- sr-users - kamailio.org

[Serusers] Need help for Installing SER as Proxy
by Kai Dietrich 21 Jul '09

21 Jul '09

Maybe I am stupid, but since two weeks I will install SER with RTP Proxy as Proxy in our Network. The Situation. We have an new PBX System, with has direct SIP support. In the local network all works fine. But we want to add external Clients without VPN. For this reason I have installed a new server with two network Devices. One Device has an Public IP from the Internet, the other one a local IP. On the VoIP Device, wich I want to connect to our PBX via Internet I have adminstrate the internal PBX IP as Registra/Proxy und the Public IP as Ourgoing Proxy. With the constellation and many configurations I have found in the Internet the Register works and I can call an Internal telephone. But from the same telephone I can't call the VoIP device. And also it looks so, that the RTP Proxy dosnet work. The third problem is, that the PBX will send the UDP Packets direct to the Internet and not to the Proxy. I have also tested siproxd. With this programm I have an working constallation. The Problem is, that only one Client can be register. But I want much more. Can you help me, what I must configure that it works? Kind regards, Kai Dietrich

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[Serusers] call limit (capacity) per gateway
by Ilya Pichugin 21 Jul '09

21 Jul '09

Hi all! How can I set call limit (capacity) per gateway? What module does provide this feature? Thanks a lot. --------------- Ilya Pichugin i.pichugin(a)ptl.ru

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[Kamailio-Users] nonce_reuse protection issues
by Klaus Darilion 21 Jul '09

21 Jul '09

Hi! I really wonder if the nonce_reuse protection feature is useful and if anybody uses it without problems. One problem I have is with retransmission: e.g: ----INV1 ---> <---407------ ----ACK-----> ----INV2------> here happens a delay to the INVITE (e.g. jam in the access uplink, SIP proxy slow, ... whatever) which causes a retransmission of the INVITE ----INV3------> (retransmission of INV2) the proxy processes INV2, authenticates the user successful and forwards the requests then the proxy processes INV3, finds out that the nonce is reused and sends back 407 --> client gives up, but the request was also forwarded by the proxy :-( How do you handle such a scenario? Do you always create the transaction before authentication? One other thing I just found out is that reuse-check is done after successful authentication - shouldn't it be done the other way round? regards Klaus

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Re: [Kamailio-Users] received=IP added in Via
by Klaus Darilion 21 Jul '09

21 Jul '09

Hi! catalina oancea schrieb: > No, actually the value in the received parameter is not equal to the > IP in the Via. I looked in the code and this seems to be the intended > behavior. It would be nice to be able to disable it, but I don't think > there is one. Then it is standard conform. If the request is received from a different address then advertised, this parameter must be added. This is because SIP requires to send the response to the IP address from which the request was received. regards klaus > > Thanks anyway > Catalina > > > 2009/7/16 Klaus Darilion <klaus.mailinglists(a)pernau.at>: >> catalina oancea wrote: >>> Form what I found in >>> http://lists.kamailio.org/pipermail/devel/2008-August/015080.html, it >>> seems to me that it's a bug that kamailio adds received=IP parameter >>> although force_rport was not called. >> The bug was something else. >> >>>> Does anybody know how to disable the received=IP Via parameter that is >>>> added by default from kamailio? For the case I want to disable it in, >>>> I have no NAT handling. Also, I have: >>>> dns=no >>>> rev_dns=no >> >> In this case "received" will only be added if the advertised IP is different >> than the IP from which the request was received. (according to RFC). >> >> So, you do not use rport but kamailio adds the received parameter and the >> value in the received parameter is identical to the IP in the Via? >> >> regards >> klaus >>

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[Kamailio-Users] sanity_check question
by Dubravko Caric 21 Jul '09

21 Jul '09

Hi all, I have a question regarding sanity module (kamailio 1.5.1). Module documentation says "The function returns true if one of the checks failed", but as you can see in case 1 (request is valid) and case 2 (request is not valid because bad request uri) requests always enter function block. Does anybody else have the same problem? Any help would be appreciated... Thanks a lot /dubravko *Case 1:* if (is_method("REGISTER")) { xlog("L_INFO", "ENTER REGISTER\n"); if (sanity_check("128","6")) { xlog("L_NOTICE", "SANITY $si:$sp TIME $Tf USER $fu METHOD $rm\n"); exit; } } ENTER REGISTER Jul 20 13:30:13 [8180] DBG:core:parse_headers: flags=2000 Jul 20 13:30:13 [8180] DBG:core:get_hdr_field: cseq <CSeq>: <1> <REGISTER> Jul 20 13:30:13 [8180] DBG:core:get_hdr_field: content_length=0 Jul 20 13:30:13 [8180] DBG:core:parse_headers: flags=ffffffffffffffff Jul 20 13:30:13 [8180] DBG:core:get_hdr_field: found end of header Jul 20 13:30:13 [8180] DBG:sanity:sanity_check: all sanity checks passed SANITY xxx.xxx.xxx.xxx:22355 TIME Mon Jul 20 13:30:13 2009 USER sip:bob@exmaple.com METHOD REGISTER Jul 20 13:30:13 [8180] DBG:core:destroy_avp_list: destroying list (nil) Jul 20 13:30:13 [8180] DBG:core:receive_msg: cleaning up *Case 2:* ... if (sanity_check("130","6")) { ... ENTER REGISTER Jul 20 13:34:06 [8226] WARNING:sanity:check_ruri_scheme: failed to parse request uri Jul 20 13:34:06 [8226] DBG:core:parse_headers: flags=2000 Jul 20 13:34:06 [8226] DBG:core:get_hdr_field: cseq <CSeq>: <1> <REGISTER> Jul 20 13:34:06 [8226] DBG:core:get_hdr_field: content_length=0 Jul 20 13:34:06 [8226] DBG:core:parse_headers: flags=ffffffffffffffff Jul 20 13:34:06 [8226] DBG:core:get_hdr_field: found end of header Jul 20 13:34:06 [8226] SANITY xxx.xxx.xxx.xxx:22359 TIME Mon Jul 20 13:34:06 2009 USER sip:bob@exmaple.com METHOD REGISTER Jul 20 13:34:06 [8226] DBG:core:destroy_avp_list: destroying list (nil) Jul 20 13:34:06 [8226] DBG:core:receive_msg: cleaning up

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[Kamailio-Users] Runtime errors from libpurple when using purple module
by German Becker 20 Jul '09

20 Jul '09

Hi, I'm trying to test the purple module, and I am getting the following runtime errors, which I think are generated by libpurple: ** (process:13916): CRITICAL **: x509_tls_peers_init: assertion `ret == 0' failed ** (process:13916): CRITICAL **: purple_presence_set_status_active: assertion `status != NULL' failed ** (process:13916): CRITICAL **: purple_presence_set_status_active: assertion `status != NULL' failed ** (process:13916): CRITICAL **: purple_presence_is_online: assertion `presence != NULL' failed ** (process:13916): CRITICAL **: purple_status_type_get_attr: assertion `status_type != NULL' failed ** (process:13916): CRITICAL **: purple_status_type_get_id: assertion `status_type != NULL' failed ** (process:13916): CRITICAL **: purple_account_set_status_list: assertion `status_id != NULL' failed ** (process:13916): CRITICAL **: purple_status_type_get_attr: assertion `status_type != NULL' failed ** (process:13916): CRITICAL **: purple_status_type_get_id: assertion `status_type != NULL' failed ** (process:13916): CRITICAL **: purple_account_set_status_list: assertion `status_id != NULL' failed Attached the logfile. Any ideas about what might be causing this? I'm using version 1.5.0

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Re: [Kamailio-Users] 603 goes to onreply_route?
by catalina oancea 18 Jul '09

18 Jul '09

Even if I don't activate reply_route, it still doesn't use the failure route. I also tried 503 and 403, for these replies kamailio goes through both reply and failure routes. 2009/7/17 Klaus Darilion <klaus.mailinglists(a)pernau.at>: > > > catalina oancea schrieb: >> >> In my case, when 603 reply arrives, only the reply route is triggered. >> I added debug messages in both reply and failure routes and only the >> one in reply route is printed. > > What happens if you do not activated the reply_route. Is then the > failure_route called? > > Is it only for 603 or for all responses >400 ? > > regards > klaus > >> >> >> 2009/7/17 Klaus Darilion <klaus.mailinglists(a)pernau.at>: >>> >>> catalina oancea schrieb: >>>> >>>> Hello >>>> >>>> From my tests I found the following situation: >>>> >>>> Kamailio ------INVITE----------> I set t_on_failure and t_on_reply >>>> Kamailio <----100 Trying------- >>>> Kamailio <----603 Declined--- Here the reply goes in onreply_route (I >>>> added an xlog to prove it), not failure_route as expected. Is this the >>>> intended behavior or is it a bug (if it's a bug I can provide more >>>> details)? >>> >>> IIRC both routes should be executed. >>> >>> regards >>> klaus >>> >

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[Kamailio-Users] 603 goes to onreply_route?
by catalina oancea 17 Jul '09

17 Jul '09

Hello >From my tests I found the following situation: Kamailio ------INVITE----------> I set t_on_failure and t_on_reply Kamailio <----100 Trying------- Kamailio <----603 Declined--- Here the reply goes in onreply_route (I added an xlog to prove it), not failure_route as expected. Is this the intended behavior or is it a bug (if it's a bug I can provide more details)? Thanks, Catalina

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Re: [Kamailio-Users] Problem upgrading from 1.4.x to 1.5.1 notls
by Henning Westerholt 17 Jul '09

17 Jul '09

On Donnerstag, 16. Juli 2009, Uriel Rozenbaum wrote: > I downloaded 1.5.2-notls and installed it. Hello Uriel, > The issue was fixed but I get the following: > Jul 16 16:56:52 sbci1pri /opt/kamailio/sbin/kamailio[3921]: > INFO:core:init_mod: initializing module exec > Jul 16 16:56:52 sbci1pri /opt/kamailio/sbin/kamailio[3921]: > ERROR:carrierroute:domain_fixup: could not find domain name '0' in map you need to specify the domain name, not the id. The id is used internally, the name is what is visible external. > [..] > In domain_name table i have this configured: > id domain > 0 domain0 > 1 domain1 Just try "domain0" as parameter. Cheers, Henning

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Re: [Kamailio-Users] received=IP added in Via
by catalina oancea 17 Jul '09

17 Jul '09

No, actually the value in the received parameter is not equal to the IP in the Via. I looked in the code and this seems to be the intended behavior. It would be nice to be able to disable it, but I don't think there is one. Thanks anyway Catalina > 2009/7/16 Klaus Darilion <klaus.mailinglists(a)pernau.at>: >> catalina oancea wrote: >>> >>> Form what I found in >>> http://lists.kamailio.org/pipermail/devel/2008-August/015080.html, it >>> seems to me that it's a bug that kamailio adds received=IP parameter >>> although force_rport was not called. >> >> The bug was something else. >> >>>> Does anybody know how to disable the received=IP Via parameter that is >>>> added by default from kamailio? For the case I want to disable it in, >>>> I have no NAT handling. Also, I have: >>>> dns=no >>>> rev_dns=no >> >> >> In this case "received" will only be added if the advertised IP is different >> than the IP from which the request was received. (according to RFC). >> >> So, you do not use rport but kamailio adds the received parameter and the >> value in the received parameter is identical to the IP in the Via? >> >> regards >> klaus >> >

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

17 Jul '09

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

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

16 Jul '09

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

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

16 Jul '09

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

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

16 Jul '09

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

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

16 Jul '09

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

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

16 Jul '09

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

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

16 Jul '09

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

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

16 Jul '09

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

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

16 Jul '09

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

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

16 Jul '09

Hi serusers(a)iptel.org, I set up a Facebook profile where I can post my pictures, videos and events and I want to add you as a friend so you can see it. First, you need to join Facebook! Once you join, you can also create your own profile. Thanks, Saiful To sign up for Facebook, follow the link below: http://www.facebook.com/p.php?i=1086097402&k=Z513ZV52PWZCUGAGYKYTWR&r 557896866 was invited to join Facebook by Tuhin Saiful. If you do not wish to receive this type of email from Facebook in the future, please click on the link below to unsubscribe. http://www.facebook.com/o.php?k=cebebd&u=557896866&mid=c8ac3aG2140d4a2G0G8 Facebook's offices are located at 1601 S. California Ave., Palo Alto, CA 94304.

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

15 Jul '09

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

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

15 Jul '09

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

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

15 Jul '09

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

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

14 Jul '09

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

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

14 Jul '09

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

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

14 Jul '09

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

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

14 Jul '09

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

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

14 Jul '09

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

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

14 Jul '09

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

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

14 Jul '09

Hello, thanks for testing and reporting back. Please cc all the time to the list, so everybody interested gets updated about conclusions. Daniel On 14.07.2009 0:05 Uhr, Marco Bungalski wrote: > Hello, Daniel! > > Thank you for you documentation - with this my linux-knowledge increased .-) > > > I've just downloaded an testet - now Kamailio don't crash with a loaded > dialog-module after > a receive of SIP-message containing an empty record-route-field. So good > luck for tomorrow :) > > Message now: > Jul 13 23:59:37 [23474] ERROR:core:print_rr_body: failed to parse RR > Jul 13 23:59:37 [23474] ERROR:dialog:populate_leg_info: failed to print > route records > > > Thanks and cheers, > > Marco > > > -----Ursprüngliche Nachricht----- > Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Gesendet: Montag, 13. Juli 2009 17:05 > An: Marco Bungalski; kamailio > Betreff: Re: AW: AW: [Kamailio-Users] preparing kamailio 1.5.2 > > Hello Marco, > > On 13.07.2009 14:28 Uhr, Marco Bungalski wrote: > >> Hello, Daniel, >> >> i know how to compile, but where to download the version with the bugfix >> included? >> >> > if you fetch from svn the branch 1.5 then it has the fix. Here is a > short description of how the versioning was done so far for kamilio > (openser) from svn: > - when a major version is released, a branch with the first two numbers > in the version is created (e.g., 1.5) > - for each minor version (including the .0 - corresponding to the major > version) a tag is created with name of the three numbers in version > - any fix for the major version is committed in the branch. Once a new > minor release is out, a new tag is created. Therefore, the branch has > all the time the most updated code for a specific major release. The > tags are snapshots at the time of release. > > For the case of 1.5.x series: > - there is one branch 1.5 > - now there are two tags on svn: 1.5.0 and 1.5.1, soon there will be 1.5.2. > - the tarballs are created from a svn checkout done as exampled in the > wiki page I sent to you. > > probably reading a bit about svn (subversion) will help you. Anyhow, > next major version will use GIT. > > Now, just to download the latest 1.5.x version, do: > svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 > kamailio-1.5 > > You need subversion package installed. The continue installation as you > do with the tarball. > > Cheers, > Daniel > >> Hello, >> >> this one should help you: >> http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn >> >> Cheers, >> Daniel >> >> On 07/13/2009 01:30 PM, Marco Bungalski wrote: >> >> >>> Hello, >>> >>> i am sorry to say that i've not testet it because i don't know where to >>> download the fix and how to implementate that :-( >>> >>> But if you give me a link i will test it today (i can simulate the bug) >>> >>> >> and >> >> >>> will send you a feedback. >>> >>> Greetings, >>> >>> Marco >>> >>> >>> >>> >>> Mit freundlichen Grüßen >>> >>> Marco Bungalski >>> >>> >>> Marco Bungalski GmbH >>> Traversale 5 >>> 27283 Verden >>> >>> Telefon: +49 4231 - 776 9999 >>> Fax: +49 4231 - 776 9998 >>> Mobil: +49 172 4204774 >>> >>> e-mail: Marco(a)Bungalski.de >>> web: www.bungalski.de >>> >>> Geschäftsführender Gesellschafter: Marco Bungalski >>> Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586 >>> >>> -----Ursprüngliche Nachricht----- >>> Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Gesendet: Montag, 13. Juli 2009 12:32 >>> An: Marco Bungalski >>> Cc: users(a)lists.kamailio.org >>> Betreff: Re: [Kamailio-Users] preparing kamailio 1.5.2 >>> >>> Hello >>> >>> On 07/13/2009 12:28 PM, Marco Bungalski wrote: >>> >>> >>> >>>> Hello, Daniel! >>>> >>>> >>>> >>>> Will the bug "Kamailio crashes when receiving an empty >>>> Record-Route-field with using the dialog-module" be fixed in 1.5.2? >>>> >>>> >>>> >>>> If yes: Thanks for that :-) >>>> >>>> >>>> >>>> >>> it should be already in the svn (therefore yes). Have you tested it? >>> >>> Cheers, >>> Daniel >>> >>> >>> >>> >> >> > > >

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

13 Jul '09

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

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

13 Jul '09

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

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

13 Jul '09

Hello Marco, On 13.07.2009 14:28 Uhr, Marco Bungalski wrote: > Hello, Daniel, > > i know how to compile, but where to download the version with the bugfix > included? > if you fetch from svn the branch 1.5 then it has the fix. Here is a short description of how the versioning was done so far for kamilio (openser) from svn: - when a major version is released, a branch with the first two numbers in the version is created (e.g., 1.5) - for each minor version (including the .0 - corresponding to the major version) a tag is created with name of the three numbers in version - any fix for the major version is committed in the branch. Once a new minor release is out, a new tag is created. Therefore, the branch has all the time the most updated code for a specific major release. The tags are snapshots at the time of release. For the case of 1.5.x series: - there is one branch 1.5 - now there are two tags on svn: 1.5.0 and 1.5.1, soon there will be 1.5.2. - the tarballs are created from a svn checkout done as exampled in the wiki page I sent to you. probably reading a bit about svn (subversion) will help you. Anyhow, next major version will use GIT. Now, just to download the latest 1.5.x version, do: svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio-1.5 You need subversion package installed. The continue installation as you do with the tarball. Cheers, Daniel > Hello, > > this one should help you: > http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn > > Cheers, > Daniel > > On 07/13/2009 01:30 PM, Marco Bungalski wrote: > >> Hello, >> >> i am sorry to say that i've not testet it because i don't know where to >> download the fix and how to implementate that :-( >> >> But if you give me a link i will test it today (i can simulate the bug) >> > and > >> will send you a feedback. >> >> Greetings, >> >> Marco >> >> >> >> >> Mit freundlichen Grüßen >> >> Marco Bungalski >> >> >> Marco Bungalski GmbH >> Traversale 5 >> 27283 Verden >> >> Telefon: +49 4231 - 776 9999 >> Fax: +49 4231 - 776 9998 >> Mobil: +49 172 4204774 >> >> e-mail: Marco(a)Bungalski.de >> web: www.bungalski.de >> >> Geschäftsführender Gesellschafter: Marco Bungalski >> Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586 >> >> -----Ursprüngliche Nachricht----- >> Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Gesendet: Montag, 13. Juli 2009 12:32 >> An: Marco Bungalski >> Cc: users(a)lists.kamailio.org >> Betreff: Re: [Kamailio-Users] preparing kamailio 1.5.2 >> >> Hello >> >> On 07/13/2009 12:28 PM, Marco Bungalski wrote: >> >> >>> Hello, Daniel! >>> >>> >>> >>> Will the bug "Kamailio crashes when receiving an empty >>> Record-Route-field with using the dialog-module" be fixed in 1.5.2? >>> >>> >>> >>> If yes: Thanks for that :-) >>> >>> >>> >> it should be already in the svn (therefore yes). Have you tested it? >> >> Cheers, >> Daniel >> >> >> > >

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

13 Jul '09

Hello, this one should help you: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn Cheers, Daniel On 07/13/2009 01:30 PM, Marco Bungalski wrote: > Hello, > > i am sorry to say that i've not testet it because i don't know where to > download the fix and how to implementate that :-( > > But if you give me a link i will test it today (i can simulate the bug) and > will send you a feedback. > > Greetings, > > Marco > > > > > Mit freundlichen Grüßen > > Marco Bungalski > > > Marco Bungalski GmbH > Traversale 5 > 27283 Verden > > Telefon: +49 4231 - 776 9999 > Fax: +49 4231 - 776 9998 > Mobil: +49 172 4204774 > > e-mail: Marco(a)Bungalski.de > web: www.bungalski.de > > Geschäftsführender Gesellschafter: Marco Bungalski > Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586 > > -----Ursprüngliche Nachricht----- > Von: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Gesendet: Montag, 13. Juli 2009 12:32 > An: Marco Bungalski > Cc: users(a)lists.kamailio.org > Betreff: Re: [Kamailio-Users] preparing kamailio 1.5.2 > > Hello > > On 07/13/2009 12:28 PM, Marco Bungalski wrote: > >> Hello, Daniel! >> >> >> >> Will the bug "Kamailio crashes when receiving an empty >> Record-Route-field with using the dialog-module" be fixed in 1.5.2? >> >> >> >> If yes: Thanks for that :-) >> >> > it should be already in the svn (therefore yes). Have you tested it? > > Cheers, > Daniel > > -- Daniel-Constantin Mierla http://www.asipto.com/

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

13 Jul '09

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

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

13 Jul '09

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

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

13 Jul '09

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

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

13 Jul '09

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

1 0

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

13 Jul '09

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

1 0

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

13 Jul '09

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

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

10 Jul '09

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

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

09 Jul '09

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

2 1

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

08 Jul '09

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

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

08 Jul '09

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

2 3

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

07 Jul '09

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

2 3

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

07 Jul '09

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

1 0

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

07 Jul '09

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

3 2

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

07 Jul '09

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

3 3

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

07 Jul '09

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

3 2

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

07 Jul '09

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

3 2

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

07 Jul '09

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

3 9

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

07 Jul '09

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

3 2

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

06 Jul '09

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

3 10

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

06 Jul '09

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

3 2

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

06 Jul '09

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

4 9

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

06 Jul '09

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

3 3

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

06 Jul '09

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

2 1

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

04 Jul '09

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

4 6

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

04 Jul '09

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

2 6

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

03 Jul '09

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

4 5

[Kamailio-Users] incomplet uri
by towhid islam 02 Jul '09

02 Jul '09

why this is happning ? uuid | username | domain | attribute | value | type | modified | +------+----------+--------------------+-----------+----------------+------+---------------------+ | 1001 | 100001 | sip:192.168.30.131 | abc | 192.168.30.131 | 0 | 2009-07-02 07:36:05 | | 1002 | 100002 | sip:192.168.30.131 | abc | 192.168.30.131 | 0 | 2009-07-02 07:36:18 | | 1003 | 100003 | sip:192.168.30.131 | abc | 192.168.30.131 | 0 | 2009-07-02 07:36:29 | if(avp_db_load("$ruri/domain","$avp(s:abc)")){ xlog("L_INFO","testing"); } ERROR:avpops:ops_dbload_avps: incomplet uri <sip:192.168.30.131> TOWHID

3 2

[Kamailio-Users] forking on URIs from DB
by Robert McGilvray 02 Jul '09

02 Jul '09

I'm trying to setup custom aliases out of a sql db that will sometimes need to fork and need some help realizing how I can assign q values and process them correctly. I'm using kam 1.5. Let's assume I query for our main support line and return 3 URIs from the db along with a separate column for the q. Sip:us-support@mycomp.com q=1.0 Sip:uk-support@mycomp.com q=1.0 Sip:in-support@mycomp.com q=0.90 Based on the values I'd want to parallel fork on the first two to send this to my UK and US support line, if that fails cancel the first two branches and route to India. Just to see if I was on the right track I tried adding them as branches using $br, which from what I understand is the only way to add branches from avps/variables. I added them by appending ;q=1.0 on the end of the URI like you would see in a Contact header but I don't think that's valid. Calls to $(branch(q) return <null> so it obviously doesn't take. If I were to just fork sequentially I could just load them into avps and call it in a failure route, but how do I handle a combination of the two? Am I wrong in trying to specify the q along with the URI? I noticed serialize_branches was removed in 1.4, which is what I use in production today in 1.3 - do I need to use the replacements in TM to order them and add the branches? Thanks in advance. -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

3 6

[Kamailio-Users] Nathelp module Option message Empty
by Matteo Piazza 01 Jul '09

01 Jul '09

Hi to everybody, I have an issue with the nathelper module and openser 1.3.4 . Basically I want generate the Option message after the registration of a client, but the server generate a UDP message with the correct ip and port but empty. Below the openser.cfg file and as attachment a pcap file where you can see the registration message and the UDP message empty. Any clue? With the same configuration in another enviroment I don't have any issue Kind Regards Matteo # # $Id: openser.cfg 4424 2008-06-27 10:25:22Z henningw $ # # OpenSER basic configuration script # by Anca Vamanu <anca(a)voice-system.ro> # # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "//etc/openser/tls/user/user-cert.pem" #tls_private_key = "//etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "//etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:77.72.192.99:5060 alias= leo.trentinonetowrk.it ####### Modules Section ######## #set module path mpath="//lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "postgres.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "domain.so" #loadmodule "presence.so" #loadmodule "presence_xml.so" loadmodule "dialog.so" loadmodule "lcr.so" loadmodule "avpops.so" loadmodule "pdt.so" loadmodule "permissions.so" loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) # ---- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "max_contacts", 10) modparam("registrar", "received_avp", "$avp(s:rcv)") # ----- uri_db params ----- modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "postgres://openser:openserrw@localhost/openser") # ----- acc params ----- modparam("acc", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("acc", "early_media", 1) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) modparam("acc", "failed_transaction_flag", 3) #modparam("acc", "log_flag", 1) #modparam("acc", "log_missed_flag", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; to_user=$tU; source_ip=$si") # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("usrloc", "nat_bflag", 6) # ----- auth_db params ----- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- modparam("alias_db", "db_url", "postgres://openser:openserrw@localhost/openser") # ----- domain params ----- modparam("domain", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence|presence_xml", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") # ----- avpops param ----- modparam("avpops","avp_url","postgres://openser:openserrw@localhost/openser") modparam("avpops","avp_table","avptable") modparam("avpops","avp_table","usr_preferences") modparam("avpops","use_domain",1) modparam("avpops","uuid_column","uuid") modparam("avpops","username_column","username") modparam("avpops","domain_column","domain") modparam("avpops","attribute_column","attribute") modparam("avpops","value_column","value") modparam("avpops","type_column","type") # ----- lcr params ----- modparam("lcr","db_url","postgres://openser:openserrw@localhost/openser") modparam("lcr","fr_inv_timer",90) modparam("lcr","fr_inv_timer_next",30) modparam("lcr", "dm_flag", 25) modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "contact_avp", "$avp(i:711)") # ------- PDT ---------- modparam("pdt", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("pdt", "db_table", "pdt") modparam("pdt", "domain_column", "source_domain") modparam("pdt", "prefix_column", "prefix") modparam("pdt", "domain_column", "domain") #modparam("pdt","check_domain",0) #modparam("pdt", "char_list", "0123456789*+[]-.") # ---------- permission ------------- modparam("permissions", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "allow_suffix", ".allow") modparam("permissions", "deny_suffix", ".deny") # ---------- dialog ------------- modparam("dialog", "dlg_flag", 4) modparam("dialog", "dlg_match_mode", 1) # ------- nathelper ------------ modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7890") modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 0) #modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(s:rcv)") modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:ping@leo.trentinonetwork.it") route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { unforce_rtp_proxy(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[1] { if(isflagset(22)) { unforce_rtp_proxy(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); setbflag(7); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("1"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { unforce_rtp_proxy(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } exit; } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[1] { route(10); route(1); } -- ========================================================================== Matteo Piazza Trentino Network s.r.l. Area Innovazione Via Gilli, 2 - 38100 TRENTO Tel (+39) 0461.020224 Mob (+39) 335.5378482 Fax (+39) 0461.020201 Cap. Soc. sottoscritto 7.573.248,00 - i. v. REG. IMP. C.F. e P. IVA 01904880224 Società soggetta a direzione e controllo da parte della Provincia Autonoma di Trento. C.F. e P. IVA 00337460224 ==========================================================================

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[Kamailio-Users] kamailio with sqlops fails to start if DB is down
by logivoip＠gmail.com 01 Jul '09

01 Jul '09

Hi All, First off - thank you for this module. I have configured two redundant DB sources in my kamailio confiuration. The DB's are accessed from Kamailio for redundancy pruposes. Kamailio and the primary DB are on the same hardware node. The secondary DB is on a remote host. If Kamailio is up and both MySQL DB's are up, the config works like a charm. If the primary or the secondary DB becomes unreachable while Kamailio is up, the config handles the exception and all works well. However, if any of the DB servers becomes unreachable and Kamailio is restarted, Kamailio can no longer start. Below is an excerpt from my config file: # -- sqlops params modparam("sqlops","sqlcon","pridb=>mysql://root:@localhost/db_table") modparam("sqlops","sqlcon","secdb=>mysql://root:@192.168.1.231/db_table") /var/log/messages Jun 26 06:16:00 test-ser /sbin/kamailio[20217]: ERROR:db_mysql:db_mysql_new_connection: driver error: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) Jun 26 06:16:00 test-ser /sbin/kamailio[20217]: ERROR:core:db_do_init: could not add connection to the pool Jun 26 06:16:00 test-ser /sbin/kamailio[20217]: ERROR:sqlops:sql_connect: failed to connect to the database [pridb] Jun 26 06:16:00 test-ser /sbin/kamailio[20217]: ERROR:core:init_mod_child: failed to initializing module sqlops, rank 1 Jun 26 06:16:00 test-ser /sbin/kamailio[20217]: ERROR:core:main_loop: init_child failed for UDP listener What is the proper way of using multiple database servers with Kamailio? Thanks in advance. --JR

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[Kamailio-Users] kamailio at linuxtag 2009
by Henning Westerholt 01 Jul '09

01 Jul '09

Hi all, i'm pleased to announce that i give a talk about the kamailio project on the linuxtag 2009 in Berlin. The title of the talk is "Building carrier grade voice over IP systems with kamailio", on wednesday the 24th june afternoon. Further details can be found in the conference program at: http://www.linuxtag.org/2009/de/program/freies-vortragsprogramm/mittwoch/vo… I'll be also present on the linuxtag on thursday, so if you would like to meet for some discussions, just drop me a mail (off-list). With best regards, Henning Westerholt

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[Kamailio-Users] new feature: execute mi commands via rpc
by Daniel-Constantin Mierla 30 Jun '09

30 Jun '09

Hello, there is a new module in repository that allow you to execute most of the mi commands from Kamailio modules (see readme) via RPC interface inherited from SER: http://sip-router.org/docbook/sip-router/branch/master/modules/mi_rpc/mi_rp… http://sip-router.org/docbook/sip-router/branch/master/rpc/ser_rpc.html One of the nicest things you can out of this is usage of sercmd - command line interface tool. It is located in utils/sercmd. Thanks to Andrei, sercmd does now tab completion for mi command. How to use it: - get sip router from git: git clone --depth 1 git://git.sip-router.org/sip-router cd sip-router make all - add to your config file: loadmodule "ctl.so" # located in 'modules_s' loadmodule "mi_rpc.so" # located in 'modules' - use sercmd cd utils/sercmd make ./sercmd Then you should get: sercmd> mi <tab><tab> arg nh_show_rtpp t_reply ul_rm_contact cleanup ps t_uac_cancel ul_show_contact debug pwd t_uac_dlg uptime dum refreshWatchers ul_add version kill shv_get ul_dump which nh_enable_ping shv_set ul_flush nh_enable_rtpp t_hash ul_rm sercmd> mi uptime 200 OK + Now:: Tue Jun 30 12:47:54 2009 + Up since:: Tue Jun 30 12:47:26 2009 + Up time:: 28 [sec] Via sercmd you can control lot of internals at runtime -- see 'help' command -- more about: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ut… Enjoy, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Kamailio-Users] [ZIP-Attachment has been removed] OpenSER stability problems in pilot project
by Munder Albert (CI/ISE) 30 Jun '09

30 Jun '09

The ZIP-attachment sent to you was blocked. Please inform the original sender to rename ZIP-files. Die Ihnen gesandte ZIP-Datei wurde geblockt. Bitte informieren Sie den ursprünglichen Absender. (Dateiendung umbenennen). Sender: Albert.Munder(a)de.bosch.com Recipients: users(a)lists.kamailio.org Subject: "OpenSER stability problems in pilot project" Datei / Filename: OpenSER_stability.zip Time: Jun 30, 2009 11:49:04 AM The cleaned message is attached. ------------------------------------------------ Hello, We are running OpenSER in a pilot project and unfortunately have some stability problems. Related information are summarized in the attached document: Any help or hints are appreciated. Mit besten Grüßen | Best regards Albert Munder Robert Bosch GmbH IT Systems Engineering (CI/ISE) Postfach 30 02 20 70442 Stuttgart GERMANY www.bosch.com Tel. +49 711 811-40562 Fax +49 711 811-5113333 Albert.Munder(a)de.bosch.com Robert Bosch GmbH, Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart HRB 14000 Aufsichtsratsvorsitzender: Hermann Scholl; Geschäftsführung: Franz Fehrenbach, Siegfried Dais; Bernd Bohr, Wolfgang Chur, Rudolf Colm, Gerhard Kümmel, Wolfgang Malchow, Peter Marks; Volkmar Denner, Peter Tyroller.

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Re: [Kamailio-Users] Problem upgrading from 1.4.x to 1.5.1 notls
by Uriel Rozenbaum 30 Jun '09

30 Jun '09

I'll try first with the debug in kamailio... I'll let you know in a few minutes... 2009/6/27 Raúl Alexis Betancor Santana <rabs(a)dimension-virtual.com> > On Saturday 27 June 2009 02:10:36 Uriel Rozenbaum wrote: > > Oki doki... how can I do that? I really don't know how to do it. > > I don't know witch server are you using, but on PostgreSQL, you have to set > an > option on the postgresql.conf file that allow your to dump by syslog all > the > DDL statements the server receives. > > I also beleive that kamailio, if you put it on debug level 6 or more, will > dump the SQL statements it's sending out. > > > -- > Raúl Alexis Betancor Santana > Dimensión Virtual >

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[Kamailio-Users] Kamailio / OpenSer billing platform
by Meftah Tayeb 30 Jun '09

30 Jun '09

hello all, please any billing modules or billing application to use with Kamailio / OpenSer? i need a prepay / Postpay application sam to Asterisk to billing (A2Billing) thanks for your help __________ Information from ESET NOD32 Antivirus, version of virus signature database 4191 (20090626) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

3 2

[Kamailio-Users] new feature: fine access to xml documents
by Daniel-Constantin Mierla 29 Jun '09

29 Jun '09

Hello, several days ago I committed support for accessing components of xml documents using pseudo-variables based on XPath syntax. The new $xml(...) PV is implemented in the presence_xml module as it works with xml docs and links libxml2. Short description in the cookbook: https://sip-router.org/wiki/cookbooks/pseudo-variables/devel#presence_xml_p… As an example: - publish has the body: <?xml version='1.0' encoding='UTF-8'?> <presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:daniel@asipto.com'> <tuple id='t8b484917'> <status><basic>open</basic></status> </tuple> <dm:person id='p7562ef60'><rpid:activities><rpid:unknown/></rpid:activities></dm:person> </presence> Getting the presence status: # this is needed because libxml2 cannot work with default no-prefix ns, so we alias the "p" prefix. modparam("presence_xml", "xml_ns", "p=urn:ietf:params:xml:ns:pidf") $xml(a=>doc) = $rb; $xml(a=>xpath:/p:presence/p:tuple/p:status/p:basic) Next is to update the presence modules to be able to work with the documents that have been updated in the config. Comments, opinions and improvements are welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Serusers] ser avpops {Scanned}
by sszasz＠digicomm.ro 29 Jun '09

29 Jun '09

Hi, I'm using SER and OpenSIPS for tests and I have a problem with avpops. How can I translate the following code snipplet ( working in opensips) to work in SER? if(avp_db_load("$fu/username","$avp(s:CBlocal)")) { sl_send_reply("403", "Call is barred"); exit; }; Thanks in advance Szasz Szabolcs -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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[Kamailio-Users] is avp the best option?
by Juan Perez 29 Jun '09

29 Jun '09

hello I need to load into memory a list of ip address associated with an index: ex: index, IP 1, 192.168.1.1 2. 192.168.1.2 3, 192.168.1.3 ......... ......... n, 192.168.1.n then at some point in the config I will call the variable, passing the index as argument and retrieving the IP as result can I do it with the avp module? cheers jp

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Re: [Kamailio-Users] sipsak
by Saúl Ibarra 29 Jun '09

29 Jun '09

Well, I think you should just need to locate the user and relay the request... but first you need to figure out why you afecta replying 501 to a notify... El 29 de jun de 2009, 11:32 a.m., "Chandrakant Solanki" < solanki.chandrakant(a)gmail.com> escribió: How can I explicitly permitting notify request from kamailio config...??? On Mon, Jun 29, 2009 at 2:45 PM, Saúl Ibarra <saghul(a)gmail.com> wrote: > > Are you explicitly not ... -- Regards, Chandrakant Solanki

1 0

[Kamailio-Users] sipsak
by Chandrakant Solanki 29 Jun '09

29 Jun '09

Hi I am using sipsak for following details... Below details are store in /tmp/mwi-immediate file NOTIFY sip:1212@mytest.com <sip%3A1212(a)mytest.com> SIP/2.0 From: <sip:asterisk@172.18.100.73 <sip%3Aasterisk(a)172.18.100.73>> To: <sip:1212@mytest.com <sip%3A1212(a)mytest.com>> Contact: <sip:1212@72.18.100.73 <sip%3A1212(a)72.18.100.73>> Call-ID: 023798(a)72.18.100.73 CSeq: 023798 NOTIFY Event: message-summary Content-Type: application/simple-message-summary Content-Length: 39 Messages-Waiting: yes Voicemail: 2/0 But when I execute sipsak command it gives following bad request #sipsak shoot -a number4 -f /tmp/mwi-immediate -s sip:1212@mytest.com<sip%3A1212(a)mytest.com>-v SIP/2.0 501 Method Not Supported Here Via: SIP/2.0/UDP 127.0.0.1:34389 ;branch=z9hG4bK.3242d1c3;rport=34389;alias;received=172.18.100.73 From: <sip:asterisk@172.18.100.73 <sip%3Aasterisk(a)172.18.100.73>> To: <sip:1212@mytest.com <sip%3A1212(a)mytest.com>> ;tag=d17c868d3732913d731aeb2b10d8dba7.1b83 Call-ID: 023798(a)72.18.100.73 CSeq: 023798 NOTIFY Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 Any idea... regarding the above problem.... -- Regards, Chandrakant Solanki

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[Kamailio-Users] Problem upgrading from 1.4.x to 1.5.1 notls
by Uriel Rozenbaum 29 Jun '09

29 Jun '09

Guys, I've successfully upgraded schema and binaries for Kamailio, but when I start it, I get: Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:core:db_check_table_version: invalid version 2 for table carrierfailureroute found, expected 3 (check table structure and table "version") Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:carrierroute:carrierroute_db_init: during table version check. Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:core:init_mod: failed to initialize module carrierroute Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:core:main: error while initializing modules If I try and set 3 to the version table (for carrierfailureroute) I get the following: Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:core:db_check_table_version: invalid version 3 for table carrierfailureroute found, expected 2 (check table structure and table "version") Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:carrierroute:carrierroute_db_init: during table version check. Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:core:init_mod: failed to initialize module carrierroute Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:core:main: error while initializing modules Is there anything I have to do (appart from following the guidelines on http://kamailio.org/dokuwiki/doku.php/install:1.4.x-to-1.5.0)? Thanks! Uriel

3 8

[Serusers] auto-attendant problem
by IL'dar AKHmetgaleev 28 Jun '09

28 Jun '09

Greetings. I have a problem with auto-attendant. It accepts call immediately without playing my greeting message. How this problem can be fixed? My sip is akhilman(a)iptel.org -- Ильдар Ахметгалеев aka AkhIL Сбт Июн 27 23:52:48 KRAST 2009 Sat Jun 27 15:52:48 UTC 2009 ---------------------------------- Visit my home page http://akhilman.blogspot.com/ jabber: akhil(a)jabber.ru ---------------------------------- Маг формирует намерение совершить то, что он намерен совершить, просто за счет того, что он намеревается это совершить. (К.Кастанеда) ---------------------------------- Linux artstation 2.6.29-gentoo-r5 #1 SMP Mon Jun 1 19:50:26 KRAST 2009 x86_64 AMD Phenom(tm) 9550 Quad-Core Processor AuthenticAMD GNU/Linux up 12 days, 14:35, 1 user, load average: 0.20, 0.35, 0.38

2 2

[Kamailio-Users] No Audio with clients Behind NAT, audio is fine with clients using public IPs (Im using rtpproxy)
by rubenrojas - Trc.es 26 Jun '09

26 Jun '09

Hello everyone, this is my first post on this list, I have installed kamailio 1.5.1 and set up a vanilla default kamailio.cfg, then I have modified the cfg to activate mysql, domain, presence, nathelper and authentication with md5, everything works as supposed to, and the clients can register, send txt messages and talk to each other. The only problem is with the audio when the two clients are behind a NAT, the phones can make a call and it does ring too, but when you pick up there is no audio both ways. when the phones have a public IP everything goes fine, it also works when I use a Linksys PAP2T whith the options to "Insert VIA received", "Insert VIA rport", "Handle VIA received", "Handle VIA rport" and "NAT mapping enable" turned on, with the Qutecom softphone works too. This is happening with thomson phones (model ST 2022), and GrandStream Budge Tone 200, it happens no matter what options I set for NATting on the phones, I've even used stun with stunserver.org or the ekiga stunserver, the phones register and can make and recieve calls, but there is no audio when you pick up the call. With a kamctl ul show, you can see that the phones have registered the Contact with their local IPs and the Received have the public IPs and ports for the NAT The only difference with the working Linksys is that they register the Contact with the public IP. Here you can see two NATed phones on the proxy Domain:: location table=512 records=2 max_slot=1 AOR:: 20000004(a)212.4.107.250 Contact:: sip:20000004@192.168.254.110:5060;transport=udp;user=phone Q= Expires:: 1150 Callid:: 72ed03f6d2f390f9(a)192.168.254.110 Cseq:: 10003 User-agent:: Grandstream BT200 1.1.6.27 Received:: sip:212.4.97.115:35379 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:212.4.107.250:5060 Methods:: 7807 AOR:: 20000000(a)212.4.107.250 Contact:: sip:20000000@192.168.254.101:5060;user=phone Q= Expires:: 2945 Callid:: 17fe-c0a80101-5-1(a)192.168.254.101 Cseq:: 6 User-agent:: THOMSON ST2022 hw2 fw3.56 00-18-F6-B5-7E-06 Received:: sip:212.4.97.115:55128 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:212.4.107.250:5060 Methods:: 4294967295 Im using rtpproxy and there is no log error that indicates that rttpproxy isn't working, in fact doing a SIP trace shows rtpproxy setting ports for the audio. I run rtpproxy with this command: rtpproxy -l 212.4.107.250 -s udp:localhost:7722 -F Any help would be greatly appreciated, I've been two weeks looking for a solution Im attaching my kamailio.cfg so you can take a look, at the end of the message Im gonna attache the SIP Trace of a call between two NATed phones (a Thomson and a GrandStream) in case anyone can help me decypher whats wrong here: this is my cfg file ************************************************************************************************** # # $Id: kamailio.cfg 5800 2009-04-20 11:01:49Z miconda $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's/#m#//g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's/#a#//g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # 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 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 dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5060 /* 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:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_mwi.so"#manually added loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- 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) # ----- rr 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) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* 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", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ #u#modparam("usrloc", "db_mode", 2) #u#modparam("usrloc", "db_url", #u# "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #a#modparam("auth_db", "calculate_ha1", yes) #a#modparam("auth_db", "password_column", "password") #a#modparam("auth_db", "db_url", #a# "mysql://openser:openserrw@localhost/openser") #a#modparam("auth_db", "load_credentials", "") #parametros de autentificacion modificados manualmente modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "ha1") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 0) #modparam("auth_db", "use_domain", 0) modparam("auth_db", "use_domain", 1)#0 encendemos con 1 porque utilizaremos multi-domain modparam("auth_db", "load_credentials", "rpid") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ modparam("presence|presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:212.4.107.250:5060") # -- nathelper modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@212.4.107.250") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) modparam("nathelper", "sipping_method", "OPTIONS") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } ##if (!uri==myself) /* replace with following line if multi-domain support is used */ if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) { route(2); } if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); setbflag(7);# sipping } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } # Auth only on registration #a# } else { #a# # authenticate if from local subscriber (uncomment to enable auth) #a# if (from_uri==myself) #a# { #a# if (!proxy_authorize("", "subscriber")) { #a# proxy_challenge("", "0"); #a# exit; #a# } #a# if (is_method("PUBLISH")) #a# { #a# if ($au!=$tU) { #a# sl_send_reply("403","Forbidden auth ID"); #a# exit; #a# } #a# } else { #a# if ($au!=$fU) { #a# sl_send_reply("403","Forbidden auth ID"); #a# exit; #a# } #a# } #a# #a# consume_credentials(); #a# # caller authenticated #a# } #a# } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } ************************************************************************************************** ************************************************************************************************** And here goes the SIP Trace for a NATed to NATed hardphones: ************************************************************************************************** U +0.161561 212.4.97.115:35379 -> 212.4.107.250:5060 INVITE sip:20000000@212.4.107.250;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone> Supported: replaces, timer, path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 332 v=0 o=20000004 8000 8000 IN IP4 192.168.254.110 s=SIP Call c=IN IP4 192.168.254.110 t=0 0 m=audio 40000 RTP/AVP 4 3 18 0 8 9 97 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:60 # U +0.000407 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668;rport=35379;received=212.4.97.115 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 # U +0.000034 212.4.107.250:5060 -> 212.4.97.115:55128 INVITE sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone> Supported: replaces, timer, path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 348 v=0 o=20000004 8000 8000 IN IP4 192.168.254.110 s=SIP Call c=IN IP4 212.4.107.250 t=0 0 m=audio 35752 RTP/AVP 4 3 18 0 8 9 97 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:60 a=nortpproxy:yes # U +0.019311 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Content-Length: 0 # U +0.030480 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Length: 0 # U +0.000083 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Length: 0 # U +6.510103 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000365 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes # U +0.034122 212.4.97.115:35379 -> 212.4.107.250:5060 ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bKdf5e0ceed72f3797 Route: <sip:212.4.107.250;lr=on;nat=yes> From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone> Supported: path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 ACK User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.000245 212.4.107.250:5060 -> 192.168.254.101:5060 ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.2 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bKdf5e0ceed72f3797 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone> Supported: path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 ACK User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.458031 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000246 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes # U +0.999724 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000295 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes

5 5

[Kamailio-Users] NAThelper and multipart SDP
by Pascal Maugeri 26 Jun '09

26 Jun '09

Hi Does NATHelper support properly multipart SDP (eg. INVITE with application/sdp and application/resource-lists+xml) ? Regards, Pascal

12 53

[Kamailio-Users] Kamailio + Asterisk MWI Implementation
by Chandrakant Solanki 25 Jun '09

25 Jun '09

Hi All, Is there any link or document for implementing MWI (kamailio + asterisk) step by step.. I tried lot of searching but don;t found any such a good link or document. Please help me out. -- Regards, Chandrakant Solanki

4 5

[Serusers] Server dimensioning
by inge 25 Jun '09

25 Jun '09

Hi all, I wonder about the "child" parameter in ser.cfg. What's the average processing capacity (number of requets/process) ? Currently, the parameter is set to 4, the default value. What's the risk of the increase ? SER runs on RedHat AS4 with Intel Xeon Quad 3.0 Ghz. Thanks for your support.

2 2

[Kamailio-Users] new feature: PV class $T_inv(pv)
by Daniel-Constantin Mierla 25 Jun '09

25 Jun '09

Hello, by loading the tmx module, one can get access to INVITE attributes while processing the CANCEL, see more at: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#t_inv_pv Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

2 1

[Kamailio-Users] LRN routing - 302 sip code
by Mik Cheez 24 Jun '09

24 Jun '09

One of our carriers gives us an option to check LRN routing via SIP before we attempt to send out a call. Our request will go out as a regular invite, then we'll get a 302 return code back, with the result contained in the header. My question is, how exactly is the flow of this kind of call supposed to work? I thought I could do a regular failover, but that doesn't work. Does anyone have a suggestion? Mik

3 4

[Kamailio-Users] script var or avp in ds_select_dst (dispatcher)?
by Robert McGilvray 24 Jun '09

24 Jun '09

Can someone please confirm that I cannot use a script var or avp in the set field when calling ds_select_dst from the dispatcher module? I'm trying to use dialplan to return a gateway group via attr, then call ds_select_dst with that but get the following error in my logs. If this is my problem are there any workarounds? The goal is to call ds_select_dst with a dynamic value. Example - ds_select_dst("$var(dpattr)", "0") Jun 24 14:50:49 [1125] ERROR:core:fixup_get_ivalue: no valid PV value found (error in scripts) Jun 24 14:50:49 [1125] ERROR:dispatcher:w_ds_select_dst: no dst set value If I hardcode the value which in this case is 30 it works as expected. I've verified that the correct value is in $var(dpattr) with an xlog right before the call to ds_select_dst. if ($ruri.user =~ "[0-9]{4}") { xlog("L_NOTICE", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu -> $ru - Internal 4 digit call\n"); if (dp_translate("2", "$ruri.user/$var(dpuser)")) { xlog("L_NOTICE", "KAM_HOST: ($ci) -> DP result='$var(dpuser)', attr='$var(dpattr)'\n"); setflag(FLAG_DIALPLAN); } else { xlog("L_NOTICE", "KAM_HOST: ($ci) -> No matching dialplan for $ru\n"); } } some other if statements here with other pattern matches... if (isflagset(FLAG_DIALPLAN)) { xlog("L_NOTICE", "KAM_HOST: ($ci) -> DP Attr is $var(dpattr)\n"); if (ds_select_dst("$var(dpattr)", "0")) { xlog("L_NOTICE", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu -> DST='$avp(s:dispatcher-dst)', \ GRP='$avp(s:dispatcher-grp)', CNT='$avp(s:dispatcher-cnt)'\n"); } else { xlog("L_NOTICE", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu -> $ru - No dst gateways found\n"); sl_send_reply("404", "No destination gateways found\n"); } } Thanks -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

1 1

[Kamailio-Users] SIP forking proxy
by Zeljko Lovrincevic 24 Jun '09

24 Jun '09

Hi, I need SIP forking proxy, and I would like to use OpenSER for that purpose. I have few questions if you could help. I am interesting for behavior when 200 OK INVITE is received for each forked dialog. Should only first 200 OK INVITE be passed to UA (the other will be terminated by proxy), or 200 OK for all forked dialogs should be passed to UA? Is it possibly to define some static configuration how to fork a dialogs or we always need REGISTER? Thanks in advance. BR/Zeljko

3 3

Re: [Kamailio-Users] [sr-dev] new version of lcr module
by Daniel-Constantin Mierla 24 Jun '09

24 Jun '09

Hi Juha, please add a short note in the wiki: https://sip-router.org/wiki/features/new-in-devel Makes easier to build new release notes and spread details about what is new in development version. Thanks, Daniel PS. I cc-ed users mailing list as this was a new feature announcement. On 06/22/2009 04:55 PM, Juha Heinanen wrote: > i just pushed a new version of lcr module, which resulted from > discussions on the mailing lists. > > most important new feature is support for any number of independent lcr > instances, which improves virtualization of sr. for example, different > domains can now manage their own least cost routing to their own gws. > > i also replaced aliveness check with new defunct_gw() function, which > script writer can use to defunct an unresponsive or busy gw for a > desired period of time. > > in order to keep lcr module manageable, i removed gw group related > functions (which were not efficient anyway). if someone is using them, i > hope that they can be replaced by lcr instances. > > -- juha > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev > > -- Daniel-Constantin Mierla http://www.asipto.com/

2 1

[Kamailio-Users] Failure at startup
by Francisco Javier Lizarán Vilches 24 Jun '09

24 Jun '09

Hi all: Running Openser v1.2.3-notls. It has been suffering failures at startup since some weeks ago and it seems to be related with database operation. Nothing has changed in config. The only remarkable change in the system is a slight increase in database registers at subscribers, location and lcr tables. Log excerpt from openser startup: Jun 19 15:23:32 openser[2196]: INFO: statistics manager successfully initialized Jun 19 15:23:32 openser[2196]: StateLess module - initializing Jun 19 15:23:32 openser[2196]: TM - initializing... Jun 19 15:23:32 openser[2196]: Maxfwd module- initializing Jun 19 15:23:32 openser[2196]: INFO:ul_init_locks: locks array size 512 Jun 19 15:23:32 openser[2196]: AUTH module - initializing Jun 19 15:23:32 openser[2196]: AUTH_DB module - initializing Jun 19 15:23:32 openser[2196]: TextOPS - initializing Jun 19 15:23:32 openser[2196]: AVPops - initializing Jun 19 15:23:32 openser[2196]: permissions - initializing Jun 19 15:23:32 openser[2196]: WARNING: File not found: /etc/openser/permissions.allow Jun 19 15:23:32 openser[2196]: Default allow file (/etc/openser/permissions.allow) not found => empty rule set Jun 19 15:23:32 openser[2196]: WARNING: File not found: /etc/openser/permissions.deny Jun 19 15:23:32 openser[2196]: Default deny file (/etc/openser/permissions.deny) not found => empty rule set Jun 19 15:23:32 openser[2196]: ACC - initializing Jun 19 15:23:32 openser[2196]: Dialog module - initializing Jun 19 15:23:32 openser[2196]: INFO: SNMPStats: Starting up the SNMPStats Module Jun 19 15:23:32 openser[2196]: UAC - initializing Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type (32) instead of <Refer-To> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type (7) instead of <Contact> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Referred-By> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Replaces> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Forwarder> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Referred-by> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <RG> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Origin> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Referred-By> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Replaces> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type name <Forwarder> Jun 19 15:23:32 openser[2196]: xl_fill_extra_spec: found extra item [DLG_status] Jun 19 15:23:32 openser[2196]: xl_fill_extra_spec: found extra item [T_reply_code] Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type (4) instead of <From> Jun 19 15:23:32 openser[2196]: INFO:textops:hname_fixup: using hdr type (3) instead of <To> Jun 19 15:23:32 openser[2196]: xl_fill_extra_spec: found extra item [T_reply_code] Jun 19 15:23:32 openser[2196]: xl_fill_extra_spec: found extra item [T_reply_code] Jun 19 15:23:32 openser[2196]: xl_fill_extra_spec: found extra item [T_branch_idx] Jun 19 15:23:32 openser[2196]: INFO: udp_init: SO_RCVBUF is initially 109568 Jun 19 15:23:32 openser[2196]: INFO: udp_init: SO_RCVBUF is finally 262142 ...Some registers, invites, etc logs... Jun 19 15:23:34 openser[2204]: new_connection: Can't connect to MySQL server on 'bbdd.bariksip.es' (4) Jun 19 15:23:34 openser[2201]: new_connection: Can't connect to MySQL server on 'bbdd.bariksip.es' (4) Jun 19 15:23:34 openser[2201]: ERROR:ul:child_init(5): Error while connecting database Jun 19 15:23:34 openser[2204]: ERROR:ul:child_init(-1): Error while connecting database Jun 19 15:23:34 openser[2201]: ERROR:init_mod_child: Error while initializing module usrloc, rank 5 Jun 19 15:23:34 openser[2204]: ERROR:init_mod_child: Error while initializing module usrloc, rank -1 Jun 19 15:23:34 openser[2201]: init_child failed Jun 19 15:23:34 openser[2204]: timer: init_child failed Jun 19 15:23:34 openser[2203]: INFO: signal 15 received Jun 19 15:23:34 openser[2201]: INFO: SNMPStats: The SNMPStats module got the kill signal Jun 19 15:23:34 openser[2201]: Shutting down the AgentX Sub-Agent! Jun 19 15:23:34 openser[2201]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill Jun 19 15:23:34 openser[2202]: INFO: signal 15 received Jun 19 15:23:34 openser[2200]: INFO: signal 15 received Jun 19 15:23:34 openser[2199]: INFO: signal 15 received Jun 19 15:23:34 openser[2198]: INFO: signal 15 received Jun 19 15:23:34 openser[2197]: INFO: signal 15 received Jun 19 15:23:34 openser[2204]: INFO: SNMPStats: The SNMPStats module got the kill signal Jun 19 15:23:34 openser[2204]: Shutting down the AgentX Sub-Agent! Jun 19 15:23:34 openser[2204]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill Jun 19 15:23:34 openser[2196]: new_connection: Can't connect to MySQL server on 'bbdd.bariksip.es' (4) Jun 19 15:23:34 openser[2196]: ERROR:ul:child_init(0): Error while connecting database Jun 19 15:23:34 openser[2196]: ERROR:init_mod_child: Error while initializing module usrloc, rank 0 Jun 19 15:23:34 openser[2196]: main: error in init_child Jun 19 15:23:34 openser[2196]: INFO: SNMPStats: The SNMPStats module got the kill signal Jun 19 15:23:34 openser[2196]: Shutting down the AgentX Sub-Agent! Jun 19 15:23:34 openser[2196]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill After multiple attempts the result is the same. This is the list of modules that use db: acc|auth_db|permissions|uri_db|usrloc|avpops|lcr|avpops|domain|speeddial|alias_db Looking at the traces, when openser places the request packet carrying this query "select prefix,from_uri,grp_id,priority from lcr", mysql server begins to perform retransmissions of the requested data; although openser eventually sends back the ack. Later request's responses from mysql server needs also retransmit until openser fails. If I change, for example, usrloc's dbmode to 0 (disables db use), the module that fails this time is auth_db. The mySql server is located in the same LAN that Openser(0.112s rtt). I have realized that Openser can be started without any problem when: -Log level is raised to 4(!). or -Another mysql server in a remote location is used (10s rtt). Has anybody experienced an issue like this? Thanks: Francisco Javier Lizarán Vilches ---------------------------------------------- BARIK (Grupo Ormazabal) Departamento Técnico Tel.: +34 917 479 900 www.barik.es <http://www.barik.es/> ----------------------------------------------

1 0

[Serusers] LCR module
by Ron 24 Jun '09

24 Jun '09

Hi All, Sorry about this post but i just can't figure out the logic: i'm using lcr to send calls to pstn, when my user dials a number that begins with 00 i re-route it to my termination providers. e.g.: if ((uri=~"^(sip:)?00[0-9]*@([a-z]+\.)?mydomain.com") || (uri=~"^(sip:)?00[0-9]*@22\.22\.22\.22")) { strip(2); prefix("1234"); if(!load_gws()){ sl_send_reply("500", "Internal server error, unable to load gateways"); break; } } fortunately i have the same prefix on my termination providers. so on my lcr table i have defined all country codes to have the 1234 prefix, e.g 12341 for US , 123465 for singapore. etc. now i have a new provider, but prefix is different, how can i detect on lcr when all calls fail. i change prefix. i hope my question is clear sorry. regards, ron

1 0

[Serusers] LCR module
by Ron 24 Jun '09

24 Jun '09

Hi All, Sorry about this post but i just can't figure out the logic: i'm using lcr to send calls to pstn, when my user dials a number that begins with 00 i re-route it to my termination providers. e.g.: if ((uri=~"^(sip:)?00[0-9]*@([a-z]+\.)?mydomain.com") || (uri=~"^(sip:)?00[0-9]*@22\.22\.22\.22")) { strip(2); prefix("1234"); if(!load_gws()){ sl_send_reply("500", "Internal server error, unable to load gateways"); break; } } fortunately i have the same prefix on my termination providers. so on my lcr table i have defined all country codes to have the 1234 prefix, e.g 12341 for US , 123465 for singapore. etc. now i have a new provider, but prefix is different, how can i detect on lcr when all calls fail. i change prefix. i hope my question is clear sorry. regards, ron

1 0

[Kamailio-Users] [REQ, $] db_rest (or db_http) module for database access
by Konstantin Cherkasov 23 Jun '09

23 Jun '09

Hello! Our company seeking for programmer who can create module for flexible database access. I call it db_rest (or db_http). This module must provide internal database API for other modules like db_auth. For access to external data it must use HTTPS-server and use REST-style data access (http://en.wikipedia.org/wiki/Representational_State_Transfer). Instead of making query like "SELECT f1,f2,f3 FROM tableA WHERE f4 = 'somevalue'" it must call HTTPS-server like this: GET http://hostname/database/tableA&c=f1,f2,f3&k=f4&op=eq&v=somevalue Server must return delimiter-separated data. Similarly (using POST) it must implement support for insert, update and so on. Of course we would like to see this module inside the main project repository. If you are interested please contact me for details (requirements, $$). -- Konstantin

4 17

[Kamailio-Users] Avoid encoding of CDR's written to database by Radius
by CSB 23 Jun '09

23 Jun '09

When a CDR is written that contains certain characters (e.g. +, ;) it is encoded (like a URL gets encoded). The SQL insert statement query is produced by OpenSER/Radius: 10625 Query INSERT INTO radacct (RadAcctId, AcctSessionId, AcctUniqueId, UserName, Realm, NASIPAddress, NASPortId, AcctStartTime, AcctStopTime, AcctSessionTime, AcctInputOctets, AcctOutputOctets, CalledStationId, CallingStationId, AcctTerminateCause, ServiceType, FramedProtocol, FramedIPAddress, AcctStartDelay, AcctStopDelay, SipResponseCode, SipMethod, SipTranslatedRequestURI, SipToTag, SipFromTag, SipRPID, SourceIP, SourcePort, CanonicalURI) VALUES (NULL, '149a4cbd403b071e109d89b50a17b3a5(a)202.180.1.1', 'e04d0b07ab2eca3e', SUBSTRING_INDEX('sip:094455667@202.180.1.1',':',-1), SUBSTRING_INDEX('sip:094455667@202.180.1.1', '@',-1), '127.0.0.1', '5060', '2009-06-20 11:44:49', '0', '0', '0', '0', SUBSTRING_INDEX('sip:=2B6012345678@147.202.1.1',':',-1), SUBSTRING_INDEX('sip:094455667@202.180.1.1',':',-1), '200', 'SIP', '', '', '0', '0', '200', 'Invite', SUBSTRING_INDEX('',':',-1), 'as03a3d864', 'as69938f56', '', '', '', SUBSTRING_INDEX('',':',-1)) However the records written to the file system by OpenSER/Radius are not encoded: Sat Jun 20 11:44:49 2009 Acct-Status-Type = Start Service-Type = SIP Sip-Response-Code = 200 Sip-Method = Invite Event-Timestamp = "Jun 20 2009 11:44:49 NZST" Sip-From-Tag = "as69938f56" Sip-To-Tag = "as03a3d864" Acct-Session-Id = "149a4cbd403b071e109d89b50a17b3a5(a)202.180.1.1" Called-Station-Id = "sip:+6012345678@147.202.1.1" Calling-Station-Id = "sip:094455667@202.180.1.1" User-Name = "sip:094455667@202.180.1.1" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "e04d0b07ab2eca3e" Timestamp = 1245455089 So how can I write +6012345678 to the database, rather than =2B6012345678? Any advice appreciated. Regards Cameron

2 2

[Serusers] SER Groups
by Shahbaz Ali 23 Jun '09

23 Jun '09

plz help me in setting the ser group mangment... how i add this feature....

1 0

[Kamailio-Users] Siputils error
by Carlos A. Alvarez 22 Jun '09

22 Jun '09

Hello, I keep getting this error when I try to setup ring_timeout ERROR:core:set_mod_param_regex: parameter <ring_timeout> not found in module <utils> I tried setting the modparam for siputils as well to no avail. I already read throught the documentation and I made sure that I am loading all pre requisites for this module. Also the command options is recorded wrong in the 1.5.1 documentation: modparam("options", "options_accept", "application/*") <-according to docs Looking through the source I found out that the correct command is modparam("siputils", "options_accept", "application/*") I wonder is the same is the issue with ring_timeout. Please help

5 8

[Kamailio-Users] Question about manually manipulating RURI
by Geoffrey Mina 22 Jun '09

22 Jun '09

I have a need to perform some manipulation of the user part of the RURI. I have some dialers which send INVITES with the RURI in the following format: d15555551212@123.345.123.345:5060 I need to check for the existence of the "d" and strip it off if it is present. Basically what I want is pseduo code is: if(left($ruri),1 == "d"){ $ruri = right($ruri,len($ruri)-1); } Thanks!

4 5

[Kamailio-Users] Cancel messes with ACC
by Mark Kent 22 Jun '09

22 Jun '09

Oops, found it... there is a typo in the example config file: # ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; See the missing 's' on the table name... I guess I didn't see the error message when entering this. Thanks, -mark

2 1

[Kamailio-Users] Sending from 1 of N ip address
by Stagg Shelton 21 Jun '09

21 Jun '09

I have a requirement from an upstream provider that is providing multiple termination only gateways to me. The requirement is that I have to have a unique source ip address for each gateway that I will be terminating calls to. I would like to use only one kamailio system bound to multiple ip addresses to meet this requirement. I have my LCR built with the various routes to the gateways. I have identified the function force_send_socket([proto:]address[:port]) which seems like it can be used to meet this functional requirement. I am a little confused how to implement this function with LCR. My basic requirement is that if upstream gateway ip address == 1.2.3.4 then originate and maintain the dialog from my source ip 4.3.2.1 else if gateway ip == 6.7.8.9 then originate and maintain dialog using soure ip 9.8.7.6, else if ... N upstream gateways Is force_send_socket([proto:]address[:port]) the function that I need to be trying to use to meet this requirement? Is there a simpler method that can be used instead? Any help will be appreciated. Stagg

3 5

[Kamailio-Users] Cancel messes with ACC
by Mark Kent 20 Jun '09

20 Jun '09

I've been using openser 1.1.0 in a small pstn gateway config for a few years. I recently upgraded to kamailio 1.5.1. I started completely from scratch, with no DB, no users, and started with the example kamailio.cfg file. I made a few changes mostly at the bottom of the main request routing logic to ship things off to my pstn gateway, and re-entered things into the db. Works fine. Then I tried to enable accounting, with mysql, and this works when calls complete, but on a CANCEL I get this: kamailio[87208]: ACC: call missed: timestamp=1245456629;method=INVITE;from_tag=6343AD47-243F70A2;to_tag=2486;call_id=4529cab6-c7e08a51-dd349b2c(a)192.0.2.214;code=487;reason=Request Terminated;src_user=foouser;src_domain=sip.example.net;dst_ouser=18005551212;dst_user=18005551212;dst_domain=172.16.16.16 kamailio[87208]: ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'dst_user' in 'field list' kamailio[87208]: ERROR:core:db_do_insert: error while submitting query kamailio[87208]: ERROR:acc:acc_db_request: failed to insert into database My CANCEL processing is still the generic processing in the example kamailio.cfg file. Any hints? Thanks, -mark

1 0

Re: [Kamailio-Users] Question about manually manipulating RURI
by Carlos A. Alvarez 19 Jun '09

19 Jun '09

I don't know if this is what you are looking for, but I do something similar to process the From and To message based on a dialed prefix. Please note below my if statement and feel free to modify it for your own use: if ( search("To: <[Ss][Ii][Pp]:56011") ) { subst('/^To:(.*)sip:56011([0-9]+)@(.*)$/To:\1sip:+\2@\3/ig'); subst('/^From: "(.*)sip:([0-9]+)@(.*)$/From: "+\1sip:+\2@\3/ig'); }; On Fri, 2009-06-19 at 15:43 +0000, Geoffrey Mina wrote: Thanks, but I believe I just found the simpler way to strip, but I'm still not sure how I would check for the existence of the "d". http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#strip_int I need to do something like: if(left($ruri,1) == "d"){ //how do i write this statement? strip(1); } t_relay(); is that correct? thanks, Geoff On Fri, Jun 19, 2009 at 3:23 PM, Pascal Maugeri<pascal.maugeri(a)gmail.com<mailto:pascal.maugeri@gmail.com>> wrote: > Hi Geoffrey > > Why don't you try: > http://www.kamailio.org/docs/modules/1.5.x/textops.html#id2467793 > > Regards, > Pascal > > On Fri, Jun 19, 2009 at 9:01 PM, Geoffrey Mina <geoffreymina(a)gmail.com<mailto:geoffreymina@gmail.com>> > wrote: >> >> I have a need to perform some manipulation of the user part of the >> RURI. I have some dialers which send INVITES with the RURI in the >> following format: >> >> d15555551212(a)123.345.123.345<mailto:d15555551212@123.345.123.345>:5060 >> >> I need to check for the existence of the "d" and strip it off if it is >> present. Basically what I want is pseduo code is: >> >> if(left($ruri),1 == "d"){ >> $ruri = right($ruri,len($ruri)-1); >> } >> >> Thanks! >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org<mailto:Users@lists.kamailio.org> >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org<mailto:Users@lists.kamailio.org> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

1 0

[Kamailio-Users] Dialog from 1 of N IP Addresses
by Stagg Shelton 19 Jun '09

19 Jun '09

I have a requirement from an upstream provider that is providing multiple termination only gateways to me. The requirement is that I have to have a unique source ip address for each gateway that I will be terminating calls to. I would like to use only one kamailio system bound to multiple ip addresses to meet this requirement. I have my LCR built with the various routes to the gateways. I have identified the function force_send_socket([proto:]address[:port]) which seems like it can be used to meet this functional requirement. I am a little confused how to implement this function with LCR. My basic requirement is that if upstream gateway ip address == 1.2.3.4 then originate and maintain the dialog from my source ip 4.3.2.1 else if gateway ip == 6.7.8.9 then originate and maintain dialog using soure ip 9.8.7.6, else if ... N upstream gateways Is force_send_socket([proto:]address[:port]) the function that I need to be trying to use to meet this requirement? Is there a simpler method that can be used instead? Any help will be appreciated. Stagg

2 1

[Kamailio-Users] [OT] SIP ALG Detector released
by Iñaki Baz Castillo 19 Jun '09

19 Jun '09

Hi, as part of my personal battle against SIP ALG routers, I've created an utility to detect such routers: http://dev.sipdoc.net/wiki/sip-stuff/SIP-ALG-Detector It has two parts: client and server: Basically, the client node running into the LAN sends an INVITE to the server node (running in a host with public IP). The request could be modified by the LAN router if SIP ALG is enabled. The server encodes the received request in Base64 and appends it to the SIP response. The client receives the response, decodes the body and gets the request as it arrived to the server. Then it creates a diff between the original request and the the request the server node received. These differences are displayed in the screen. Both, UDP and TCP, tests are performed. The client node is coded in Ruby and should work in any operating system (if Ruby is installed). The server node is also coded in Ruby. For those interested in try it, I have a server node running in my personal server: 87.98.230.161:5060 You can test the client against my server. Feedback is welcome :) Regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 13

[Kamailio-Users] lcr - tag value NULL prefixed as "NULL"
by Andreas Granig 19 Jun '09

19 Jun '09

Hi, Kamailio 1.5: When having gw.tag column set to NULL in db (which seems to be allowed according to db schema), then next_gw() prefixes the R-URI user with the string "NULL". Setting it to empty string (''), nothing is prefixed. I'd expect that from a value of NULL as well. Could anyone please confirm this? Also, in http://kamailio.org/docs/modules/1.5.x/lcr.html the parameter prefix_mode is missing. It's available in README.lcr though, so probably the page is out of date? Andreas

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