:5060 udp: 192.168.111.88:5060 Aliases: udp: localhost:5060 config file ok, exiting... sense it's a local address it's not possible to do a rev. resolve of that address. Is there a way to turn off the rev. resolve. I've looked at this in the trouble shooting faq. How do I avoid Reverse-DNS-Lookups? A: Simply not use src_ip = "11.22.33.44" and use instead src_ip = 11.22.33.44! We had in our installation some serious trouble with requests coming from systems, where the hostname could not be reverse resolved (the DNS-Server responsible for this IP-Range was poorly configured and would not answer at all).

3 3

Remote Commands
by Ed Brady 22 Oct '12

22 Oct '12

Hi, I am trying to get a company who handles our billing and account provisioning connected to our Kamailio instance. I understand that XMLRPC is likely the best method for doing this, but I am unable to find direct commands equivalent to "kamctl add" or "kamctl rm". Is there a resource that lists the MI or XMLRPC commands, or do I need to create even simple commands? Thanks, Ed

2 4

Hangup call on leg B side
by pars3c 22 Oct '12

22 Oct '12

Hi , i don't know how to realize this: A --- INVITE---->proxy ----INVITE----> B <--- 200 OK ---BYE----> ---INVITE ----->C <--- 200 OK proxy <--- 200 OK -------ACK ---> -------ACK ---> When B answer the call, the proxy should immediately hang up the call and make a new invite. For user A is always the same call. I tried this : onreply_route { ################################ connect if (is_method("INVITE") && status == "200") { dlg_bye("callee"); drop(); } } But it doesn't work. When arrive 200 , the proxy drop the message to the user a , wait for a while, and then send a cancel to b (it should send a BYE, instead send a CANCEL, probably because it not have a ACK). How can I try this? Thanks

2 1

Kamailio 3.1.5 crashing during TLS connection setup
by Kamal Palei 22 Oct '12

22 Oct '12

Dear All I have setup Kamailio 3.1.5 with TLS enabled. Whenever I receive the REGISTER request from endpoint, trying to forward to next proxy. For that I have added below code in kamailio.cfg * if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:5061"); exit(); } *I see once REGISTER received at Kamailio proxy, it is sending TCP synch packet to next proxy, next proxy sends back a synch+reset packet. Then kamailio proxy is crashing with log as below. *root@B2BUA:/usr/local/src/scripts# 9(2347) ERROR: <core> [tcp_main.c:4139]: connect 115.114.48.75:5061 failed 9(2347) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting * The complete log is given as below. Please let me know how to get out of this issue. I used -m 64 option while starting proxy, but same result. Thanks, Kamal, NECS, Bangalore * 0(2336) INFO: <core> [tcp_main.c:4730]: init_tcp: using epoll_et io watch method (config) 0(2338) INFO: usrloc [hslot.c:53]: locks array size 512 0(2338) INFO: tls [tls_init.c:511]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1c 10 May 2012" (0x1000103f), kerberos support: off, compression: on 0(2338) INFO: tls [tls_init.c:519]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1c 10 May 2012" (0x1000103f), kerberos support: off, zlib compression: off compiler: gcc -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -DL_ENDIAN -DTERMIO -O3 -fomit-frame-pointer -Wall -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM 0(2338) INFO: tls [tls_init.c:373]: tls: init_tls: disabling compression... 0(2338) WARNING: tls [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 5242880 and 2621440 bytes 0(2338) INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 5242880 0(2338) INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2621440 0(2338) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2338) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 0(2338) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2338) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 0(2338) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 0(2338) INFO: tls [tls_domain.c:176]: TLSs<default>: tls_method=9 0(2338) INFO: tls [tls_domain.c:188]: TLSs<default>: certificate='/etc/certs/pocserver.com/cert.pem' 0(2338) INFO: tls [tls_domain.c:195]: TLSs<default>: ca_list='(null)' 0(2338) INFO: tls [tls_domain.c:202]: TLSs<default>: crl='(null)' 0(2338) INFO: tls [tls_domain.c:206]: TLSs<default>: require_certificate=0 0(2338) INFO: tls [tls_domain.c:213]: TLSs<default>: cipher_list='(null)' 0(2338) INFO: tls [tls_domain.c:220]: TLSs<default>: private_key='/etc/certs/pocserver.com/key.pem' 0(2338) INFO: tls [tls_domain.c:224]: TLSs<default>: verify_certificate=0 0(2338) INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 0(2338) INFO: tls [tls_domain.c:544]: TLSs<default>: No client certificate required and no checks performed 0(2338) INFO: tls [tls_domain.c:176]: TLSc<default>: tls_method=9 0(2338) INFO: tls [tls_domain.c:188]: TLSc<default>: certificate='(null)' 0(2338) INFO: tls [tls_domain.c:195]: TLSc<default>: ca_list='(null)' 0(2338) INFO: tls [tls_domain.c:202]: TLSc<default>: crl='(null)' 0(2338) INFO: tls [tls_domain.c:206]: TLSc<default>: require_certificate=1 0(2338) INFO: tls [tls_domain.c:213]: TLSc<default>: cipher_list='(null)' 0(2338) INFO: tls [tls_domain.c:220]: TLSc<default>: private_key='(null)' 0(2338) INFO: tls [tls_domain.c:224]: TLSc<default>: verify_certificate=1 0(2338) INFO: tls [tls_domain.c:227]: TLSc<default>: verify_depth=9 0(2338) INFO: tls [tls_domain.c:529]: TLSc<default>: Server MUST present valid certificate 2(2340) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 1(2339) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 4(2342) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 0(2338) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 6(2344) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 3(2341) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 7(2345) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 6(2344) INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_et io watch method (config) root@B2BUA:/usr/local/src/scripts# 9(2347) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 8(2346) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled root@B2BUA:/usr/local/src/scripts# 9(2347) ERROR: <core> [tcp_main.c:4139]: connect 115.114.48.75:5061 failed 9(2347) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting 0(2338) ALERT: <core> [main.c:742]: child process 2347 exited by a signal 6 *

2 2

Listen on all IPv6 addresses
by Anton Kvashenkin 22 Oct '12

22 Oct '12

Hello guys. In the Kamailio v.3.3.0 release notes http://www.kamailio.org/w/kamailio-v3-3-0-release-notes/ it says: all IPv6 network interfaces are auto-detected and Kamailio start listening on them (if no strict listening rules are set) But it doesn't happen. version: kamailio 3.3.1 (x86_64/linux) d75cb5 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d75cb5 listen is commented out @ cfg

2 1

Re: [SR-Users] rtpproxy delete fails
by Juha Heinanen 22 Oct '12

22 Oct '12

the patch below adds 't' flag to rtpproxy_destroy flags. if 't' flag is present when rtpproxy_destroy is called, to tag is not included in D command sent to rtpproxy thus causing destroy of the full call. 't' flag can be used to avoid piling up of unused rtpproxy calls by making it possible to call rtpproxy_destroy when 200 ok is received from a branch where rtpproxy is not used. is it ok for me to commit this patch or do long term rtpproxy users have a better idea? it would still make sense to include a timer in rtpproxy that cleans up un-used calls at an interval that corresponds to sip proxy's invite timeout. -- juha *** /usr/src/orig/sip-router/modules/rtpproxy/rtpproxy.c 2012-10-19 14:31:27.658485534 +0300 --- modules/rtpproxy/rtpproxy.c 2012-10-20 09:52:10.308389743 +0300 *************** *** 1678,1683 **** --- 1678,1684 ---- str callid, from_tag, to_tag, viabranch; char *cp; int via = 0; + int to = 1; int ret; struct rtpp_node *node; struct iovec v[1 + 4 + 3 + 2] = {{NULL, 0}, {"D", 1}, {" ", 1}, {NULL, 0}, {NULL, 0}, {NULL, 0}, {" ", 1}, {NULL, 0}, {" ", 1}, {NULL, 0}}; *************** *** 1700,1705 **** --- 1701,1710 ---- via = 1; break; + case 't': + case 'T': + to = 0; + break; case 'a': case 'A': case 'i': *************** *** 1736,1742 **** return -1; } to_tag.s = 0; ! if (get_to_tag(msg, &to_tag) == -1) { LM_ERR("can't get To tag\n"); return -1; } --- 1741,1748 ---- return -1; } to_tag.s = 0; ! to_tag.len = 0; ! if ((to == 1) && get_to_tag(msg, &to_tag) == -1) { LM_ERR("can't get To tag\n"); return -1; }

2 1

mediaproxy-ng fails to parse start_recording() command
by Juha Heinanen 22 Oct '12

22 Oct '12

i tried what happens when i call start_recording() after calling rtpproxy_manage() and got this kind of error to syslog: Oct 20 19:57:43 siika mediaproxy-ng[8423]: Failed to properly parse UDP command line '8554_13 R ngsgznnwgzpuovt(a)siika.tutpro.com mgpwe' from 127.0.0.1:51497, using fallback RE why the error? it is so that mediaproxy-ng does not support recording? -- juha

4 3

Re: [SR-Users] dangers of excluding 100rel?
by Alex Balashov 19 Oct '12

19 Oct '12

You can strip 100rel as long as it's Supported. However, not so good things happen if you strip it from Required. :-) -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Andrew Mortensen <admorten(a)isc.upenn.edu> wrote:Can I safely strip 100rel support from INVITEs sent by callers registered to my proxy? We're using proxy-generated 302 redirects for call forwarding in our deployment. In general this works better for us than appending branches, as we rapidly ran into branch limits when dealing with large multiple appearance lines. However, we've run into a case where the caller, a Polycom 550, seems to get confused when a parallel forked call is redirected, 100rel is in use and the redirection target is also a multiple appearance line. The caller is correctly responding with PRACKs to all 180s, before and after the redirection. However, hanging up the call on the caller after the redirection, but before any remote party has answered, does *not* cause the caller to send a CANCEL to tear down the call. In fact, the caller doesn't send anything at all. The remote parties continue to ring until the ring timer expires, and the proxy sends CANCELs to all callee endpoints. That's clearly a Polycom bug, and we'll need to work with them to get it resolved. In the meantime, I'm looking for workarounds. One quick, crude option is to strip 100rel from the Supported header in the caller's INVITE. I've tested this, and it does solve the problem, but I'm concerned about side effects. I'm aware of the value 100rel has for calls to the PSTN, but in this case the calls never leave the local network. Are there situations where 100rel is critical for a parallel forked call? andrew _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

dangers of excluding 100rel?
by Andrew Mortensen 19 Oct '12

19 Oct '12

Can I safely strip 100rel support from INVITEs sent by callers registered to my proxy? We're using proxy-generated 302 redirects for call forwarding in our deployment. In general this works better for us than appending branches, as we rapidly ran into branch limits when dealing with large multiple appearance lines. However, we've run into a case where the caller, a Polycom 550, seems to get confused when a parallel forked call is redirected, 100rel is in use and the redirection target is also a multiple appearance line. The caller is correctly responding with PRACKs to all 180s, before and after the redirection. However, hanging up the call on the caller after the redirection, but before any remote party has answered, does *not* cause the caller to send a CANCEL to tear down the call. In fact, the caller doesn't send anything at all. The remote parties continue to ring until the ring timer expires, and the proxy sends CANCELs to all callee endpoints. That's clearly a Polycom bug, and we'll need to work with them to get it resolved. In the meantime, I'm looking for workarounds. One quick, crude option is to strip 100rel from the Supported header in the caller's INVITE. I've tested this, and it does solve the problem, but I'm concerned about side effects. I'm aware of the value 100rel has for calls to the PSTN, but in this case the calls never leave the local network. Are there situations where 100rel is critical for a parallel forked call? andrew

1 0

rtpproxy_destroy([flags]) does not accept flags
by Juha Heinanen 19 Oct '12

19 Oct '12

according to rtpproxy module readme, rtpproxy_destroy function takes optional flags param: 5.4. rtpproxy_destroy([flags]) however, the code does not support it: {"unforce_rtp_proxy", (cmd_function)unforce_rtp_proxy_f, 0, 0, 0, ANY_ROUTE}, {"rtpproxy_destroy", (cmd_function)unforce_rtp_proxy_f, 0, 0, 0, ANY_ROUTE}, what has happened? -- juha

2 3

MSILO Configuration
by Ed Brady 19 Oct '12

19 Oct '12

Hi, I am having trouble configuring MSILO. I copied the example config into my kamailio.cfg file and edited the areas that apply to my domain, but when restarting Kamailio I am faced with an error. Below is the error followed by my complete config file. Please advise what I am doing wrong. Is there some sort of resource for HowTos on Kamailio? The module documentation seems very thorough but the examples at the end seem to only be useful if that module is the only one you need to configure. I am trying to do a fairly straight forward configuration I think (SIP messaging with offline message storing and a single incoming/outgoing gateway that is used for all accounts that do not locally exist), but I am unable to find straight forward documentation on how to configure something like this. Please let me know what I am missing. Thanks, Ed Not starting Kamailio: invalid configuration file! > -e > 0(30863) INFO: tls [tls_init.c:377]: tls: init_tls: disabling > compression... > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: syntax error > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: bad command > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: bad command > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 878, column 12: syntax error > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 878, column 12: > ERROR: bad config file (5 errors) > *Config* #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_TLS #!define WITH_NAT #!define WITH_ACCDB # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:R4U8oO8Er872sNHr@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "cfgutils.so" loadmodule "msrp.so" loadmodule "htable.so" loadmodule "msilo.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # -- msilo params -- modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@MYDOMAIN.com") modparam("msilo","contact_hdr","Contact: registrar(a)MYDOMAIN.com :5060;msilo=yes\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","*** User $rU is offline!") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=1800;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no 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 exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # Route to MSILO 10-18-12 route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay(); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } event_route[msrp:frame-in] { xdbg("============#[[$msrp(method)]]===========\n"); xdbg("============*[[$si:$sp]]\n"); xdbg("============ crthop: [$msrp(crthop)]\n"); xdbg("============ prevhop: [$msrp(prevhop)]\n"); xdbg("============ nexthop: [$msrp(nexthop)]\n"); xdbg("============ firsthop: [$msrp(firsthop)]\n"); xdbg("============ lasthop: [$msrp(lasthop)]\n"); xdbg("============ prevhops: [$msrp(prevhops)]\n"); xdbg("============ nexthops: [$msrp(nexthops)]\n"); xdbg("============ srcaddr: [$msrp(srcaddr)]\n"); xdbg("============ srcsock: [$msrp(srcsock)]\n"); xdbg("============ sessid: [$msrp(sessid)]\n"); msrp_reply_flags("1"); if(msrp_is_reply()) { msrp_relay(); exit; } # handle AUTH MSRP requests if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } # frame for local server - send Use-Path # -- passwd can be loaded from DB based on $au $var(passwd) = "xyz123"; if(!pv_www_authenticate("myrealm", "$var(passwd)", "0")) { if(auth_get_www_authenticate("myrealm", "0", "$var(wauth)")) { msrp_reply("401", "Authorization Required", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } $var(cnt) = $var(cnt) + 1; pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)"); $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr); $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock); # - Use-Path: the MSRP address for server + session id $var(UsePath) = "Use-Path: msrp://127.0.0.1:5060/" + $var(sessid) + ";tcp\r\n"; msrp_reply("200", "OK", "$var(UsePath)"); exit; } if($msrp(method)=="SEND") { if($msrp(nexthops)>1) { msrp_reply("200", "Received"); msrp_relay(); exit; } $var(sessid) = $msrp(sessid); if($sht(msrp=>$var(sessid)::srcaddr) == $null) { # one more hop, but we don't have address in htable msrp_reply("481", "No Such Session"); exit; } msrp_relay_flags("1"); msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)", "$sht(msrp=>$var(sessid)::srcsock)"); msrp_relay(); exit; } msrp_relay(); }

3 2

Re: [SR-Users] Kamailio v3.3.2 Released
by Paul Cupis 18 Oct '12

18 Oct '12

On 16/10/12 22:29, Jon Bonilla (Manwe) wrote: > El Tue, 16 Oct 2012 17:00:03 +0200 > Daniel-Constantin Mierla<miconda(a)gmail.com> escribió: >> >> Kamailio SIP Server v3.3.2 stable release is out. > > Debian and Ubuntu packages are ready. Please find the instructions > below: > > http://www.kamailio.org/dokuwiki/doku.php/packages:debs I'm seeing a problem with the 3.3.2 packages on Debian at the moment. There is a file conflict between the kamailio package and kamailio-mysql-modules preventing the latter from installing. $ dpkg-deb -c /var/cache/apt/archives/kamailio_3.3.2+squeeze1_amd64.deb | grep srdb -rw-r--r-- root/root 74296 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1.0 -rw-r--r-- root/root 28360 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1 -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1 -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so -> libsrdb2.so.1.0 $ dpkg-deb -c /var/cache/apt/archives/kamailio-mysql-modules_3.3.2+squeeze1_amd64.deb | grep srdb -rw-r--r-- root/root 28360 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1.0 -rw-r--r-- root/root 74296 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1 -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1 -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so -> libsrdb1.so.1.0 Regards,

2 3

Kamailio- Redirect message 300 with LCR reroute
by Fatima Chahrour~Vanrise Support 18 Oct '12

18 Oct '12

Dear List, I'd like to get help regarding my case. I have the following script where many thanks to Daniel has helped me in if (is_method("INVITE")) { if (!load_gws(1, $rU, $fu)) { sl_send_reply("502", "Unable To lOad GatEwAyS"); exit;} if(!next_gw()){ sl_send_reply("503", "Unable To fInD a gateWaY"); exit;} while(next_gw()){ km_append_branch(); } sl_send_reply("302","Moved Temporary"); exit; } The problem am facing is that the call is rerouting from the first gateway to the next gateway successfully when I have two gateways. BUT when I have three gateways, and the first two gateways are off, the call is not rerouted to the third gateway. It keeps hitting the seconds gateway and gives request time out at the end Why is that? Thanks in advance, F Chahrour

1 0

Kamailio v3.3.2 RPM Packages
by Daniel-Constantin Mierla 18 Oct '12

18 Oct '12

Hello, for the record, RPM packages of Kamailio v3.3.2 were built for Centos 5.5, Centos 6, RedHat 5, RedHat 6, Fedora 15, Fedora 16, Fedora 17, OpenSuse 11.4 and OpenSuse 12.1, on i586 and x86_64 architectures. They are available on opensuse build service repos, see the links at: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Question about compliance with RCS/OMA Simple Presence
by Sangeeta Shah 17 Oct '12

17 Oct '12

Hello All, I need some guidance on whether the Kamailio presence/presence_xml and integrated xcap module implementations are RCS/OMA Simple presence compliant. Based on a few things I've seen in the code I don't see any handling for the oma presence rule xml extensions. Is there any plan to be RCS/OMA compliant in the new future? I can get SIMPLE PRESENCE with XCAP to work but not with OMA presence rules and resource lists. Thanks, Sangeeta -- Sangeeta Shah

1 0

has_sdp()??
by Juha Heinanen 17 Oct '12

17 Oct '12

while reading rtpproxy module readme, i found this kind of example: if (is_method("INVITE")) { if (has_sdp()) { if (rtpproxy_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } if (is_method("ACK") && has_sdp()) rtpproxy_answer(); does has_sdp() function really exist? it is not listed on wiki page: http://www.kamailhttp://www.kamailio.org/wiki/alphaindexes/3.4.x/modfunctio… -- juha

4 13

make option to clean the installed binaries
by Ajay Sabat 17 Oct '12

17 Oct '12

Hi, I am a newbie here and recently started using kamailio/ser. I am wondering is there a makefile option to cleanup the installed binaries and library files. Thanks, Ajay

2 3

how to search previous messages in the forum
by Ajay Sabat 17 Oct '12

17 Oct '12

Hi, Many times I get doubts and questions about Kamailio that I want to search in the SR-Users forum before posting a new message, as it might have already been answered in the past. Is there a easier way to do that than looking up the subject lines in the monthly archive? Thanks, Ajay

2 1

Errors issued by Kamailio
by Sangeeta Shah 17 Oct '12

17 Oct '12

Hi, I am seeing the following errors in syslog: Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence_xml [notify_body.c:635]: while extracting tuple node Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence_xml [notify_body.c:75]: while constructing offline body Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence [presentity.c:1220]: preparing body Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence [publish.c:171]: Marking presentity Oct 8 16:21:55 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event message-summary Oct 8 16:21:55 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event presence.winfo Oct 8 16:21:57 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event presence I am using Kamailio 3.3 with Presence and RLS with integrated XCAP. The only significant change i made today is update the presence and RLS config to DB ONLY mode. Thanks, Sangeeta -- Sangeeta Shah

2 9

$fs syntax
by Juha Heinanen 17 Oct '12

17 Oct '12

wiki tells: $fs - reference to the forced socket for message sending (if any) in the form proto:ip:port should it be [proto:]ip:port instead? -- juha

3 7

Kamailio 3.2.4 presence
by Vitaliy Aleksandrov 17 Oct '12

17 Oct '12

Hi all, I'm facing problems with active subscription restoration after sipproxy restart. db_mode is 0 (DB_MEMORY_ONLY) and according to comments in the presence.h kamailio holds subscription in memory and periodically updates to db, but retrieves from db only at startup. And it's true, I see that kamailio tries to update the active_watchers table, but it never inserts new subscriptions there. There are no problems with access to the database since when db_mode is 2 (DB_ONLY) kamailio inserts new records into the active_watchers without any problems. Here are presence module settings: modparam("presence", "db_url", "sqlite:////var/db/presence.db") modparam("presence", "db_mode", 0) modparam("presence", "timeout_rm_subs", 1) modparam("presence", "pres_htable_size", 8) modparam("presence", "subs_htable_size", 8) modparam("presence", "max_expires", 3600) modparam("presence", "db_update_period", 60) modparam("presence", "clean_period", 60) Does anybody have success with db_mode = DB_MEMORY_ONLY and active_watchers db table ?

2 2

is kamilio support RFC5626 now ?
by mike 16 Oct '12

16 Oct '12

dear kamilio developers : i read the kamilio 3.3.0' s new features , it say it support outbound , and i have place to handle reg-id and sip.instance . so , i wonder if the kamilio3.3.0 support RFC5626 now，so that the kamilio can be used as a edge proxy . i ask why because i can't find where kamilio add flow-token and path for the register request. thanks

5 8

IPv6 addresses in dispatcher on IPv4 only proxy
by Alex Hermann 16 Oct '12

16 Oct '12

Hello, I am using the dispatcher module with a database table shared among multiple proxies. Some proxies do both IPv4 and IPv6, others only do IPv4. The problem is when i use an IPv6 address in the dispatcher table. Then the IPv4-only proxies fail to load the table. ERROR: dispatcher [dispatch.c:325]: could not resolve [IPv6 address] WARNING: <core> [mem/f_malloc.c:474]: WARNING:fm_free: free(0) called ERROR: dispatcher [dispatcher.c:321]: could not initiate a connect to the database ERROR: <core> [sr_module.c:889]: init_mod(): Error while initializing module dispatcher (/usr/lib/kamailio/modules_k/dispatcher.so) Is there any setting i can use on the IPv4-only proxies to make it skip the IPv6 records? (apart from listening on an IPv6 address) -- Greetings, Alex Hermann

2 9

Error while running TLS enabled Kamailio proxy
by Kamal Palei 16 Oct '12

16 Oct '12

Dear All While executing TLS enabled Kamailio proxy (3.1.5), getting below error. * 0(15895) INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 0(15895) ERROR: tls [tls_domain.c:393]: TLSs<default>: Unable to load certificate file './modules/tls/ser-selfsigned.pem' 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:02001002:system library:fopen:No such file or directory 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:20074002:BIO routines:FILE_CTRL:system lib 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:140DC002:SSL routines:SSL_CTX_use_certificate_chain_file:system lib 0(15895) ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module tls (/usr/local/lib/kamailio/modules/tls.so) 0(15895) ERROR: <core> [main.c:1532]: ERROR: main: error in init_child(PROC_INT) -- exiting 0(15895) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting 0(15893) ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe * I performed below steps to compile and install. 1. I had a working Kamailio-3.1.5 code base without TLS 2. Downloaded latest openssl source code 3. Compiled and generated libssl.a and libcrypto.a 4. Next, I recompiled kamailio proxy, compilation did not go through. I had to modify Makefile in modules/tls module, added path for openssl/bio.h and path for libssl.a 5. Then executed make all clean and make all include_modules=tls , this time compilation succeeded 6. Next I did make install 7. Generated certificates using openssl library that comes by default with ubuntu 10.04 lte 8. Added below entries in kamailio.cfg #!define WITH_TLS #!ifdef WITH_TLS loadmodule "tls.so" modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") modparam("tls", "private_key", "/etc/certs/localb2bua.com/key.pem") modparam("tls", "certificate", "/etc/certs/localb2bua.com/cert.pem") modparam("tls", "ca_list", "/etc/certs/demoCA/cert.pem") enable_tls=yes #!endif 9. Then I tried running kamailio proxy It complained not getting tls.so and tls.cfg Copied these files from modules/tls to /usr/local/etc/kamailio/ path 10. Then I was able to run Kamailio proxy and got the errors pasted at top of email (* Unable to load certificate file './modules/tls/ser-selfsigned.pem') * Please guide me what should I do to avoid this error. Thanks a lot. -Kamal * *

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unsubscribe
by Pablo R. Digonzelli 16 Oct '12

16 Oct '12

Ing. Pablo Digonzelli Sofware Solutions IP Soluciones SRL 25 de Mayo 521. Entrepiso A email: pdigonzelli(a)softsargentina.com email: pdigonzelli(a)gmail.com twitter: @pdigonzelli Tel: 0381 4227378 Cel: 0381 155982714

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Re: [SR-Users] Kamailio 3.3.0 CDR logging call "duration"
by Daniel-Constantin Mierla 16 Oct '12

16 Oct '12

Hello, do not reply to private email only when it is about a discussion on the mailing list - such messages will be ignored. Try to store the value in an avp when processing the BYE, after loose_route(), and store that avp in acc. Daniel On 10/15/12 1:13 PM, Nord7 wrote: > Hello, > no I'm not getting error messages. > Problem: > I can not get kamailio record duration dialogue created field "duration". > I make a test call, answer it, wait 5-6 seconds and hang up. > Now I have to get 5-6 seconds duration in the "duration", but the > field is blank. The rest of the field information about the call is > entered correctly. > So the variable $ DLG_lifetime information about the duration of the > call does not get. > But most of all I am wrong somewhere. > > > 2012/10/15 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > do you get any error message or what is the problem you encounter? > > Cheers, > Daniel > > > On 10/15/12 10:14 AM, Nord7 wrote: >> Hello >> >> Kamailio 3.3.0 >> All default "extra fields" created in database. I create in >> Database new field "duration" and trying write in this field >> variable $DLG_lifetime (from dialog module) , but can't do this. >> What i'm doing wrong? >> >> installed modules: ACC, DIALOG >> # ----- acc params ----- >> /* what special events should be accounted ? */ >> modparam("acc", "early_media", 0) >> modparam("acc", "report_ack", 0) >> modparam("acc", "report_cancels", 0) >> modparam("acc", "detect_direction", 0) >> /* account triggers (flags) */ >> modparam("acc", "log_level", 1) >> modparam("acc", "log_flag", FLT_ACC) >> modparam("acc", "log_missed_flag", FLT_ACCMISSED) >> modparam("acc", "log_extra", >> "src_user=$fU;src_domain=$fd;src_ip=$si;" >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" >> "duration=$DLG_lifetime") >> modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) >> /* enhanced DB accounting */ >> modparam("acc", "db_flag", FLT_ACC) >> modparam("acc", "db_missed_flag", FLT_ACCMISSED) >> modparam("acc", "db_url", DBURL) >> modparam("acc", "db_extra", >> "src_user=$fU;src_domain=$fd;src_ip=$si;" /* extra fields >> created in DataBase */ >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" >> "duration=$DLG_lifetime") >> modparam("acc", "cdr_enable", 1) >> modparam("acc", "cdr_start_on_confirmed", 0) >> modparam("acc", "cdr_facility", "LOG_DAEMON") >> >> # ----- dialog params ----- >> modparam("dialog", "enable_stats", 1) >> modparam("dialog", "hash_size", 4096) >> modparam("dialog", "rr_param", "did") >> modparam("dialog", "dlg_flag", DLG_FLAG) /* DLG_FLAG=4 */ >> modparam("dialog", "default_timeout", 21600) >> modparam("dialog", "timeout_avp", "$avp(i:10)") >> modparam("dialog", "dlg_match_mode", 0) >> modparam("dialog", "detect_spirals", 1) >> modparam("dialog", "db_url", DBURL) >> modparam("dialog", "db_mode", 1) >> modparam("dialog", "db_fetch_rows", 500) >> >> >> ########Routing Logic######## >> >> route[RELAY] { >> >> # enable additional event routes for forwarded requests >> # - serial forking, RTP relaying handling, a.s.o. >> if (is_method("INVITE|SUBSCRIBE")) { >> t_on_branch("MANAGE_BRANCH"); >> t_on_reply("MANAGE_REPLY"); >> } >> if (is_method("INVITE")) { >> t_on_failure("MANAGE_FAILURE"); >> } >> setflag(DLG_FLAG); # flag 4 duration >> setflag(FLT_ACC); # do accounting ... >> setflag(FLT_ACCFAILED); # ... even if the transaction fails >> if (!t_relay()) { >> sl_reply_error(); >> } >> exit; >> } >> >> >> >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -http://asipto.com/u/katu > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

segfault in siptrace module using setflag?
by samuel 16 Oct '12

16 Oct '12

Hi folks, I was making some tests with kamailio (both 3.3.0 and 3.3.1 tgz) with siptrace module sending "homer" packets to a remote central agent where the data is stored and presented to the user. The related config is: # ----- siptrace params from HOMER ----- modparam("siptrace", "duplicate_uri", "sip:A.B.C.D:9060") modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #HEPv2 == timestamp will be included to HEP header modparam("siptrace", "hep_version", 2) modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_capture_id", 1) basically for all incoming messages, a route is called with the following actions: route[HOMER] { setflag(22); sip_trace(); } I've been making some tests and if I disable the setflag action, there is no problem (well, no transaction messages are sent to the central agent), but as soon as I activated setflag, kamailio crashes with the following error: <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 28 There's no concrete point in the config file where it fails so I think the problem is more tm-internals related. I can provide further information in case you need it. Just for adding information, and without any intention to start a flame-war, with opensips 1.8.0-tls the server does not crash... Thanks in advance! Samuel.

3 14

Supporting TLS and DTLS in RTP Proxy
by Kamal Palei 16 Oct '12

16 Oct '12

Hi All I am planning to enhance RTP proxy to support TLS and DTLS. We have some requirements where we need to send RTP packets either over TLS or over DTLS. I really do not know where to start :-) Few things that I have thought over is 1. I will take opnssl code base and integrate with rtpproxy code. 2. Will refer kamailio proxy code how tls connection framework is there and try to copy the code from kamailio code base to rtp proxy code base, OR is there a better alternative here. If somebody has a better idea, kindly share. Thanks a lot. -kamal

5 6

Guidance on using Kamailio as Presence Server and Scalability issues for millions of subsribers
by Thomas Hoffman 16 Oct '12

16 Oct '12

We're trying to determine how best to design/architect a solution where we use Kamailio as a Presence/XCAP server and be scalable to support millions of subscribers. I see the Dispatcher module, but it appears that requires having dedicated instances of the server running just in Dispatcher mode - then you can load balance across the actual instances of the presence servers. It's not clear if a server can be configured to work as a dispatcher and a presence server at the same time. Does anyone have any high-level guidance how to approach this problem? Thank you, Tom Hoffman ______________________________________________________________________________________________________________________________ This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify so to the sender by e-mail and delete the original message. In such cases, please notify us immediately at sanchar-sadhan(a)infinite.com . Further, you are not to copy, disclose, or distribute this e-mail or its contents to any unauthorized person(s) .Any such actions are considered unlawful. This e-mail may contain viruses. Infinite has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachments. Infinite reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infinite e-mail system. ***INFINITE******** End of Disclaimer********INFINITE********

2 1

ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1)
by MARINA SERRANO MONTES 16 Oct '12

16 Oct '12

Hi, My kamailio installation usually works right, but sometimes, it crashes and it shows the following error: "ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) " Do you know why it happens randomly or what's the cause ? Many thanks. Br, Marina ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at: http://www.tid.es/ES/PAGINAS/disclaimer.aspx

2 1

preparing kamailio 3.3.2
by Daniel-Constantin Mierla 16 Oct '12

16 Oct '12

Hello, I am considering packaging Kamailio v3.3.2 at the beginning of next week (Monday or Tuesday). If any of the devs have backports to be done, do them before the end of this week. Also, if someone spotted some fixes in the master that were not backported, write to the lists so we can care of them. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 2

Delay message
by pars3c 15 Oct '12

15 Oct '12

Hi, how can i delay sending of a message ? A ----> proxy (after x seconds) ---> B If I use the sleep function of the cfgutils module,I block all the calls for the sleep time or the single one? Thanks

3 4

Error on compiling Kamailio with TLS support in Ubuntu 10.04LTE
by Kamal Palei 15 Oct '12

15 Oct '12

Hi All I am trying to compile Kamailio - 3.1.5 in Ubuntu 10.04LTE . Tried below comands one by one make -C modules/tls make modules modules=modules/tls make all include_modules=tls But always getting the error *In file included from tls_bio.c:31: tls_bio.h:32:25: warning: openssl/bio.h: No such file or directory * It looks that openssl is not installed in machine. I tried installing openssl as belowB sudo apt-get install libssl-dev But it fails always. The error it gets is *root@B2BUA:/usr/local/src/kamailio-3.1.5# sudo apt-get install libssl-dev sudo: unable to resolve host B2BUA Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: libssl-dev: Depends: libssl0.9.8 (= 0.9.8k-7ubuntu8) but 0.9.8k-7ubuntu8.8 is to be installed E: Broken packages* * root@B2BUA:/usr/local/src/kamailio-3.1.5# * Has anybody faced this issue previously. Whats the workaround here, if somebody knows please share. Thanks Kamal

2 1

Re: [SR-Users] R: Delay message
by Alex Balashov 15 Oct '12

15 Oct '12

You can't, exactly. But you can t_continue() inside an rtimer route that fires at a given interval. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/pars3c <pars3c(a)gmail.com> wrote:With t_suspend of tmx module, how can I set seconds to sleep? -----Messaggio originale----- Da: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] Per conto di Alex Balashov Inviato: lunedì 15 ottobre 2012 15:52 A: sr-users(a)lists.sip-router.org Oggetto: Re: [SR-Users] Delay message It sounds like you're after the recent asynchronous TM innovations: http://www.kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2535968 http://www.kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2525882 On 10/15/2012 09:49 AM, pars3c wrote: > Hi, > > how can i delay sending of a message ? > > A ----> proxy (after x seconds) ---> B > > If I use the sleep function of the cfgutils module,I block all the > calls for the sleep time or the single one? > > Thanks > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ----- Nessun virus nel messaggio. Controllato da AVG - www.avg.com Versione: 2012.0.2221 / Database dei virus: 2441/5332 - Data di rilascio: 14/10/2012 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Kamailio 3.3.0 CDR logging call "duration"
by Nord7 15 Oct '12

15 Oct '12

Hello Kamailio 3.3.0 All default "extra fields" created in database. I create in Database new field "duration" and trying write in this field variable $DLG_lifetime (from dialog module) , but can't do this. What i'm doing wrong? installed modules: ACC, DIALOG # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_level", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "duration=$DLG_lifetime") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" /* extra fields created in DataBase */ "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "duration=$DLG_lifetime") modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 0) modparam("acc", "cdr_facility", "LOG_DAEMON") # ----- dialog params ----- modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag", DLG_FLAG) /* DLG_FLAG=4 */ modparam("dialog", "default_timeout", 21600) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 0) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "db_fetch_rows", 500) ########Routing Logic######## route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } setflag(DLG_FLAG); # flag 4 duration setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails if (!t_relay()) { sl_reply_error(); } exit; }

2 1

DTLS support in Kamailio
by Kamal Palei 15 Oct '12

15 Oct '12

Hi List Is DTLS supported in Kamailio? If not, can somebody give me initial idea/pointer how can we support it with minimal code change. Best Regards Kamal

2 1

Re: [SR-Users] Get sip clients to work with sip:user@domain and ldap
by Christophe ROY 15 Oct '12

15 Oct '12

Hello I have created a SRV record for sip.domain.tld pointing to our kamailio server, the domain by itself resolves to our DNS IP. Regarding the sockets: root@kamailio1:~# netstat -lnp | grep kamailio tcp 0 0 192.168.14.25:5060 0.0.0.0:* LISTEN 4752/kamailio unix 2 [ ACC ] STREAM LISTENING 17100 4745/kamailio /tmp/kamailio_ctl Thanks 2012/10/11 Daniel-Constantin Mierla <miconda(a)gmail.com> > Hello, > > I recommend using 3.3.x if you start a new deployment. > > Is the domain resolvable by DNS? What are the sockets kamailio is > listening on? > > Cheers, > Daniel > > On 10/10/12 5:21 PM, Christophe ROY wrote: > > Hello > > I'm quite new to SIP and TOIP so I may not understand the concepts > correctly. > I'm supposed to make work SIP clients (for now, linphone on android and > windows) with a Kamailio 3.2 server and ldap authentication. > > Ldap authentication seems to be working fine, as the login/pass is > validated against the Active Directory before registering the SIP client. > However, Linphone seems to register the SIP client as > sip:user@true_ip_address instead of sip:user@domain. > > If I try to make a call to sip:anotheruser@domain it doesn't work, and I > can find in /var/log/syslog the following lines: > > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [ut.h:333]: > no corresponding socket for af 2 > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm > [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding > listening socket) > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm > [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: sl > [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server > error occurred (7/SL) > > However, if I try to make a direct call to sip:anotheruser@his_true_IP, > it works > > How can I make the registered users to be reachable with user@domain with > Kamailio? > And by the way, is there a way to get a list of registered users? > > Thank you for the help > > Christophe > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

3 4

Wesip Netbeans
by pars3c 15 Oct '12

15 Oct '12

Hi , anyone know if it is possible to use Wesip with Glassfish instead of Tomcat? I'm new in using Glassfish and sip servlet, and I don't know how to start the sip servlet. In tomcat there is the file server.xml, and the connector . How "say" to glassfish to start a listener on sip packet ? If it's not possible to use Glassfish, how i can open a sar project with netbeans e debug it? Thanks to all

1 0

update_subs_rlsdb: NO MATCH
by Sangeeta Shah 12 Oct '12

12 Oct '12

I have Kamailio 3.3 with Presence, RLS and integrated XCAP. I see this information message in the log file: Oct 8 14:34:05 RCS-Presence /usr/local/sbin/kamailio[4530]: INFO: rls [rls_db.c:779]: update_subs_rlsdb: NO MATCH Anyone know why the application is calling that method if it doesn't exist in rls_db.c. I still haven't been able to get presence with rls and integrated xcap working so not sure if this is an issue or no. Currently for both presence and RLS i am using "DB-Only" mode. Any help is appreciated. It seems like when client A adds client B as a contact, the kamalio server is not sending out an auth request to client B. Even though it's registered for presence.winfo events. Thanks, Sangeeta -- Sangeeta Shah

2 6

TLS support in Kamailio
by Kamal Palei 12 Oct '12

12 Oct '12

Hi List I have a specific requiremnt where I want Kamailio proxy need to work as a intermideiate proxy . The endpoints will talk to Kamailio using UDP as transport and Kamailio should interact with main proxy using TLS. Is this scenario possible with Kamailio ? If so, can you give me some pointer how can I acheive it. Thanks Kamal Net clooud systems Bangalore

3 5

Kamailio Group voice Call
by Vijay Thakur 12 Oct '12

12 Oct '12

All Experts, I have setup kamailio 3.1.5 server and it is working fine in my production environment. I want to implement group voice calls (audio conferencing). How can i do this. Kindly guide me. With Regards, Vijay Thakur

3 2

Skype-like service in less than one hour
by శ్రీధర్ 12 Oct '12

12 Oct '12

I set up my server following the tutorial at http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour, with the latest kamailio 3.3, and updated the config file accordingly. After the reboot the sip server is not running here are the logs. When I do 'sudo /etc/init.d/kamailio start' the server starts, but has problems communicating with the rtpproxy, the log for the same is also below. Please advice if any changes needed to make this working with the latest version Thanks S $ cat /var/log/syslog | grep kamailio Oct 12 01:07:48 sips kamailio: INFO: tls [tls_init.c:377]: tls: init_tls: disabling compression... Oct 12 01:07:48 sips kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: usrloc [hslot.c:53]: locks array size 512 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: tls [tls_init.c:541]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, compression: on Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: tls [tls_init.c:549]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, zlib compression: on#012 compiler: cc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DL_ENDIAN -DTERMIO -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Wformat-security -Werror=format-security -D_FORTIFY_SOURCE=2 -Wl,-Bsymbolic-functions -Wl,-z,relro -Wa,--noexecstack -Wall -DOPENSSL_NO_TLS1_2_CLIENT -DOPENSSL_MAX_TLS1_2_CIPHER_LENGTH=50 -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: WARNING: tls [tls_init.c:603]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [db.c:320]: could not add connection to the pool Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: usrloc [dlist.c:513]: failed to open database connection Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: registrar [reg_mod.c:472]: failed to register domain Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:623 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:626 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:628 Oct 12 01:54:55 sips kamailio: INFO: tls [tls_init.c:377]: tls: init_tls: disabling compression... Oct 12 01:54:55 sips kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: usrloc [hslot.c:53]: locks array size 512 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_init.c:541]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, compression: on Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_init.c:549]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, zlib compression: on#012 compiler: cc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DL_ENDIAN -DTERMIO -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Wformat-security -Werror=format-security -D_FORTIFY_SOURCE=2 -Wl,-Bsymbolic-functions -Wl,-z,relro -Wa,--noexecstack -Wall -DOPENSSL_NO_TLS1_2_CLIENT -DOPENSSL_MAX_TLS1_2_CIPHER_LENGTH=50 -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: tls [tls_init.c:603]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:187]: TLSs<default>: tls_method=12 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:199]: TLSs<default>: certificate='/etc/kamailio/kamailio-selfsigned.pem' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:206]: TLSs<default>: ca_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:213]: TLSs<default>: crl='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:217]: TLSs<default>: require_certificate=0 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:224]: TLSs<default>: cipher_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:231]: TLSs<default>: private_key='/etc/kamailio/kamailio-selfsigned.key' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:235]: TLSs<default>: verify_certificate=0 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:238]: TLSs<default>: verify_depth=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:576]: TLSs<default>: No client certificate required and no checks performed Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:187]: TLSc<default>: tls_method=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:199]: TLSc<default>: certificate='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:206]: TLSc<default>: ca_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:213]: TLSc<default>: crl='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:217]: TLSc<default>: require_certificate=1 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:224]: TLSc<default>: cipher_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:231]: TLSc<default>: private_key='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:235]: TLSc<default>: verify_certificate=1 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:238]: TLSc<default>: verify_depth=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:561]: TLSc<default>: Server MUST present valid certificate Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily

4 6

log alerting time, progress time connect time
by pars3c 12 Oct '12

12 Oct '12

Hi , i need to register the time when arrive the following messages: invite, alerting, progress , connected and hangup. How can I make it ? it's is possible to store these value in the extra field in the acc module ? Thanks

2 1

TCP error
by Ed Brady 12 Oct '12

12 Oct '12

I am testing connecting an MSRP client to my Kamailio instance, and I keep getting the following error in my syslog: /usr/sbin/kamailio[20055]: ERROR: <core> [tcp_read.c:293]: error reading: Connection reset by peer (104) I believe this must be a misconfiguration on my part, I just don't know where to start. Any ideas what may cause a TCP read error? If this is an uncommon error, I can supply configs to help determine the cause. Thanks, Ed

3 2

Wrong onreply_route is called after serial forking
by Alex Hermann 11 Oct '12

11 Oct '12

Hello, i noticed thet in serial forking, replies to earlier branches arriving after sending out a new branch use the onreply_route of the later branch instead of the onreply_route set before sending the earlier branch.... How to reproduce: 1) set onreply_route to A 2) relay 1st branch 3) 1st branch times out, internal 408 is created 4) tm send CANCEL to 1st branch 5) in failure route, onreply_route is set to B 6) relay 2nd branch 7) 1st branch responds with 487, and goes into reply_route B instead of A I think each branch should take the reply_route which was set before it got relayed and not pick up later changes meant for other branches. How would i fix this? -- Greetings, Alex Hermann

5 7

Question about how the RLS module handle subscribe
by Sangeeta Shah 11 Oct '12

11 Oct '12

Hello All, I am trying to understand how the RLS module handles a subscribe request. My client is sending the following: Subscribe, event = reg, no Allowed header - hence it is getting handled by the presence module (entry make its to active_watchers table) Subscribe, event - presence.winfo, no Allowed header - hence it's getting handled by the presence module (entry makes it to active_watchers table) Publish - handled by presence module (makes it to presentity table) Subscribe, event = presence, Allowed = eventlist - hence it's handled by the rls module. RLS module is having trouble sending a notify per this data in the syslog. Also I don't see any entries in the watcher table or rls_watchers or rls_presentity table. Errors: Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: presence [hash.c:342]: record not found in hash table Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [notify.c:387]: updating in hash table Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [notify.c:273]: in function agg_body_sendn_update Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x7fb2a8daa220 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x7fb2a8daa598 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x7fb2a8da9f90 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x7fb2a8da9b48 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x7fb2a8da9b28 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x7fb2a8daa608 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x7fb2a8daa570 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x7fb2a8da9410 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [subscribe.c:732]: failed sending full state notify Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [subscribe.c:781]: occured in rls_handle_subscribe Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <script>: ========= RLS Handle subscriber returned with ret code: -1 === What's it trying to do here? Any help is appreciated. Thanks, Sangeeta -- Sangeeta Shah

1 0

Kamailio presence with RLS
by Sangeeta Shah 11 Oct '12

11 Oct '12

Hello Everyone, Has anyone had any luck getting RLS working with presence. I am seeing the following errors in my log file: ct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: presence [hash.c:342]: record not found in hash table Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [notify.c:385]: updating in hash table Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [notify.c:271]: in function agg_body_sendn_update Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x7fbf415c4178 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x7fbf415c48a0 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x7fbf415c38a8 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x7fbf415c3bc8 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x7fbf415c4880 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x7fbf415c3c28 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x7fbf415c3648 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x7fbf415c3940 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [subscribe.c:732]: failed sending full state notify Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [subscribe.c:781]: occured in rls_handle_subscribe Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <script>: ========= RLS Handle subscriber returned with ret code: -1 === I think this is my first problem. There is an error later in the log file as well, but it seems Apparently from what i understand we are trying to update the entry in the presence_shtable and we cannot find it and hence we are failing to send the full state notify. Any help is appreciated. If further information is required please let me know. I can try to print out more data in the update_presence_shtable function. I have another htread with a lot of log info but for some reason the thread went cold and I just need to keep moving on getting this to work. Anyone familiar with this please let me know what the likely cause of this issue might be. Thanks, Sangeeta -- Sangeeta Shah

1 0

How can I directly access the domain of additional branches??
by Stephen Dodge (Bistech) 11 Oct '12

11 Oct '12

Hello, Could someone advise if it is possible to directly access the domain of additional branches as $rd references the domain of the main URI only. If it's not possible does anyone have any tips on how I can achieve this. Many Thanks. Steve. ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

4 5

Kamailio direct interconnectivity with PRI
by SamyGo 11 Oct '12

11 Oct '12

Hello, I've a scenario in which I've to deploy a couple Sangoma PRI cards with kamailio. What I wish is that I've some drivers for this purpose and so I don't ned to install FreeSWITCH or Asterisk in between the PRIs and Kamailio. Kindly give any feedback on what are the possibilities and options. Thanks, Sammy

4 7

Get sip clients to work with sip:user@domain and ldap
by Christophe ROY 11 Oct '12

11 Oct '12

Hello I'm quite new to SIP and TOIP so I may not understand the concepts correctly. I'm supposed to make work SIP clients (for now, linphone on android and windows) with a Kamailio 3.2 server and ldap authentication. Ldap authentication seems to be working fine, as the login/pass is validated against the Active Directory before registering the SIP client. However, Linphone seems to register the SIP client as sip:user@true_ip_address instead of sip:user@domain. If I try to make a call to sip:anotheruser@domain it doesn't work, and I can find in /var/log/syslog the following lines: Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) However, if I try to make a direct call to sip:anotheruser@his_true_IP, it works How can I make the registered users to be reachable with user@domain with Kamailio? And by the way, is there a way to get a list of registered users? Thank you for the help Christophe

2 1

kamailio 3.3.1 - uac - remote register - init delay possible?
by René Bormann 11 Oct '12

11 Oct '12

Hello @all, want to use the remote register function of the UAC module. We need to register 42 accounts on one openSER server. At kamailio startup those 42 remote register requests will be startet at the same time. This leads to a kind of FlodAttact on the openSER server of our provider, which will block us :-( We would like to delay each initial remote register by 1 second. We searched the UAC module and taught that at some place the uac.so module has to load the uacreg table and do some for loop to start the remote register process. We found function child_init() in uac.c where the function uac_reg_load_db() is called followed uac_reg_timer(). Our change to uac_reg_timer() is the sleep(1). After compiling it seems to us - that this function is not called. Is there anybody how has a tip for us? Thanks, René /** * */ void uac_reg_timer(unsigned int ticks, unsigned int initial_reg_call) { LM_ERR("----------------------- uac_reg_timer() -- enter \n"); <------>int i; <------>reg_item_t *it = NULL; <------>time_t tn; <------>tn = time(NULL); <------>for(i=0; i<_reg_htable->htsize; i++) <------>{ <------><------>/* free entries */ <------><------>it = _reg_htable->entries[i].byuuid; <------><------>while(it) <------><------>{ LM_ERR("----------------------- uac_reg_timer() -- 1 \n"); <------><------><------>if (initial_reg_call == 1) <------><------><------>{ LM_ERR("----------------------- uac_reg_timer() -- 2 \n"); <------><------><------><------>sleep(1); <------><------><------>} <------><------><------>uac_reg_update(it->r, tn); <------><------><------>it = it->next; <------><------>} <------>} }

3 8

MSRP Config
by Ed Brady 10 Oct '12

10 Oct '12

Hi, I am working on an MSRP project with Kamailio. I have Kamailio installed and configured for SIP with several test accounts created, but am unable to send MSRP messages so far. I am using Blink as a test client, though the final intentions of the project will be to use iDoubs and IMSDroid for client connections. We are having these developed to our needs right now though, so I must use another client to begin testing. I have been unable to find good documentation on the initial configuration of MSRP in Kamailio. I have seen the module documentation, but am unclear how to utilize the information there to configure Kamailio properly. Is there any documentation on getting started with a basic MSRP relay setup so that two clients can chat and send files? I am not new to SIP, but this is my first project with MSRP. Thanks, Ed

4 10

OpenSER/Kamailio Support? - Forwarding SIP INVITE Packets
by Eddie Elric 10 Oct '12

10 Oct '12

Hello fellow OpenSER/Kamailio users, I hope this is the right method of contact - I was unable to find a forum on which to post any queries. In essence, I've set up a Kamailio server of the latest version - logging on to clients works fine, etc. The idea of my setting up was for forwarding SIP messages to our test server, that's where I got stuck. The following link explains everything on Kamailio it seems - http://www.kamailio.org/wiki/cookbooks/3.3.x/core#forward So, I've added to the kamailio.cfg the following - forward("10.0.0.10","5060"); But you see, I'm not sure where I should have added it. I've tried adding to a variety of places, such as - route[SIPOUT] and onreply_route[MANAGE_REPLY] - but none are working. Cheers guys/girls. Best, Eddie

1 1

Kamailio/RTP Proxy issue
by Kamal Palei 10 Oct '12

10 Oct '12

Hi All Facing a strange issue in Kamailio / RTP proxy setup. Using 1. Kamailio - 3.1.5 2. RTP proxy 1.1 I have pne sip client, lets say it is 'A'. 'A' sent INVITE to Kamailio proxy. Kamailio proxy modified the IP address present in SDP to RTP proxy IP address, and sent to main proxy. Main proxy did respond back with 200 OK to INVITE. In 200 OK SDP, Main proxy put the IP address of Media server (media server ip address is different than main proxy server ip address). After the call establishment, it is expected that RTP proxy should send media packets to media server, instead it is sending to proxy server. Pleae do let know, how to put a work around to avoid this problem. Thanks Kamal NECS, Bangalore

2 2

TCP replace alias can cause core
by Jijo 10 Oct '12

10 Oct '12

We found a problem regarding TCP connection alias in the following code at tcp_main.c: int tcpconn_finish_connect( struct tcp_connection* c, union sockaddr_union* from) { : : /* remove all the aliases except the first one and re-add them * (there shouldn't be more then the 3 default aliases at this * stage) */ for (r=1; r<c->aliases; r++){ a=&c->con_aliases[r]; tcpconn_listrm(tcpconn_aliases_hash[a->hash], a, next, prev); } c->aliases=1; As TCP_ALIAS_REPLACE flag is set for the default TCP options value, in the function _tcpconn_add_alias_unsafe() a TCP connection alias can be moved from connection A to connection B based on the TCP alias hash. In this case, the number of aliases is incremented in the connection A, and decremented from connection B. However, in the connection B the number of aliases can reach zero (no alias). And the code above can be executed for connection B setting the number of aliases to 1 unconditionally. When this case happens, the connection B keeps an invalid alias (already excluded from connection B by tcpconn_add_alias_unsafe() function called from connection A). When the connection A is released, the aliases are also released, and this memory area can be filled with different data. As connection B has references to an invalid alias it can try to access invalid areas, and can crash Kamailio. This access happens, for example, when another alias is added to connection B. To fix it we include a check before the code: if (c->aliases>0) { for (r=1; r<c->aliases; r++){ a=&c->con_aliases[r]; tcpconn_listrm(tcpconn_aliases_hash[a->hash], a, next, prev); memset(a,0xbb,sizeof(struct tcp_conn_alias)); } c->aliases=1; } Please let us know if any comments. Thanks Jijo

2 1

[snmpstats] openserSIPRegRejectedRegistrations not being updated
by Anatoly Ivanov 10 Oct '12

10 Oct '12

Hi, I am observing weird behavior of snmpstats.so - I see openserSIPRegAcceptedRegistrations being updated, but openserSIPRegRejectedRegistrations stays at zero. Could somebody please explain under what conditions this counter is being incremented? I am using auth_radius, so I suspect I simply do not reject failed registartions the 'proper' way, thus causing the counter to stay at 0. Thank you, Anatoly.

2 3

how to use fix_nated_sdp("2") with multiple media types
by MARINA SERRANO MONTES 10 Oct '12

10 Oct '12

Hi, I have a doubt about how to use "fix_nated_sdp("2")" function when we have an scenario with multiple media types (audio and video) and the param "c" in sdp for each media type has a different IP. We are using a SCOPIA 400 MCU to multi-party in a call, and it will set up different IP to send RTP depend on media type. I have tested this function, but it rewrites always all the "c" parameter with the same value. Is there any option to use the function or to rewrite "c" value for each media type independently? Many thanks. Br, Marina ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at: http://www.tid.es/ES/PAGINAS/disclaimer.aspx

3 11

Nginx dispatching of websocket requests?
by Juha Heinanen 09 Oct '12

09 Oct '12

just to complete the thread, nginx does not currently support proxying of requests that use websocket transport. it is in the roadmap, but no date is set. when nginx receives request over websocket it converts 'Upgrade: Connection: Upgrade' header to 'Connection: close' header and removes Upgrade header. -- juha

1 0

moving all to kamailio namespace
by Daniel-Constantin Mierla 09 Oct '12

09 Oct '12

Hello, some bits were left with the old project name OpenSER, not to disturb too much at that time and see where everything goes. Other voices expressed same idea in the past, now everything is stable and development goes smooth, so it is good to make a decision because openser is too way back and new comers could be confused. In my list is renaming from openser to kamailio next attributes: - default database name (exiting configs should not be impacted that much as most of them have database provided by module parameter) - default database access usernames and passwords, respectively openser, openserro and openserrw should become kamailio, kamailioro and kamailiorw - perl module API is packed as OpenSER - this could affect some people using it, but a search and replace should do it - snmpstats mibs - probably it is going to have some impact as well Eventually we can start doing it gradually, one or two per release, but is probably better to do all at once, so people will do one upgrade with next major release. Another thing is sercmd, it is in conflict for packaging as both ser and kamailio install it. It means that one cannot install both kamailio and ser at same time from packages. One option is to install it as kamcmd. Because these changes affect users and devels, I sent the message to both mailing lists. Opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

6 7

storing and using route-set in dialog module
by Klaus Darilion 09 Oct '12

09 Oct '12

Hi! Are there any means to store the route set of a dialog (on dialog creation) and use the stored route set when forwarding in-dialog requests instead of using the requests' route set? thanks Klaus

1 0

Force to use mediaproxy/rtpproxy
by Mino Haluz 09 Oct '12

09 Oct '12

Hi, I am doing some performance tests and I cannot force SIP INVITEs to use mediaproxy within LAN, without any NAT. I'm using engage_media_proxy or rtpproxy_manage but phones are still exchanging voice data between them, not through the proxy. Is there some trick how to forcibly tell kamailio to use media/rtp proxy ? Maybe kamailio finds out that there is no nat and therefore will not use any nat traversal mechanism. Mino

1 1

get $sht array of values
by Uri Shacked 09 Oct '12

09 Oct '12

Hi, If i have two keys on a HTABLE that are the same. i undestand i need to set the key type to 1 (array). i use the $shtcn and get 3 (size is an entry as well). so far, so good. But, now i am trying to get all the values from the table to an AVP oto say it in another way - i need all values because i parse it afterwards. So, what is the syntax? using $sht() didn't work..... Uri

2 1

kamailio cores on corrupted route header
by Jijo 09 Oct '12

09 Oct '12

Hello, kamailio cores when receives a corrupted route header. For example, this was causing the core. Route: sip:10.236.236.100;transport=tcp;r2=on;lr;ftag=1348218287134-Test-553188;osb-tag=NM;nat=yes;twan=yes?[=& [=<sip:10.236.236.100;transport=tcp;r2=on;lr;ftag=1348218287134-Test-553188;osb-tag=NM;nat=yes;twan=yes?[=&%20[=> I found the problem, the pointer was not initializing to null after freeing it. Please apply this fix in the next version. Here is the diff with the original(3.2.2) and changed version. PGA:/mnt/o/kamailio-3.2.2/parser # diff -u parse_param.c.orig parse_param.c --- parse_param.c.orig 2012-10-09 09:42:58.372003500 -0300 +++ parse_param.c 2012-10-09 21:34:14.556367900 -0300 @@ -545,6 +545,7 @@ error: if (t) pkg_free(t); free_params(*_p); + *_p = 0; return -2; ok: Thanks Jijo

1 0

Kamailio XMPP support
by Vijay Thakur 09 Oct '12

09 Oct '12

Dear All, I have set up my kamailio server successfully and it is working very fine. i have followed the below link for setup: http://kb.asipto.com/kamailio:presence:k31-made-simple. Now i want to enable xmpp module. At Kamailio wiki page, i have found that there can be two way to implement xmpp module in kamilio. And they are (1) Inside the server (2) Remoter jabber server. But there is no specific "how to" to enable this module. How can i compile and enable this module in kamailio. And after this what will be the setting of my kamailio.cfg. Currently jabberd2 is not available as directed in wiki. Can i use openfire with kamailio for xmpp? How ? Kindly point me any direction. With Warm Wishes, Vijay Thakur information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

1 0

Kamailio XMPP Module Setup
by Vijay Thakur 09 Oct '12

09 Oct '12

Dear All, I have set up my kamailio server successfully and it is working very fine. i have followed the below link for setup: http://kb.asipto.com/kamailio:presence:k31-made-simple. Now i want to enable xmpp module. At Kamailio wiki page, i have found that there can be two way to implement xmpp module in kamilio. And they are (1) Inside the server (2) Remoter jabber server. But there is no specific "how to" to enable this module. How can i compile and enable this module in kamailio. And after this what will be the setting of my kamailio.cfg. Kindly point me any direction. With Warm Wishes, Vijay Thakur

1 0

Kamailio FAQ Week
by Daniel-Constantin Mierla 09 Oct '12

09 Oct '12

Hello, I'm looking at getting richer content for the FAQ wiki page: * http://www.kamailio.org/wiki/tutorials/faq/main Probably we can get it good enough with a bit of mobilization for one week. If anyone is having some questions fitting the purpose of the FAQ, just add them to the page or send via email. If you don't have the right answer, just leave it as TBA in the wiki, via email is fine to send just the question. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Nginx dispatching of websocket requests?
by Juha Heinanen 09 Oct '12

09 Oct '12

peter suggested using nginx for dispatching xcap requests and that works fine. has someone tried using nginx to dispatch also sip over websocket requests? is there some drawbacks? -- juha

2 1

Kamailio issues errors
by Sangeeta Shah 08 Oct '12

08 Oct '12

I have Kamailio 3.3 with Presence, RLS and integrated XCAP. Now I am noticing the errors in the attached syslog capture. The only thing I have changes is the DB mode to DB Only. Thanks, Sangeeta -- Sangeeta Shah

1 0

implementing call-quota with dialog module
by Aft nix 08 Oct '12

08 Oct '12

Hi, I'm trying to implement a "callquota" based on destination domains. The idea is following: every destation domain has a limit of, say 50 calls. As the sip clients are registered with the "destination domains", i've extracted the domain using "$fd" because it results as same value as destination domain. Problem is, i wanted to have separate quota for each domains. But it seems, that the quota is reached "as total" not as per domain basis. Another thing is, the number of calls reported by it is also unusually large than actual calls. This is implemenated as bellow: #!ifdef WITH_CALL_LIMIT modparam("dialog","enable_stats",1) modparam("dialog","dlg_flag", DLG_FLAG) modparam("dialog","hash_size", 4096) modparam("dialog", "profiles_with_value","callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) #!endif #!ifdef WITH_CALL_LIMIT if (is_method("INVITE")) { dlg_manage(); $var(100) = 0; get_profile_size("callquota", "$fd", "$var(100)"); if ($var(100) >= 50 ) { xdbg("DEBUG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$fd"); if (get_profile_size("callquota","$fd", "$var(100)")) { xdbg("DEBUG: there are $var(100) total calls for $fd"); } } #!endif Thanks in advance. -- -aft

2 2

Newbie question about RLS/XCAP/batch'd NOTIFYs
by Dmytro Bogovych 06 Oct '12

06 Oct '12

Greetings. I try to make iOS voip application with presence information. iOS does not let too frequent network updates from server side when runs in background mode. So I cannot rely on plain PUBLISH / SUBSCRIBE scheme - it leads to restart of application by iOS in many cases. Please see this post for details: https://devforums.apple.com/message/565667#565667 The question. Can Kamailio provide way to reduce the number of NOTIFYs? I see it as 1) iOS application creates & uploads RLS via XCAP 2) iOS application asks server about changes in presence information once per 10 minutes or when application goes to foreground state. The best way is to receive single batch'd NOTIFY answer for all watched peers. Is it possible to implement with kamailio? Thank you :)

2 2

compilation warnings
by Vieri 05 Oct '12

05 Oct '12

Hi, On a Gentoo Linux system I'm getting the following compile-time warnings (kamailio-3.3.1 ebuild): * python_msgobj.c:529:5: warning: dereferencing type-punned pointer will break strict-aliasing rules * ld_con.c:241:4: warning: implicit declaration of function 'ldap_simple_bind_s' * perl.c:340:2: warning: null argument where non-null required (argument 2) * perl.c:340:2: warning: null argument where non-null required (argument 1) Hopefully I can just ignore these messages. Thanks, Vieri

3 3

kamailio 3.3 sipcapture and Homer Capture Agent 4
by Juha Heinanen 05 Oct '12

05 Oct '12

is Homer Capture Agent 4 hep protocol incompatible with Kamalio 3.3 sipcapture module? in captagent.xml i set core_hep.conf version to 2, but looks like kamailio is not able to parse the messages: Oct 5 12:40:18 siika sip-proxy[6779]: INFO: <core> [parser/parse_fline.c:145]: ERROR:parse_first_line: method not followed by SP Oct 5 12:40:18 siika sip-proxy[6779]: INFO: <core> [parser/parse_fline.c:242]: ERROR:parse_first_line: bad message Oct 5 12:40:18 siika sip-proxy[6779]: ERROR: <core> [parser/msg_parser.c:705]: ERROR: parse_msg: message=<#002#020#002#006#023ĳ��bg#012�bg#012#002�nP#037#035#010> Oct 5 12:40:18 siika sip-proxy[6779]: ERROR: <core> [receive.c:146]: ERROR: receive_msg: parse_msg failed in kamailio config i have modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) -- juha

2 2

Does Kamailio support ECDHE ciphers for TLS?
by AJ 05 Oct '12

05 Oct '12

Hi, I have Kamailio 3.3.1 working fine with TLS 1.0, running on CentOS 6.3 However, now I want to use some ECDHE ciphersuites, for instance: ECDHE-ECDSA-AES128-SHA I have built an appropriate OpenSSL (1.0.0j) in CentOS to support this, and have generated the appropriate certs: openssl ecparam -out ca-privkey.pem -name secp256r1 -genkey -outform pem openssl req -x509 -new -key ca-privkey.pem -out ca_cert.pem -outform PEM -days 3650 openssl ecparam -out privkey.pem -name secp256r1 -genkey -outform pem openssl req -new -nodes -key privkey.pem -outform pem -out cert_req.pem openssl ca -keyfile ca-privkey.pem -cert ca_cert.pem -in cert_req.pem -out cert.pem However, when I try to connect from my phone I am always getting: Oct 3 20:27:54 /usr/sbin/kamailio[2905]: ERROR: tls [tls_server.c:1190]: TLS accept:error:1408A0C1:SSL routines:SSL3_GET_CLIENT_HELLO:no shared cipher Oct 3 20:27:54 /usr/sbin/kamailio[2905]: ERROR: <core> [tcp_read.c:1127]: ERROR: tcp_read_req: error reading If I use the OpenSSL s_server test-tool, I can connect from my phone just fine with these certs: openssl s_server -accept 8888 -cert cert.pem -key key.pem -pass stdin -CAfile calist.pem -cipher ECDHE-ECDSA-AES128-SHA Does Kamailio support ECDHE ciphersuites? If so, what am I doing wrong? Thanks! AJ

3 2

Kamailio question
by Jostein Aardal 05 Oct '12

05 Oct '12

Hi! I am a student at Oslo and Akershus University college of applied sciences. I'm working on a project where I am supposed to setup a kamailio server. Up until now I have been able to install kamailio and add a user. When I use a sip-client I am able to register and authenticate this user. And I am able to make calls between different users, as long as they both are registered on the same server. But when I try to make a call between users that are registered on different servers/proxies I fail to establish the call. When I look at the sip-messages in wireshark, I see that there is a message "407 proxy authentication required". I thought that enabling "multidomain" in kamailio.cfg would fix this problem, but it doesn't. What I have written in the kamailio.cfg-file is: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_MULTIDOMAIN #!define WITH_ALIASDB #!define WITH_DEBUG yours Jostein Aardal Oslo and Akershus University College of Applied Sciences

3 2

hello i want to active international call
by mark shaw 05 Oct '12

05 Oct '12

hello i want to active international call in my account for important . radioyork2 thedept

2 1

Client info
by Sangeeta Shah 04 Oct '12

04 Oct '12

Hello All, I am trying to use Kamailio 3.3 configured as a presence server and rls server with integrated XCAP. I am trying to test the presence and rls functionality with clients such as Jitsi (doesn't seem to support rls and doesn't let me specify presence rules per contact) and Boghe (doesn't seem to be behaving as expected!). Anyone recommend other clients that work for the functionality I described above. Thanks, Sangeeta -- Sangeeta Shah

1 0

Kamailio LCR - From Uri
by Fatima Chahrour~Vanrise Support 04 Oct '12

04 Oct '12

Dears, Am able to route call using LCR module from anywhere. I need to use certain option in the lcr tables in order to specify the ip of which calls is generated. I suppose that the From_uri column in LCR module meets my requirement, yet am not able to make it work and confused after reading several docs. My question is in what form should the from_uri column be? Is the ip enough? Are the Load params used below missing? modparam("lcr", "db_url","mysql://openser:openserrw@localhost/openser") modparam("lcr","lcr_gw_table","lcr_gw") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","gw_uri_avp", "$avp(i:709)") modparam("lcr","ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr","lcr_id_avp","$avp(s:lcr_id_avp)") What should I change in the routing script to consider the from_uri? if (!load_gws("1")) { sl_send_reply("503", "Unable To lOad GatEwAyS"); exit;} if(!next_gw()){ sl_send_reply("503", "Unable To fInD a gateWaY"); exit;} Thanks in advance, F Chahrour

3 3

sipcapture unsupported family
by Juha Heinanen 04 Oct '12

04 Oct '12

after adding a few module params modparam("sipcapture", "db_url", "MYSQL_SIP_PROXY_USAGE_URL") modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "capture_node", "EXTERNAL_HOSTNAME") i restarted my sip proxy and started to get these kind of errors to syslog: Oct 4 18:09:02 siika /usr/sbin/sip-proxy[22914]: ERROR: sipcapture [sipcapture.c:733]: ERROR: sipcapture:hep_msg_received: unsupported family [71] why they appear and how to turn them off? -- juha

3 4

Presence with RLS and integrated XCAP on Kamailio 3.3
by Sangeeta Shah 04 Oct '12

04 Oct '12

Hello All, Does anyone have presence working on Kamailio 3.3 with integrated XCAP. In theory my setup should work. I have two clients: Client A: Registers with the presence server Subscribe 1 - event = reg Subscribe 2 - event = presence.winfo Subscribe 3 - event = presence Publishes its status I see entries in the foll databases for A: Active Watchers DB: presence, presence.winfo, and reg event entries Watchers DB: No entries Presentity: presence, reg, and presence???? entries PUA list: 1 entry RLS watchers: No entries RLS presentity list: No entries XCAP List: 3 entries - rls services, resource list, and pres rules Client B: Registers with the presence server Subscribe 1 - event = presence.winfo Subscribe 2 - event = presence Subscribe 3 - event = message-summary Publishes its status I see entries in the foll databases for B: Active Watchers DB: message-summary, presence.winfo Watchers DB: 1 entry - with B as the presentity and A as the watcher Presentity: reg, presence PUA list: 1 entry RLS watchers: No entries RLS presentity list: No entries XCAP List: 2 entries resource-lists, pres rules My first question is - Is this all correct or is something going missing? Now the real problem: When Client A adds Client B to it's contact list, on Client A I see the following message: Client B Package (presence), Pending (Subscribe). This information from what I gather is coming back from the periodic notifies for the server because of the presence.winfo event. However, the notify going out to Client B does not have any indication that Client A is a watcher and needs to be authorized. This should be the case per RFC 3857. Excerpts from notifies to client A and client B: Client A NOTIFY sip:8475551001@10.51.2.181:58150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.251.12;branch=z9hG4bKdcef.e5df11e.0 To: sip:8475551001@10.50.251.12;tag=1988526 From: sip:8475551001@10.50.251.12;tag=a6a1c5f60faecf035a1ae5b6e96e979a-c52c CSeq: 3 NOTIFY Call-ID: 19607474-addd-0ffb-a9da-031235052e9d Content-Length: 347 User-Agent: kamailio (3.3.1 (x86_64/linux)) Max-Forwards: 70 Event: presence.winfo Contact: <sip:10.50.251.12:5060;transport=udp> Subscription-State: active;expires=3180 Content-Type: application/watcherinfo+xml <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2" state="partial"> <watcher-list resource="sip:8475551001@10.50.251.12" package="presence"> <watcher id="9e65d804dbc4fc269c5f1d2f6eae0650(a)0.0.0.0" event="subscribe" status="pending">sip:8475551002@10.50.251.12</watcher> </watcher-list> </watcherinfo> Client B: NOTIFY sip:8475551002@10.50.0.113:5060;transport=udp;registering_acc=10_50_251_12 SIP/2.0 Via: SIP/2.0/UDP 10.50.251.12;branch=z9hG4bK7f71.6fca9e81.0 To: sip:8475551002@10.50.251.12;tag=a30db42f From: sip:8475551002@10.50.251.12;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6692 CSeq: 3 NOTIFY Call-ID: f71666432f236f38c7c4dc467b4483e0(a)0.0.0.0 Content-Length: 195 User-Agent: kamailio (3.3.1 (x86_64/linux)) Max-Forwards: 70 Event: presence.winfo Contact: <sip:10.50.251.12:5060;transport=udp> Subscription-State: active;expires=3600 Content-Type: application/watcherinfo+xml <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2" state="full"> <watcher-list resource="sip:8475551002@10.50.251.12" package="presence"/> </watcherinfo> Also the frequency of notifies to client B is a lot less for event: presence.winfo. Any help is greatly appreciated since I've been trying to get presence to work for a few weeks now and keep running into new problems. Thanks, Sangeeta -- Sangeeta Shah

1 0

sip_capture schema?
by Juha Heinanen 04 Oct '12

04 Oct '12

i couldn't find in the source directory database schema for sip_capture table. why it is not included? also, looks like sipcapture module README is missing many of module's params. is this module still supported or has it become obsolete? -- juha

3 9

Kamailio - can't write any logs to file
by Nord7 04 Oct '12

04 Oct '12

Hello everyone. I have installed: CentOS6, Kamailio 3.3, rsyslogd Part of my Kamailio config: debug=9 log_stderror=yes memdbg=9 memlog=9 log_facility=LOG_LOCAL0 loaded modules: debugger.so modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_facility", "LOG_LOCAL0") modparam("debugger", "log_prefix", "from-debugger-with-love:") xlog.so modparam("xlog", "buf_size", 8192) modparam("xlog", "log_facility", "LOG_LOCAL0") rsyslog.conf default config with two added rows: # Save log messages with LOG_LOCAL0 in /var/log/kamailio local0.* /var/log/kamailio rsyslogd and kamailio restarted. When i run command "kamailio -ddddddd" i see debug all my xlog("L_INFO","blabla\n") or xlog("LOG_LOCAL0","L_INFO","blabla\n" ); messages. When i run "kamctl start" i don't have any logs from kamailio in my system. I'm stuck, any suggestions, will be helpful -- Best regards, Kirill Ushakov

2 2

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