- sr-users - kamailio.org

Problems with RTPPROXY module
by Michael Nielsen 21 Aug '15

21 Aug '15

Hello everyone, I'm new to Kamailio and are trying this tutorial: http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I've installed everything and are trying to launch Kamailio, but it fails with the following error: ----- systemctl status kamailio.service ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2015-08-21 06:26:26 CDT; 8s ago Process: 1474 ExecStart=/etc/init.d/kamailio start (code=exited, status=1/FAILURE) Aug 21 06:26:26 sip kamailio[1474]: error in config file /etc/kamailio/kamailio.cfg, line 234, column 12-25: failed to load module 0(1483) ERROR: <core> [sr_module.c:563]: load_module(): could not find module <rtpproxy> in </usr/lib64...3432]: yyerror_at(): Aug 21 06:26:26 sip kamailio[1474]: r in config file /etc/kamailio/kamailio.cfg, line 270, column 35: Can't set module parameter 0(1483) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <registrar> found 0(148...ailio/kamailio.cfg, Aug 21 06:26:26 sip kamailio[1474]: : <core> [modparam.c:150]: set_mod_param_regex(): No module matching <acc> found 0(1483) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 295, column ...mod_param_regex(): N Aug 21 06:26:26 sip kamailio[1474]: 1483) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 321, column 36: Can't set module parameter 0(1483) ERROR: <core> [modparam.c:150]: set_mod_para...]: yyerror_at(): par Aug 21 06:26:26 sip kamailio[1474]: file /etc/kamailio/kamailio.cfg, line 378, column 50: Can't set module parameter 0(1483) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command is_method (params 1) 0(1483) : <co...mailio.cfg, line 417 Aug 21 06:26:26 sip kamailio[1474]: fg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 436, column 24: unknown command, missing loadmodule? 0(1483) ERROR: <core> [pvapi.c:810]: pv_parse_spec2(): erro...): wrong char [U/85] Aug 21 06:26:26 sip kamailio[1474]: failed! Aug 21 06:26:26 sip systemd[1]: kamailio.service: control process exited, code=exited status=1 Aug 21 06:26:26 sip systemd[1]: Failed to start LSB: Start the Kamailio SIP proxy server. Aug 21 06:26:26 sip systemd[1]: Unit kamailio.service entered failed state. Hint: Some lines were ellipsized, use -l to show in full. ----- It seems that if it cannot find the module RTPPROXY. I have installed it via apt-get install rtpproxy and it seems to be running. What to do?

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Debian nightly builds
by Jon Bonilla (Manwe) 21 Aug '15

21 Aug '15

Hi all Trying to install latest nightly build of several branches I find that they are from June. Is the repo outdated? http://deb.kamailio.org/kamailiodev-nightly/pool/main/k/kamailio/ cheers, Jon

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Presence and NAT
by Pramod Venugopal 21 Aug '15

21 Aug '15

Hello everyone, I have a Kamailio 4.2 system on Amazon AWS. When subscribing to presence the contact is sent out with the internal IP rather than the external public IP. The subscribes are being sent to it with the external hostname ( which resolves to the external IP ). Where would I be able to change the IP sent in the Contact header ? Pramod Venugopal pramod(a)perseus-tech.com

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Back to the dead communications: No outgoing audio
by Manuel Camargo Lominchar 20 Aug '15

20 Aug '15

After 6 months of resolving this situation: http://lists.sip-router.org/pipermail/sr-users/2015-February/086980.html Without touching any configs, my server is back to a failure situation I'm not sure if I'm facing the exact same thing, but for some strange reason, without introducing any modification in the servers, one leg is not hearing voice (outgoing traffic), while incoming rtp traffic goes fine, so I have to assume another time that some Update has destroyed it I've mostly done the same checks as before. I started to assume that rtp-engine could be failing by the same cause, but I have updated to the latest version and still the same issue This has happened from 2-3 day to now. Any ideas on how to start trying to find the cause? Versions: Kamailio 4.1.4 RTP Engine: 4.1.0.0 Kind regards, *Manuel Camargo* Teléfono: 638000836 eMail: sir.louen(a)gmail.com

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Mitel 5000 and 3000
by Fred Posner 20 Aug '15

20 Aug '15

I rarely integrate these with Kamailio and am having some "resistance" in assisting. I was wondering if anyone would be willing to share any configuration recommendations for Mitel when using Kamailio as a SIP trunk (outbound/inbound). Sincerely, Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

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Re: [SR-Users] Mitel 5000 and 3000
by rahul.ultimate 20 Aug '15

20 Aug '15

Hi Fred I too am new to Mitel products but the attached pic might be of some use to you. Thanks RM Sent from Samsung Mobile -------- Original message -------- From: Fred Posner <fred(a)palner.com> Date:18/08/2015 7:08 PM (GMT+05:30) To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Mitel 5000 and 3000 On 08/18/2015 09:23 AM, Alex Balashov wrote: > Are there any specific issues you're encountering? > Mostly hold issues where they try to speak directly with the media server vs the sip server; as well as general audio issues in/out. Most of these seem to be acl related with a set-up issue on the Mitel. I don't work with Mitel enough to discuss in "Mitel speak" what to change. Apparently RTP ports is a different language to the people I am working with. =) --fred _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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new SIP account can not be created ... DB Error : connect failed
by Rainer Böing 20 Aug '15

20 Aug '15

Hello, want to build the following account about Ipel.Org . My old following account no longer works. Iptel.org 19.11.2013 E-Mail rainer.boeing(a)t-online.de Username: boe.rai(a)iptel.org <mailto:boe.rai@iptel.org> Password: xd7ghg Sip address: <sip:boe.rai@iptel.org> sip:boe.rai@iptel.org I can also create no new no account . Please Help !!! clamp THANKS Greetings from Germany Rainer Böing Disclaimer: Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

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Change ports for outgoing RTP packets in RTPproxy
by Jean-Marie Baran 19 Aug '15

19 Aug '15

Hi, Is there a way to change the ports were RTPproxy sends outgoing packets ? By rewriting the SDP before it gets to RTPproxy, or anything else ? 1 - Kamailio receives the SDP 2 - It somehow transmits the contact information to RTPproxy (ip + ports) 3 - The client start sending packets to RTPproxy which in turn relays them to the contact How to change the ports in step 2 ? Sincerely, -- *Jean-Marie Baran*

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how to use seturi with varibales in kamailio
by SureshTummala 19 Aug '15

19 Aug '15

Hi, I have the following questions 1. How to remove ruri which is already added to the destination set2. How to rewritehost in ruri with variables3. How to change the To header in INVITE which are generated by t_relay based on destination set Regards,Suresh Tummala

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Kamailio with NATted RTPproxy
by Jean-Marie Baran 18 Aug '15

18 Aug '15

Hi, I am trying to have RTPproxy works with Kamailio behind a NAT. As it seems RTPproxy cannot handle NAT natively, I installed RTPproxy 1.2.1 by miconda: https://github.com/miconda/rtpproxy Now I am able to make calls, but neither the video nor the audio are present. Wireshark showed that the RTP packets were sent to wrong IPs (private IPs). And this was because of the SDP which contained the wrong IP (fields o= and c=). Then I tried to play with rtpproxy_manage(), and Kamailio replaced the IP in o= and c= by its advertising IP (i.e. its public IP), so the packets from the client behind the NAT were sent to the public address of Kamailio (and then dropped). My question here is what should be the addresses in the SDP when redirected from Kamailio to the client ? Here is the topology when receiving an ACK with SDP: Client <--(2)-- Kamailio <--(1)-- Router (NAT) <--(1)-- Public Internet <--(1)-- Third party router (21.24.12.24) (1): SDP o= 21.24.12.24, c= 21.24.12.24 (2): SDP o= ??, c= ?? (Should it be Kamailio public IP (actually Router IP), or Kamailio private IP, or third party SIP server public IP ?) Cheer, -- *Jean-Marie Baran*

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rtpengine rewrite sdp for second time
by Virmantas Variakojis 18 Aug '15

18 Aug '15

Hi Carsten, Thanks for your reply! How can I call msg_apply_changes() in right way? If I call msg_apply_changes() for example in route[RELAY] i get error: ERROR: textopsx [textopsx.c:157]: msg_apply_changes_f(): invalid usage - not in request route if I do everything as you suggested in onreply_route (after I get for example SIP 500 message) it doesn't work and my SDP remains unchanged. Thanks! > > *From: *Carsten Bock <carsten(a)ng-voice.com> > *Subject: **Re: [SR-Users] rtpengine rewrite sdp for second time* > *Date: *17 Aug 2015 09:49:12 EEST > *To: *"Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org > > > *Reply-To: *"Kamailio $SER$ - Users Mailing List" < > sr-users(a)lists.sip-router.org> > > Hi Virmantas, > > you could try to one or a combination of the following: > - explicitly delete the first session by calling rtpengine_manage() in > the failure route > - call msg_apply_changes() from Textops before calling > rtpengine_manage() the second time (so the already made changes are > applied and the message gets reparsed) > > Thanks, > Carsten > > 2015-08-16 20:13 GMT+02:00 Virmantas Variakojis > <virmantas.variakojis(a)gmail.com>: > > Hi, > > I have a situation like this: > I have kamailio with 3 NIC's. I'm sending INVITE to first provider over > NIC1. Before sending INVITE I use rtpengine_manage to rewrite sdp body with > IP that is on NIC1: > rtpengine_manage ("replace-origin replace-session-connection > direction=intvlan direction=prov1 ICE=remove"); > > If first provider fails i want to send INVITE for the same call to second > provider. But if i use like this: > rtpengine_manage ("replace-origin replace-session-connection > direction=intvlan direction=prov2 ICE=remove"); > or (because ip of prov1 is allready set): > rtpengine_manage ("replace-origin replace-session-connection > direction=prov1 > direction=prov2 ICE=remove"); > rtpengine ignores my changes as said in documentation: "The direction must > only be specified in for initial SDP offer; answers or subsequent offers > can > omit this option." > > and if I try to set ip like this: > rtpengine_manage ("replace-origin replace-session-connection > media-address=X.X.X.X ICE=remove"); > SDP body is doubled: it leaves old values and adds the new one. > > What i'm doing wrong or how can I rewrite SDP for second time? Thanks in > advance for answer! > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Office +49 40 5247593-0 > Fax +49 40 5247593-99 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

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Routing re-INVITEs after phone port change
by David Cunningham 18 Aug '15

18 Aug '15

Hello, We have a scenario like this: SUA -> Kamailio with registrar module -> Asterisk A call from the SUA is set up with SIP timers, and after 15 minutes Asterisk sends a re-INVITE to Kamailio to forward on to the SUA. That re-INVITE has a RURI with the address and port of the SUA at the time the call started. Now if the SUA re-registers after the call starts and before the re-INVITE, and is on a new address or port number, then the re-INVITE never gets to the phone. Obviously Kamailio should send the re-INVITE to the new address/port, but is not. The re-INVITE is routed using the lookup() function. Can anyone suggest why this is happening? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

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Kamailio/Asterisk Integration - MySQL driver error
by Ben Fitzgerald 17 Aug '15

17 Aug '15

Has anyone seen this error before? " ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Host '192.168.xxx.xxx is blocked because of many connection errors; unblock with 'mysqladmin flush-hosts'" I fixed it using flush hosts but I'm curious why this popped up? Is there a way to limit connections for Kamailio and why am I reaching that limit with *only 2* SIP endpoints? This is not a heavily utilized database by any means. Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600

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Siremis docker
by Sylvain Boily 17 Aug '15

17 Aug '15

Hello, I did a docker image for siremis, if some people want to play easily like me with kamailio and siremis ;) Info here : https://github.com/sboily/dockerfile/tree/master/siremis or : https://hub.docker.com/r/quintana/siremis/ Have a good day. Sylvain

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Edge Proxy, how is it supposed to work?
by Daniel Tryba 17 Aug '15

17 Aug '15

The documentation for outbound contains sample configs for edge proxy and registrar configs: http://kamailio.org/docs/modules/4.2.x/modules/outbound.html I tried to implement these but I don't see anything happen concerning the outbound module. What is supposed to happen to make it work? The main problem I have is that calls for the endpoints coming in from the registrar fail to get past the loose_routing check and fail with a 500 in the default case of the switch. the R-URI contains username/externalip/port from the endpoint, there is 1 route header (containing the edgeproxy ip) but no username/flow-token. So per http://kamailio.org/docs/modules/4.2.x/modules/rr.html#loose-route-id loose_routing returns false: "If a Route: header is found, the function returns 1 and behaves as described in section 16.12 of RFC 3261. There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE." How should the Path header look when registering? According to http://kamailio.org/docs/modules/4.2.x/modules/path.html#idp3043008 there should be a flowtoken inserted, this is not the case. Just a plain Path: <sip:ip;lr> is added. So when is outbound required? Can anybody confirm that the examples work?

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db_mongodb compiling errors
by José Seabra 17 Aug '15

17 Aug '15

Hello all, I'm trying to build db_mongodb module but i'm facing some problems, please take a look at logs messages below: make CC (gcc) [M db_mongodb.so] mongodb_connection.o In file included from mongodb_connection.c:24: *mongodb_connection.h:26:20: warning: mongoc.h: No such file or directory* *mongodb_connection.h:27:18: warning: bson.h: No such file or directory* In file included from mongodb_connection.c:24: mongodb_connection.h:37: error: expected specifier-qualifier-list before ‘mongoc_client_t’ mongodb_connection.c: In function ‘db_mongodb_new_connection’: mongodb_connection.c:48: warning: implicit declaration of function ‘mongoc_init’ mongodb_connection.c:49: error: ‘km_mongodb_con_t’ has no member named ‘con’ mongodb_connection.c:49: warning: implicit declaration of function ‘mongoc_client_new’ mongodb_connection.c:50: error: ‘km_mongodb_con_t’ has no member named ‘con’ mongodb_connection.c: In function ‘db_mongodb_free_connection’: mongodb_connection.c:78: error: ‘km_mongodb_con_t’ has no member named ‘con’ mongodb_connection.c:79: warning: implicit declaration of function ‘mongoc_client_destroy’ mongodb_connection.c:79: error: ‘km_mongodb_con_t’ has no member named ‘con’ make: *** [mongodb_connection.o] Error 1 [root@ccp1 db_mongodb]# locate bson.h /usr/local/include/libbson-1.0/bson.h /usr/local/src/mongo-c-driver/src/libbson/src/bson/bson.h /usr/local/src/mongo-c-driver-1.1.10/src/libbson/doc/html/streaming-bson.html /usr/local/src/mongo-c-driver-1.1.10/src/libbson/src/bson/bson.h [root@ccp1 db_mongodb]# locate mongoc.h /usr/local/include/libmongoc-1.0/mongoc.h /usr/local/src/mongo-c-driver/src/mongoc/mongoc.h /usr/local/src/mongo-c-driver/tests/test-libmongoc.h /usr/local/src/mongo-c-driver-1.1.10/src/mongoc/mongoc.h /usr/local/src/mongo-c-driver-1.1.10/tests/test-libmongoc.h I have installed mongo-c-driver successfully and the header files(bson.h, mongoc.h) are located at /usr/local/src/mongo-c-driver/src/mongoc. Can someone help me? -- Cumprimentos José Seabra [image: ITCENTER Store] <http://store.itcenter.com.pt/>[image: ITCENTER Helpdesk] <http://support.itcenter.com.pt/otrs/customer.pl>[image: ITCENTER Facebook] <https://www.facebook.com/itcenterpt>[image: ITCENTER Linkedin] <https://www.linkedin.com/company/itcenter>[image: ITCENTER Twitter] <https://twitter.com/ITCenter_PT>

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rtpengine rewrite sdp for second time
by Virmantas Variakojis 17 Aug '15

17 Aug '15

Hi, I have a situation like this: I have kamailio with 3 NIC's. I'm sending INVITE to first provider over NIC1. Before sending INVITE I use rtpengine_manage to rewrite sdp body with IP that is on NIC1: rtpengine_manage ("replace-origin replace-session-connection direction=intvlan direction=prov1 ICE=remove"); If first provider fails i want to send INVITE for the same call to second provider. But if i use like this: rtpengine_manage ("replace-origin replace-session-connection direction=intvlan direction=prov2 ICE=remove"); or (because ip of prov1 is allready set): rtpengine_manage ("replace-origin replace-session-connection direction=prov1 direction=prov2 ICE=remove"); rtpengine ignores my changes as said in documentation: "The direction must only be specified in for initial SDP offer; answers or subsequent offers can omit this option." and if I try to set ip like this: rtpengine_manage ("replace-origin replace-session-connection media-address=X.X.X.X ICE=remove"); SDP body is doubled: it leaves old values and adds the new one. What i'm doing wrong or how can I rewrite SDP for second time? Thanks in advance for answer!

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multi-process not multi-threading
by Ali Taher 14 Aug '15

14 Aug '15

Hello, Can you please help me to well understand this statement : Kamailio is a multi-process application not multi-threading Does this mean that if we have a machine with one processor of 8 cores , kamailio cannot process more than 1 call at the same time ? Any help would be appreciated. Regards, Ali

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Handling Hold
by Fred Posner 14 Aug '15

14 Aug '15

Hello all, Always fun integrating with proprietary pbx's... is there a way that is "generally accepted" as handling hold invites from devices such as Mitel or Toshiba? We're not proxying the media, so these devices are trying to send an Invite to the endpoint when calls are placed on hold. Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

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CNXCC
by Keith 14 Aug '15

14 Aug '15

Hi All, I have CNXCC setup, it is kind of working but when I run kamcmd cnxcc.check_client the call information is still there even when the call is disconnected. Is this right? I basically want to make sure a customer can only have x number of simultaneous calls. Thanks Keith

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Re: [SR-Users] cnxcc
by Keith 13 Aug '15

13 Aug '15

Hi Guys, Trying to use cnxcc but cnxcc.so is not in the modules folder. What is the easiest way to install? Thanks Keith

1 0

Problems investigating a crash in the siptrace module (4.3.1)
by Sven Neuhaus 13 Aug '15

13 Aug '15

Hi, we're experiencing crashes in the siptrace.so module with Kamailio 4.3.1. I looked at the core dump with gdb but it complained about missing debugging symbols so all I could see was that it called do_action() before branching into an unknown function in the siptrace module. I then installed the kamailio-dbg package to get the debugging symbols. This had the weird effect that get the error "Cannot access memory at ..." in GDB and no longer see *any* function names (previously only the names in the siptrace module were missing. Any idea whats going on? # gdb /usr/sbin/kamailio core GNU gdb (GDB) 7.4.1-debian Copyright (C) 2012 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/sbin/kamailio...Reading symbols from /usr/lib/debug/.build-id/e7/8dc57031d294f320b360d8f2f875a005db757e.debug...done. done. [New LWP 31426] Cannot access memory at address 0x3840f881f9c13908 Cannot access memory at address 0x3840f881f9c13900 (gdb) list 1771 main.c: No such file or directory. (gdb) info threads Id Target Id Frame * 1 LWP 31426 0x00007fbc89403f0f in ?? () (gdb) bt full #0 0x00007fbc89403f0f in ?? () No symbol table info available. #1 0x0800011000d46460 in ?? () No symbol table info available. #2 0xffffffff00000001 in ?? () No symbol table info available. #3 0x0000000000d63230 in ?? () No symbol table info available. #4 0x00000000d59d000a in ?? () No symbol table info available. #5 0x0000000000000000 in ?? () No symbol table info available. # dpkg -l kamailio\* Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Inst/Conf-files/Unpacked/halF-conf/Half-inst/trig-aWait/Trig-pend |/ Err?=(none)/Reinst-required (Status,Err: uppercase=bad) ||/ Name Version Architecture Description +++-==============================-====================-====================-================================================================= ii kamailio 4.3.1+wheezy amd64 very fast and configurable SIP proxy un kamailio-berkeley-modules <none> (no description available) un kamailio-carrierroute-modules <none> (no description available) un kamailio-cpl-modules <none> (no description available) ii kamailio-dbg:amd64 4.3.1+wheezy amd64 very fast and configurable SIP proxy [debug symbols] un kamailio-ldap-modules <none> (no description available) un kamailio-lua-modules <none> (no description available) un kamailio-mysql-modules <none> (no description available) un kamailio-perl-modules <none> (no description available) ii kamailio-postgres-modules:amd6 4.3.1+wheezy amd64 PostgreSQL database connectivity module for Kamailio un kamailio-presence-modules <none> (no description available) un kamailio-python-modules <none> (no description available) un kamailio-radius-modules <none> (no description available) un kamailio-snmpstats-modules <none> (no description available) un kamailio-tls-modules <none> (no description available) un kamailio-unixodbc-modules <none> (no description available) un kamailio-xml-modules <none> (no description available) un kamailio-xmpp-modules <none> (no description available) Regards, -Sven

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Kamailio SCTP with CentOS
by Mellow Yellow 13 Aug '15

13 Aug '15

Hey guys, I've been pulling out my hair for a few days trying to get SCTP with Kamailio working in CentOS 7... with no luck at all. When attempting to start Kamailio via the 'kamctl start' command I get the following error: INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed When attempting to start Kamailio via the 'systemctl start kamailio' command, I get the following: Job for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. When checking the 'systemctl status kamailio -l' command, I get the following: kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Thu 2015-08-13 00:42:51 AEST; 14s ago Process: 19229 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: INFO: usrloc [hslot.c:51]: ul_init_locks(): locks array size 1024 Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: ERROR: ctl [ctl.c:273]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: ERROR: <core> [sr_module.c:945]: init_mod(): Error while initializing module ctl (/usr/lib64/kamailio/modules/ctl.so) Aug 13 00:42:51 Elastix5.workgroup /usr/sbin/kamailio[19242]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 13 00:42:51 Elastix5.workgroup kamailio[19229]: Starting kamailio: Aug 13 00:42:51 Elastix5.workgroup systemd[1]: kamailio.service: control process exited, code=exited status=1 Aug 13 00:42:51 Elastix5.workgroup systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Aug 13 00:42:51 Elastix5.workgroup systemd[1]: Unit kamailio.service entered failed state. So I started to think there was an issue with the SCTP config or whether SCTP was enabled in CentOS. However, everything looks fine: # cat /proc/modules sctp 217455 2 - Live 0xffffffffa03ed000 # /sbin/lsmod | grep sctp sctp 217455 2 libcrc32c 12644 2 xfs,sctp # more /proc/sys/net/sctp/addip_enable 1 # checksctp SCTP supported SCTP modules installed properly: Package lksctp-tools-devel-1.0.13-3.el7.x86_64 already installed and latest version Package lksctp-tools-doc-1.0.13-3.el7.x86_64 already installed and latest version Package lksctp-tools-1.0.13-3.el7.x86_64 already installed and latest version Running the ' sctp_test' command between two servers, SCTP is working fine... I can capture SCTP packets between the servers. I've left the Kamailio.cfg file as is... with the only modification being made to enable the SCTP module to start: enable_sctp = 1 Checking 'kamailio -v' USE_SCTP is enabled: # kamailio -v version: kamailio 4.3.1 (x86_64/linux) f38e67 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: f38e67 compiled on 18:17:56 Jul 20 2015 with gcc 4.8.2 I've tried installing Kamailio with the precompiled rpms for CentOS 7, and I also followed some guides on installing in via GIT & compiling it myself... however the issue still remains. Any help would be greatly appreciated as I've completely run out of ideas, and there isn't much info on the web about Kamailio & SCTP. Thanks, Rob

2 2

upgrade V4.2.5 to V3.1 -Build error
by Yasin CANER 13 Aug '15

13 Aug '15

3 3

XCap Server + Postgre
by Igor Olhovskiy 12 Aug '15

12 Aug '15

Hi! I'm experience really strange behavior of Kamailio. Right now it's using postgre database When trying to implement XCAP server capabilities, according to http://kamailio.org/docs/modules/4.3.x/modules/xcap_server.html#idp76784 on a block if (!www_authorize("xcap", "subscriber")) { www_challenge("xcap", "0"); exit; } While trying to test it via browser (user test, pass test) I've got these errors in log Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012] Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012] Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012] Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: ERROR: db_postgres [km_dbase.c:247]: db_postgres_submit_query(): 0x7f3fb0660b10 PQsendQuery Error: ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012 Query: select password from subscriber where extension='test' AND user_context='xcap' Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Without this block, I don't get user auth. Is it a bug or it's mine misconfiguration? -- Best regards, Igor

2 3

kamailio as SIP Agent
by Sandeep Chakravarthi 12 Aug '15

12 Aug '15

Hi, We have a requirement with one of our telco We are using asterisk in our servers and we are planning to implement SIP-I protocol and we choosed kamailio for it. In Kamailio website, i came to know that kamailio will be supporting both SIP-I and SIP-T protocols Below is what we need and pls confirm whether it is possible or not? Asterisk PBX <-------> Kamailio <--------> Telco MSC Telco will be forwarding the calls to kamailio on sip-i protocol and kamailio server has to forward the calls to our Asterisk server by converting sip-i to standard sip protocol Similiarly Asterisk will be initiating sip call to kamailio server and kamailio server should convert it into SIP-I and should forward the call to Telco MSC 1. I am able to establish the SIP trunk [sending OPTIONS from asterisk and kamailio acknowledges with 200 OK] between Asterisk and Kamailio using dispatcher module in kamailio and sip.conf in asterisk. How to establish the SIP trunk between kamailio and telco MSC? [Generally MSC will act as SIP server and kamalio should send OPTIONS packet and MSC will acknowledges with 200 OK] My telco MSC has only provided me the MSC SIP IP and there were no username/passwords provided. Means i need to use IP based authentication for the SIP Trunk establishment. In Kamailio how to achieve it? Please help and any suggestions/feedback will be highly appreciated and thankful Regards, Sandeep

2 12

sdp manipulation in connection with rtpengine_offer/answer calls
by Juha Heinanen 12 Aug '15

12 Aug '15

it was not clear to me based on rtpengine module readme if rtpengine_offer/answer are same kind of sdp manipulation functions as, for example, sdpops module functions, i.e., will rtpengine functions pass to rtpengine the sdp as it is after possible earlier sdpops function calls and will possible sdpops functions calls after rtpengine function call keep on manipulating the sdp returned from rtpengine function call. -- juha

4 16

RTPProxy incorrect port 0 in reply from rtp proxy
by Grant Bagdasarian 12 Aug '15

12 Aug '15

Hello, I've been load testing kamailio(4.2) with rtpproxy(2.0) using sipp, and when I hit 50+ CPS I start getting the following error: "incorrect port 0 in reply from rtp proxy". I'm running 4 instances of rtpproxy on the same instance as kamailio. Could somebody tell me why I'm receiving this error? Do I need to run more instances of rtpproxy? # <#----------RTPPROXY Parameters----------#> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7721 udp:127.0.0.1:7722 udp:127.0.0.1:7723 udp:127.0.0.1:7724") In request_route: if (is_method("INVITE")) { record_route(); if (has_body("application/sdp")) { rtpproxy_manage(); } } In onreply_route: if (has_body("application/sdp")) { rtpproxy_manage(); } rtpproxy -s udp:127.0.0.1:7721 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 10.0.0.1 rtpproxy -s udp:127.0.0.1:7722 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy1.pid -l 10.0.0.1 rtpproxy -s udp:127.0.0.1:7723 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy2.pid -l 10.0.0.1 rtpproxy -s udp:127.0.0.1:7724 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy3.pid -l 10.0.0.1 Regards, Grant

1 1

Re: [SR-Users] LCR Routing - Cost based routing
by DanB 12 Aug '15

12 Aug '15

Hi Alexandru, Well the concept is simple: on INVITE, you will issue a special format evapi request. You will receive the list of suppliers in the reply (which is received in a separate route) and you continue processing from there. These suppliers are simple tags so your creativity will be the limit of what you can do with them - eg: direct ip addresses received or tags of suppliers which you can further translate with drouting module or any alias related one. For more information on LCR please find this year's presentation on Kamailio World (which I strongly recommend to be attended): https://www.youtube.com/watch?v=Hsvcwleb-fY You can also find the kamailio configuration which I have used in the worshop here: https://github.com/cgrates/cgrates/tree/master/data/tutorials/kamevapi/kama… Let me know if further questions (or join our irc or mailing list so we do not create too much noise here). DanB

3 2

Help with sip balancer
by Bruno 11 Aug '15

11 Aug '15

Hello, i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP registration from VoIP provider and route the calls to the asterisk boxes where an IVR will take care to answer. Here's my network topology: +---> [asterisk1] [public_ip] | 10.50.10.131 [router] <---NAT---> [kamailio] <---+ 10.50.10.1 10.50.10.120 | +---> [asterisk2] 10.50.10.132 In my setup i planned to use UAC and DISPATCHER modules. I started from the "kamailio-basic.cfg" and added some extra lines to handle UAC and DISPATCHER. All is working fine when i do a test call from a softphone inside network 10.50.10.0/24. When a call is coming from the sip carrier, troubles occurs because asterisk boxes are sending their internal ip in SDP. I understand that i need to rewrite SDP in that case, but i actually don't know how/where. I've attached kamailio configuration and a sip trace taken with sngrep where the problem is visible. For security reasons, i would like to force the RTP through RTPProxy. I'm missing something, and need your help me to understand my errors. Best Regards, Bruno

3 3

Trailing newline with {re.subst,expression}
by Daniel Tryba 11 Aug '15

11 Aug '15

I'm unable to get rid of the \n in "multiline" content of a var. xlog("var: $var(sdp)"); $var(sdp)=$(var(sdp){re.subst,/^a=rtcp:.*//}); xlog("var: $var(sdp)"); a=sendrecv#015#012a=rtcp:31543#015#012a=X-foo:bar becomes: a=sendrecv#015#012#012a=X-foo:bar Using the s flag results in no substitution at all, /^a=rtcp:.*//s should result in a=sendrecv#015#012 according to http://www.kamailio.org/wiki/cookbooks/4.3.x/transformations#resubst_expres… Is this a bug or a feature? If it is a feature, how to get rid of the newline (without additional regexps substitutions).

1 0

force_send_socket/$fs behaviour when binding to INADDR_ANY
by Alex Balashov 11 Aug '15

11 Aug '15

Hello, When binding Kamailio to 0.0.0.0, Kamailio no longer recognises any specific IP address homed on the system as being a "socket" for purposes of forcing traffic out of any specific interface (i.e. setting $fs), or any other purpose for which ingress and egress "sockets" are tracked (e.g. double RR). As I understand it, this is because Kamailio considers only addresses explicitly specified via the 'listen' core config directive to be valid "send socket" arguments. Kamailio _will_ accept addresses already attached to the interface if it is bound interface-wise, e.g. listen=eth1:5060 but not when addresses are added to the interface without restarting the proxy: # ip addr add dev eth1 172.30.110.10/24 Aug 10 17:31:35 centosity6 /usr/local/sbin/kamailio[28627]: WARNING: pv [pv_core.c:2285]: pv_set_force_sock(): no socket found to match [udp:172.30.110.10:5060] Is there any straightforward way to modify this behaviour, so that it would be possible to dynamically add addresses to existing physical interfaces and get Kamailio to utilise them as if they were bound at boot time? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

Re: [SR-Users] Unable to limit call duration with timeout_avp
by Admin 10 Aug '15

10 Aug '15

Thanks Alex. That worked.

1 0

Unable to limit call duration with timeout_avp
by Admin 10 Aug '15

10 Aug '15

Hi, I am using dialog time_avp() parameter to limit the call duration to a configured number of seconds but unfortunately when the number is reached, i get a warning but the call is not torn down. Here is the output from the log file 3(3711) DEBUG: <script>: return value = 30 -----> $avp(i:10) 5(3713) WARNING: dialog [dlg_handlers.c:1426]: dlg_ontimeout(): timeout for dlg with CallID '1-3743(a)192.168.1.100' and tags '37431' '37401' 5(3713) DEBUG: dialog [dlg_hash.c:832]: dlg_unref(): unref dlg 0x7f32ee941140 with 2 -> 0 5(3713) DEBUG: dialog [dlg_hash.c:832]: dlg_unref(): ref <=0 for dialog 0x7f32ee941140 5(3713) DEBUG: dialog [dlg_hash.c:320]: destroy_dlg(): destroying dialog 0x7f32ee941140 (ref 0) 5(3713) DEBUG: dialog [dlg_hash.c:336]: destroy_dlg(): removed timer for dlg 0x7f32ee941140 [773:2813] with clid '1-3743(a)192.168.1.100' and tags '37431' '37401' Here is my config. file related to dialog module: #!define DLG_FLAG 4 loadmodule "dialog.so" modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "default_timeout", 3600) # 1 hour timeout modparam("dialog", "db_mode", 0) modparam("dialog", "hash_size", 2048) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "default_timeout", 3600) # 1 hour timeout modparam("dialog", "timeout_avp", "$avp(i:10)") if (is_method("INVITE")) { #$var(rv) = perl_exec("max_allowed_time"); $var(rv) = 30; $avp(i:10) = $var(rv); xlog("L_DBG", "return value = $var(rv)\n"); setflag(FLT_ACC); # do accounting setflag(DLG_FLAG); } Maybe i am missing some configuration related to the dialog module. Any help is very much appreciated. Thanks.

2 2

Handling a non-standard SIP port
by Jean-Marie Baran 10 Aug '15

10 Aug '15

Hi, This is a question for the sake of comprehension, as I actually solved the problem with a workaround. I am trying to get Kamailio working with a third party SIP provider. The SIP server of this provider pings my Kamailio with OPTIONS messages. Being located in a sub-network, the SIP messages are relayed to a router (not a SIP router, just a gateway to the Internet) which is not listening/sending on port 5060 but on 13377. When Kamailio answer to OPTIONS, it send back the message to the router on the port 5060 which is wrong. Here is the topology: *Kamailio* *Router* *SIP Server* /my.sip.domain 174.37.217.1 197.17.144.4/ | | ? <-------------------- 5060 | | 5060 <---------------- 13377 | | | 5060 ----------------−> 5060 | | | | X | You can see that, Kamailio answers to the wrong port, but I think this is somewhat coherent since the SIP message says that the port 5060 is in use. But still, I would have liked that the correct port was used: *Kamailio* *Router* *SIP Server* /my.sip.domain 174.37.217.1 197.17.144.4 / | | ? <-------------------- 5060 | | 5060 <---------------- 13377 | | | 5060 ---------------−> 13377 | | | | ? --------------------> 5060 | But I do not understand where Kamailio is supposed to find the correct port. Below are the transport frames, and SIP frames. Kamailio receives: Internet Protocol Version 4, Src: 174.37.217.1 (174.37.217.1), Dst: 172.16.8.61 (172.16.8.61) Source: 174.37.217.1 (174.37.217.1) Destination: 172.16.8.61 (172.16.8.61) User Datagram Protocol, Src Port: 13377 (13377), Dst Port: sip (5060) Source port: 13377 (13377) Destination port: sip (5060) OPTIONS sip:my.sip.domain:5060 SIP/2.0 Via: SIP/2.0/UDP 197.17.144.4:5060;branch=z9hG4bKb9340cb29ba From: <sip:197.17.144.4>;tag=1797904074 To: <sip:my.sip.domain> Date: Mon, 10 Aug 2015 10:01:51 GMT Call-ID: c7f38580-5c817680-b94-38512c6(a)197.17.144.4 User-Agent: Cisco-CUCM10.5 CSeq: 101 OPTIONS Contact: <sip:197.17.144.4:5060> Max-Forwards: 0 Content-Length: 0 ...and then sends: Internet Protocol Version 4, Src: 172.16.8.61 (172.16.8.61), Dst: 174.37.217.1 (174.37.217.1) Source: 172.16.8.61 (172.16.8.61) Destination: 174.37.217.1 (174.37.217.1) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Source port: sip (5060) Destination port: sip (5060) SIP/2.0 200 OK Via: SIP/2.0/UDP 197.17.144.4:5060;branch=z9hG4bKb9340cb29ba;received=174.37.217.1 From: <sip:197.17.144.4>;tag=1797904074 To: <sip:my.sip.domain>;tag=b27e1a1d33761e85846fc98f5f3a7e58.de7a Call-ID: c7f38580-5c817680-b94-38512c6(a)197.17.144.4 CSeq: 101 OPTIONS Allow: INVITE,ACK,BYE,CANCEL,INFO,OPTIONS,REFER,SUBSCRIBE,NOTIFY Accept: */* Accept-Encoding: Accept-Language: en Supported: Server: kamailio (4.3.1 (x86_64/linux)) Content-Length: 0 We see clearly that the port is wrong here. But in the received SIP frame, everything suggests that the correct port is 5060, and I do not think Kamailio will look at the transport frame to send the response back to the correct port. Am I correct stating this ? Maybe the Via header should contain a port along with the IP address ? I solved the problem by using TCP instead of UDP, as I know that since the TCP connection is maintained, the port would be correct (because imposed by the gateway). I am sorry if the question does not directly relates to Kamailio, but any explanation here would be appreciated, as I am a student trying to learn how all this is working :) Cheers, Jean-Marie.

2 2

rtpengine disabling offers of rtcp
by Daniel Tryba 10 Aug '15

10 Aug '15

Maybe a bit offtopic, but I can't find a rtpengine mailinglist and know the Sipwise guys are reading this list. Since upgrading to 4.2.x and rtpengine instead of rtpproxy I'm seeing problems with calls being terminated by upstream provider after a bit more than 5m. The INVITE process looks okay, the only difference with before as far as I can see is the rtcp attribute in the SDP body. According to http://kamailio.org/docs/modules/4.2.x/modules/rtpengine.html#rtpengine.f.r… and https://github.com/sipwise/rtpengine rtcp offering can be disabled. But neither rtcp-mux-demux nor rtcp-mux-accept have any impact on the adding of the a=rtcp:port attribute in SDP. rtpengine_manage("trust-address replace-origin replace-session-connection rtcp-mux-accept ICE=remove"); or rtpengine_manage("trust-address replace-origin replace-session-connection rtcp-mux-demux ICE=remove"); have no effect op the INVITE to the upstream. kamailio: 4.2.5 rtpengine: 4.1.0.0+0~mr4.1.0.0

3 4

Debian/jessie and 4.3.1 default config doesn't work
by Daniel Tryba 10 Aug '15

10 Aug '15

It looks like the compile defaults of the ctl and mi_fifo module and the default kamailio.conf/kamctlrc conflict. kamailio tries to open: /tmp/buildd/kamailio-4.3.1+jessie/debian/kamailio/var/run/kamailio//kamailio_ctl /tmp/buildd/kamailio-4.3.1+jessie/debian/kamailio/var/run/kamailio/kamailio_fifo and will fail to run, unless explicitly set to conf paths are set to /var/run/kamailio/kamailio_(ctl|fifo)

2 3

Re: [SR-Users] LCR Routing - Cost based routing
by DanB 10 Aug '15

10 Aug '15

Guys, Since I saw the thread growing, I am around if you got questions on CGRateS related LCR, fully compatible with Kamailio via evapi module. DanB

2 1

Running Kamailio 4.3 on Debian 8.1 and systemd
by Grant Bagdasarian 10 Aug '15

10 Aug '15

Hello, I just installed the latest stable release of Kamailio (I believe 4.3.1) on a debian 8.1 system using the install guide (http://www.kamailio.org/wiki/install/4.3.x/git). Starting kamailio didn't work using /etc/init.d/kamailio start. I got the following error: # /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceFailed to start kamailio.service: Unit kamailio.service failed to load: No such file or directory. failed! After some googling I found out I have to add a new file named kamailio.service to the /etc/systemd/system/ directory. Kamailio.service: [Unit] Description=Kamailio SIP Server [Service] Type=forking PIDFile=/var/run/kamailio.pid ExecStart=/usr/local/sbin/kamctl start ExecRestart=/usr/local/sbin/kamctl restart ExecStop=/usr/local/sbin/kamctl stop [Install] WantedBy=multi-user.target Now I can start kamailio using the command: service kamailio start Is this the correct way of doing this? Does the file need any changes? Regards, Grant

1 0

Wrong Call-ID and tags in response to OPTIONS
by Jean-Marie Baran 10 Aug '15

10 Aug '15

Hi, I am trying to get Kamailio working with a third party SIP provider. This provider pings Kamailio with OPTIONS messages, but Kamailio does not answer correctly: the Call-ID and the tags change where they should not. Here is an example: Third party SIP server: OPTIONS sip:false.sip.provider:5060 SIP/2.0 Via: SIP/2.0/UDP 197.17.144.4:5060;branch=z9hG4bK8e45ac1463a From: <sip:197.17.144.4>;tag=1669836943 To: <sip:false.sip.provider> Date: Fri, 07 Aug 2015 12:34:37 GMT Call-ID: a012e080-5c41a5ce-8e5-38512c6(a)197.17.144.4 User-Agent: Cisco-CUCM10.5 CSeq: 101 OPTIONS Contact: <sip:197.17.144.4:5060> Max-Forwards: 0 Content-Length: 0 Kamailio's answer: SIP/2.0 200 OK Via: SIP/2.0/UDP 197.17.144.4:5060;branch=z9hG4bK93e5f072f5f;received=174.37.217.1 From: <sip:197.17.144.4>;tag=298911055 To: <sip:false.sip.provider>;tag=b27e1a1d33761e85846fc98f5f3a7e58.8c24 Call-ID: 2defcb00-5c41aa17-93f-38512c6(a)197.17.144.4 CSeq: 101 OPTIONS Server: kamailio (4.3.1 (x86_64/linux)) Content-Length: 0 You can see that the Call-ID change from "a012e080-5c41a5ce-8e5-38512c6" to "2defcb00-5c41aa17-93f-38512c6". Likewise, the From tag change ("1669836943" to "298911055"). What could be the reason for this ? Moreover I think Kamailio should add the field Allow (with INVITE,ACK,BYE,CANCEL,INFO,OPTIONS,REFER,SUBSCRIBE,NOTIFY), but it does not. I should mention that I had to modify /kamailio.cfg/ to have Kamailio answer to OPTIONS messages (I added *the code in bold*): if (*!is_method("OPTIONS") && *!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { *#* sl_send_reply("200","Keepalive"); *options_reply();* exit; } Maybe this hack replaces an expected behavior that I failed to see/configure ? Thanks, Jean-Marie.

3 3

Kamailio 4.3 and XCAP
by Safdar Khan 10 Aug '15

10 Aug '15

Hello sir/mam How can i configure xcap with kamailio 4.3. I am finding difficulties in routing logic of xcap_server module amd xcap_client module. Will you please share proper documentation of these two technology together.

1 0

LCR Routing - Cost based routing
by Arun Kumar 09 Aug '15

09 Aug '15

Hi is there any way to achive following usecase using carrier route / LCR / droude module i have 10 different termination gateways, each gateways will have more than 10000 destinations and each gateways will have different rate, for example 91 Destination ( there will be 10000 entries like 91 ) gw1 gw2 gw3 gw4 gw5 0.001 0.007 0.003 0.002 0.003 now need to logic to load the these destination with cost and sort gateways in order of longest match of dialled number,cost, priority for the dialled destination, which will not use too much CPU and memory Regards Arun

4 9

Kamailio crash when imc_list_rooms
by SamyGo 09 Aug '15

09 Aug '15

Hi, I'm working with IMC module and noticed that when I execute fifo command "imc_list_rooms" it crashes. [root@kamailio75 ]# kamctl fifo imc_list_rooms my Kamailio version is 4.2.1 Right now the way IMC module works is that a SIP UA has to send a chat room command to create a new room, get participants added etc. If I try to add users directly from DB the kamailio don't recognize the room or participants unless restart. Would it be possible to reload the DB tables for IMC via MI command, a big plus if we can do room creation, user addition via MI as well ! Thanks, Sammy

1 0

acc Module missing BYE in database
by Dirk Teurlings - SIGNET B.V. 07 Aug '15

07 Aug '15

Hi again, Running kamailio 4.2.5 with the acc module to do accounting in a MySQL database. We are now running into the issue that sometimes there is no BYE line when there is an INVITE. Digging into this I noticed that the dialog:end event does get triggered, and that CNXCC consumes the call properly. So the BYE message does get sent, but acc doesn't seem to pick it up some of the times. It occurs around maybe once every 100 calls. Anybody got any ideas about this? -- Cheers, Dirk Teurlings

2 3

Kamailio example trace needed UAR/UAA
by Jan Devai 07 Aug '15

07 Aug '15

Hi Experts, we have trouble with the last SIP Registration from I-CSCF to S-CSCF, see below: *The flow is as follows:* UE HSS P-CSCF I-CSCF S-CSCF -----------Sip Register-----------> -------Sip Register-------> <--------------------------UAR--------------- --------------------------UAA---------------> ----SIP Register---> Almost a week ago, I've open a thread about this issue called "Error forwarding to SCSCF", and so far we did not get any further with it :-( I was wondering does somebody have similar Signaling that he could share with us? It could be either with an external or OpenHSS, does not matter at this stage. It would definitely move us forward with at least the signaling comparison as a start point. Maybe the UAA answer and anything happening before is wrong. Any reference, similar working trace would be greatly appreciated. Thank you in advance Jan

2 1

About Acc_radius rad_sttaus Function
by Yasin CANER 07 Aug '15

07 Aug '15

3 4

SDP IPv4 has been concatenate - no audio
by Loic Chabert 07 Aug '15

07 Aug '15

Hello everyone, I'm trying put kamailio in front of asterisk server farm. Fow now, 2 asterisk servers are running and i'm trying to make some basic calls between two UACc. All asterisk servers has been ofuscaded from public internet using 10.189.122.0/24 network. All trafic must be passed throught asterisk so RTPproxy is used to (and used for rtp bridging). Kamailio and rtpproxy is running with public IP address, and private ip address (mhomed=1) But a wired thing append on my SDP body: c line have two rtpproxy public addresses concatenate (see my capture attached). Any reason for this ? Only invite method from my asterisk contains 2 publics IP addresses concatenated. Does it mean than rtp_manage as been executed twice ? Thanks in advance. Regards.

3 6

Problem adding json module
by Joao Alves 07 Aug '15

07 Aug '15

Hi, Have just installed the kamailio 4.3 (from github) and was trying to include the additional modules that I require. I've edited the modules.lst and changed the extra modules to compile "include_modules= db_mysql utils json". During compilation (using "make all") getting the following warnings in relation with the json module. CC (gcc) [M json.so] json_funcs.o json_funcs.c: In function 'json_get_field': json_funcs.c:58:2: warning: implicit declaration of function 'is_error' [-Wimplicit-function-declaration] json_funcs.c:64:2: warning: 'json_object_object_get' is deprecated (declared at /usr/local/include/json-c/json_object.h:303) [-Wdeprecated-declarations] And later, when running the config file: ERROR: <core> [sr_module.c:574]: load_module(): could not open module </usr/local/lib64/kamailio/modules/json.so>: libjson-c.so.3: cannot open shared object file: No such file or directory 0(2344) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 195, column 12-20: failed to load module ERROR: bad config file (1 errors) I have already installed json-c modules and confirmed this installation by doing: root@ID14337:/usr/local/kamailio-4.3/kamailio# pkg-config --cflags json-c -I/usr/local/include/json-c root@ID14337:/usr/local/kamailio-4.3/kamailio# pkg-config --libs json-c -L/usr/local/lib -ljson-c root@ID14337:/usr/local/include/json-c# ls arraylist.h json.h json_inttypes.h json_object_private.h linkhash.h bits.h json_c_version.h json_object.h json_tokener.h printbuf.h debug.h json_config.h json_object_iterator.h json_util.h random_seed.h root@ID14337:/usr/local/include/json-c# cd ../../lib root@ID14337:/usr/local/lib# ls libjson-c.a libjson-c.la libjson-c.so libjson-c.so.3 libjson-c.so.3.0.0 pkgconfig python2.6 python2.7 root@ID14337:/usr/local/lib# So any ideas of what could be causing this to fail? Thanks Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

2 6

Kamailio w/ external routing server
by Eric Babcock 07 Aug '15

07 Aug '15

Hey Everyone, I've been banging my head against the desk for days trying to get this working, and I'm sure it's something simple that I'm not doing, but maybe someone else can add some input. Here is the best way to describe my call flow. Customer -> Kamailio -> Routing Server (302) -> Kamailio -> Vendors (from the 302 header) I have tried to configure dispatch, but that hasn't worked out for me. My routing server does the authentication, LRN lookup, and provides the vendor routes in the sip header. So when I pass the call to the routing server, it has to contain the customer information. This setup has to be stateful so I can write CDRs for billing to MySQL. The reason I tried using dispatch, is because I have multiple routing servers I would like to load balance across or even failover if one goes down. Can anyone help me out, or at least help point me in the right direction? Thanks! -Eric -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-w-external-routing-server-… Sent from the Users mailing list archive at Nabble.com.

3 4

Errors when starting Kamailio with Websocket Module
by Babu Raja 07 Aug '15

07 Aug '15

Hello, I incuded the module for websocket into my Kamailio 4.3 on my Ubuntu 12.04 server environement. For installing the packages and websocket I followed the instructions from the following link: http://www.kamailio.org/wiki/packages/debs http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket After starting the Kamailio I got 26 errors like: can not find command "handle_ruri_alias" or Syntax errors for: route[REGISTRAR] => square brackets.. ($du == "") => parameter "==" unknown.. the whole error log I will paste later.. Can anyone help me?

3 4

RLS Notify sent to wrong transport
by M S 07 Aug '15

07 Aug '15

Hi, I have a Kamailio v4.3.0 with SVN Rev. c6aa95 running an RLS based presence setup. Everything works fine when end-user has UDP transport, however, if user has TCP or TLS transport then i aggregated NOTIFY sent by RLS gives send error, -- WARNING: <core> [forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:X.X.X.X:5061, to udp:Y.Y.Y.Y:12345) -- (X.X.X.X is server IP and Y.Y.Y.Y is end user IP). Not sure if it is configuration issue or some bug, so posting it to both mailing list. Please help. Thank you.

2 8

Kamailio 4.3.1 and Debian 7
by Mathieu Boyer 07 Aug '15

07 Aug '15

Hello, i've got big issues since upgrade of Debian packages (debian repository http://deb.kamailio.org/kamailio wheezy)<http://deb.kamailio.org/kamailio> on 4.3.1. Started with rr.so can't load without outbound.so, location.so find version mismatch (6 vs 8) etc ... Have you got a quick workaround in order to make it run properly ? dpkg -l |grep kamailio ii kamailio 4.3.1+wheezy amd64 very fast and configurable SIP proxy ii kamailio-extra-modules:amd64 4.3.1+wheezy amd64 extra modules for Kamailio ii kamailio-mysql-modules:amd64 4.3.1+wheezy amd64 MySQL database connectivity module for Kamailio ii kamailio-outbound-modules:amd64 4.3.1+wheezy amd64 Outbound module for Kamailio ii kamailio-utils-modules:amd64 4.3.1+wheezy amd64 Provides a set utility functions for Kamailio Error logs : Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4083]: ERROR: dispatcher [dispatcher.c:768]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Unknown column 'server_id' in 'field list' Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: usrloc [udomain.c:424]: preload_udomain(): db_query (1) failed Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: usrloc [ul_mod.c:439]: child_init(): child(1): failed to preload domain 'location' Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [sr_module.c:900]: init_mod_child(): Error while initializing module usrloc (/usr/lib/x86_64-linux-gnu/kamailio/modules/usrloc.so) Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [pt.c:329]: fork_process(): init_child failed for process 1, pid 4085, "udp receiver child=0 sock=XXXX:5063"? Thank you for your help BR Mathieu

3 2

Rtpengine/rtpproxy and zrtp
by Alexandru Covalschi 07 Aug '15

07 Aug '15

Sorry if writing to wrong mailing list, I am very limited to traffic now amd don't know if there is any for rtpproxy/rtpengine. My question is - can they support ZRTP at least in pass-through mode? Will rtpengine fail on trying to recognize unknown SDP fields?

2 2

Kamailio v4.2.6 Released
by Daniel-Constantin Mierla 06 Aug '15

06 Aug '15

Hello, Kamailio SIP Server v4.2.6 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.6 (or to 4.3.x series). For more details about version 4.2.6 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/07/kamailio-v4-2-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.1. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

2 3

Kamailio Redis support
by Solutions Solutions 06 Aug '15

06 Aug '15

Hi All, i want to replace MySQL with redis database in kamalio,but in kamalio supports transactions with MySQL ,ie saving data in MySql DB.but while trying to run kamailio with redis ,transaction are not happening,is there any module that i have to enable in kamalio for No SQL databases transaction support.

2 1

Re: [SR-Users] Remote SIP Registration
by Keith 05 Aug '15

05 Aug '15

Hi, I have a carrier who require registration, I have looked at the uac module but can't see how to do this. Anyone point me in the right direction? Thanks Keith

2 10

Difference between corex module's 'alias_subdomains' and 'alias' core parameter
by Iwan Aucamp 05 Aug '15

05 Aug '15

>From the descriptions these seem similar: corex alias_subdomains ( http://kamailio.org/docs/modules/devel/modules/corex.html#idp18017160): Register a domain and all its sub-domains to match the “myself” condition. It can be set many times. Its full format is: 'proto:domain:port', allowing to set restrictions on protocol and port as well. Protocol and port are optional. alias core parameter ( http://www.kamailio.org/wiki/cookbooks/devel/core#alias): Parameter to set alias hostnames for the server. It can be set many times, each value being added in a list to match the hostname when 'myself' is checked. Is the only difference that corex alias_subdomains can include 'proto' ? -- Iwan Aucamp

2 1

Question about how location server_id is supposed to be used.
by Ding Ma 05 Aug '15

05 Aug '15

Looking for some doc that describes how the new columns in 4.3.x location table is supposed to be used. Can find some email discussions about this, but seems no formal doc exists. Would appreciate if someone can provide an explanation. Thanks

3 3

Re: [SR-Users] R: Mutiple branch forking and voicemail
by Daniel-Constantin Mierla 05 Aug '15

05 Aug '15

Indeed usually there are few options for same functionality, good to hear you found a solution. Cheers, Daniel On 03/08/15 14:53, Marino Mileti wrote: > Thanks Daniel, > > I've found another way using a seturi for mediaserver that will be executed > after all branch (using t_load_contacts and t_next_contacts) > In this way I can complete my scenario in the right way > > -----Messaggio originale----- > Da: sr-users [mailto:sr-users-bounces@lists.sip-router.org] Per conto di > Daniel-Constantin Mierla > Inviato: lunedì 3 agosto 2015 12:37 > A: Kamailio (SER) - Users Mailing List > Oggetto: Re: [SR-Users] Mutiple branch forking and voicemail > > Hello, > > On 31/07/15 11:31, Marino Mileti wrote: >> Hello, >> >> I would like to implement a voicemail services but i'm trying >> everything without good results :( >> >> My scenario is: >> >> - A make a parallel forking to B(1001) and C(1002) >> - If timeout, call is routed to voicemail >> >> I've done something like this...very simple >> >> seturi("sip:1001@172.20.40.103"); >> >> append_branch("1002(a)172.20.40.103"); >> route(RELAY) >> >> on failure_route.. >> if (t_check_status("486|408")) { >> $ru = "sip:8000@172.20.40.103:5090"; >> >> route(RELAY); >> >> exit; >> >> } >> >> The scenario is working but on log there are some error... >> t_should_relay_response >> >> I would like that when a branch reach the failure route, all other >> branches will be cancelled..and call is routed to VM. >> Can anyone help me? >> >> > When the failure_route is executed, no branch is active anymore. > > To be able to assist further, you have to send here the errors you get in > the log, to understand what is wrong. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Book: SIP Routing With Kamailio - http://www.asipto.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > --- > Questa e-mail è stata controllata per individuare virus con Avast antivirus. > https://www.avast.com/antivirus > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

Maintaining order of messages in a sip dialog on forwarding
by Daniel Tryba 05 Aug '15

05 Aug '15

Since switching to 4.2.5 (from 3.x) a customer has problems with a buggy user agent (IMHO). What I am seeing now is: provider kamailio customer INVITE1-> INVITE1-> <-Trying <-Trying <-200 OK <-200 OK ACK1-> INVITE2-> INVITE2-> ACK1-> ACK1 arrives 0.007s before INVITE2, but INVITE2 is send 0.004s before ACK1. It looks like this wasn't happening with Kamailio 3.x, it worked fine then (apparently). Can this reordering be (gracefully) prevented somehow (ignoring the fact that during transport reordering can happen)? A Q&D hack looks to be just usleep the reinvite a little. The endpoint is essentially ignoring the reinvite before (and after) the ACK. After a timeout INVITE2 is canceled and after that INVITE1 will be ended with a BYE from provider. kamailio-customer communication is UDP provider-kamailio is tcp when message >= 1300b. INVITE1 is TCP, INVITE2 and ACK1 are UDP.

2 5

VIA header question.
by Chad 04 Aug '15

04 Aug '15

Hi list, I need a little help, I am a business owner trying to get Kamailio up and running as a SIP load balancer. I hired a Kamailio consultant to help me do so, but Kamailio is not working and I am getting conflicting information. My Kamailio consultant and my VOIP provider are telling me 2 different things and I don't know which one is right. Kamailio sends SIP traffic to the VOIP provider with 2 VIA headers like this (in this order): Via: SIP/2.0/UDP 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0. Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. The VOIP provider says that is incorrect because they are supposed to reply back to the topmost VIA header so they reply to the 10.10.10.254 IP (which is not public) and the call ends. The VOIP provider says Kamailio should send the VIA headers like this instead: Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. Via: SIP/2.0/UDP 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0. My Kamailio consultant says the way we are sending it is right and that the VOIP provider is processing the call incorrectly. I read that the SIP proxy is supposed to remove the internal header from the 1st example above based on this RFC: https://tools.ietf.org/html/rfc3261#section-16.7 Item: 3. Via "The proxy removes the topmost Via header field value from the response." If that applies to this situation (which I don't know if it does) then Kamailio should be removing the 10.10.10.254 VIA line and only sending 1 VIA header like this: Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. Which would sort of make the VOIP provider right in that the topmost VIA line would then be the external IP, but how they said to fix it (reversing the VIA lines) is wrong. Does anyone know what the right answer is here? Please let me know. THANKS for your help. -- ^C

3 3

MSILO: Archiving every message
by SamyGo 03 Aug '15

03 Aug '15

Hi, Tried finding anything as weird as I am trying to do on mailing list but couldn't. The idea is to save a copy of the messages on database. MSILO module can keep the message as long as the destination user is offline, or doesn't support method:MESSAGE or expiry timeout. Any ideas will be appreciated. Best Regards, Sammy

2 2

Re: [SR-Users] Mutiple branch forking and voicemail
by Daniel-Constantin Mierla 03 Aug '15

03 Aug '15

Hello, On 31/07/15 11:31, Marino Mileti wrote: > Hello, > > I would like to implement a voicemail services but i'm trying everything > without good results :( > > My scenario is: > > - A make a parallel forking to B(1001) and C(1002) > - If timeout, call is routed to voicemail > > I've done something like this...very simple > > seturi("sip:1001@172.20.40.103"); > > append_branch("1002(a)172.20.40.103"); > route(RELAY) > > on failure_route.. > if (t_check_status("486|408")) { > $ru = "sip:8000@172.20.40.103:5090"; > > route(RELAY); > > exit; > > } > > The scenario is working but on log there are some error... > t_should_relay_response > > I would like that when a branch reach the failure route, all other branches > will be cancelled..and call is routed to VM. > Can anyone help me? > > When the failure_route is executed, no branch is active anymore. To be able to assist further, you have to send here the errors you get in the log, to understand what is wrong. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

2 1

www.kamailio.org unreachable via ipv6
by Daniel Tryba 03 Aug '15

03 Aug '15

(www.)kamailio.org is unreachable at this time (the whole morning) on 2a00:d60:0:400::2. ipv4 is responsing. Never noticed whether connectivity via ipv6 was working in the past or this is new and misconfigured.

2 1

Announcement: Kamailio is now systemd-rtc-server
by Alex Balashov 03 Aug '15

03 Aug '15

For immediate release: ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software vendor specialising in Kamailio-based service delivery solutions for the VoIP ITSP market, is pleased to announce that it, in collaboration with Red Hat Software and Ringfree Communications, has finalised the absorption of the Kamailio SIP Server into the 'systemd' system management platform for Linux. The new component shall be called 'systemd-rtc-server', or 'Systemd Real-Time Communication Server'. Alex Balashov, principal of Evariste and leader of the tri-vendor collaboration effort, will officially announce the handover of the reigns of the Kamailio project to the personal leadership of Lennart Poettering at the upcoming Systemd Real Time Communications World conference, to be held in Berlin on 27-29 May of this year. John Knight, Director of GNOME 3 Integration and part-time usability consultant at Ringfree Communications, based in Hendersonville, North Carolina,was quick to summarise the triumphs of the long-standing integration effort. Remarked Knight: "The industry has recognised for years that a SIP proxy is a basic building block in the 'init' subsystem of any Linux host. In this age of multimedia communication with voice and video, it was a travesty that systemd handled time synchronisation, network configuration, login management, logging, and console, but not SIP message routing." Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker, was quick to concur: "SIP calls are much easier to troubleshoot with binary logs. Combined with packet captures of TLS-encrypted WebRTC calls, systemd-journald is the ultimate call setup troubleshooting methodology of the responsive, kinetic enterprise." To support the integration of Kamailio into the ecosystem of every major Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio' modules for the project. Balashov stated, "We fully expect to use the D-Bus API to achieve gnome-session integration with systemd-rtc-server-usrloc, but we aren't going to leave Windows users behind; KamailioSvcHost.exe will support Domain Controller policies for G.722 in Active Directory forests." Despite an aggressive delivery timeline by the tri-vendor consortium behind systemd-rtc-server, industry commentators have widely lambasted the fact that it took so long for Kamailio to become integrated into systemd. Fred Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida, recently wrote in a widely-publicised blog post: "sr-dev have been keeping their heads in the sand for too long. For years now, it has been completely obvious and self-evident to anyone with half a brain that all kinds of VoIP software should be included in systemd. It's a basic building block of the whole OS, having absorbed functionality previously provided by all kinds of packages like util-linux and wireless-tools." John Knight of Ringfree accepted the criticism readily, but advocated a forward-thinking orientation focused on breaking with the uncertainty of the past: "In the absence of a SIP component for routing calls to the PSTN, some people thought, 'systemd has no clear direction apart from the whims of its developers, and is a perpetually moving goal post.' Well, a SIP server should put an end to that whole discussion; that's exactly what was missing, and now that we have systemd-rtc-server, we've eliminated all doubts about the coherence, conceptual integrity and finality of systemd." -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

5 8

Check if user is registered before reply is routed
by Dmytro Bogovych 01 Aug '15

01 Aug '15

Greetings. Sorry for previous false letter. I want to check if user is registered before reply is routed to its target user agent. Imho it should be like this: onreply_route[MANAGE_REPLY] { if ($rm=="INVITE") { if (registered("location", $ru)) { # Do job - pause for 15 seconds and give chance to reregister } } } But kamailio complains on registered function 0(6274) ERROR: <core> [cfg.y:3295]: yyparse(): misused command registered 0(6274) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 1121, column 35: Command cannot be used in the block Documentation confirms - registered() function can be used in REQUEST_ROUTE and FAILURE_ROUTE only. Is it possible to check if user is registered when reply is processed in scripts? Thank you!

4 3

Interpretation of Contact header
by Joachim Büchse 30 Jul '15

30 Jul '15

Good day, I’m experiencing some problems with our VoiP providers handling of REGISTER requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 with SIP-Alg enabled. This setup kind of works with UDP but our provider wants us to use TCP. With TCP enforced incoming calls don’t work. I’ve done some wire tracing and to me it seems that the providers configuration is to blame, but then - there are many RFCs out there and many NAT and UAC bug workarounds. Anyway, I wanted to get the opinion of “the" experts about how the requests send to the UAS SHOULD be correctly interpreted. The REGISTER requests/responses look like this (outside of the firewall): Protocol TCP! client port 19091 <-> server port 5060 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Authorization: Digest username="90780408050", realm="pbx.peoplefone.ch", uri="sip:pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy", response="764f371a08d258157a249f8d1b852514" Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.6bda Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp>;q=0;expires=180;received="sip:212.126.160.92:19091;transport=TCP" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 The ip:port the firewall is sending those requests from is ip 212.126.160.92 port 19091. So this does NOT match the port from the Contact header. For TCP this seems rather logical to me, as one cant be listening on a TCP port and use it for sending at the same time. The UAC closes this “register connection” with TCP FIN after the register, and so does the firewall. However unfortunately subsequent requests from the provider (ie UAS) come in on port 19091 (not port 6717 from the Contact header) and the firewall simply drops them. Observations: - the server does NOT include received=212.126.160.92 in the Via of the reponse. According to RFC3581 this is mandatory when rport is present in the request, so this is probably an error in the server. - the server does include received="sip:212.126.160.92:19091;transport=TCP” in the Contact of the response. I didnt see this in any RFC (which means nothing;-) but it could be an error. - after the client received the 200 OK it closes the TCP connection. - the server tries several times to re-contact the client (incoming TCP SYN). However not on port 6717 (defined in the Contact header) but on port 19091 (where the REGISTER came from). RFC3581 defines special behaviour when “rport” is defined in the request (i.e. response should go to the same port the request came from) - however it’s not so clear if this should apply to subsequent (INVITE/OPTIONS) requests from the server to the client. Those are strictly spoken not replies (or are they?). RFC5626 defines that a “proxy” should keep track of the flows over which it received a registration and send requests over the same flow. It is not clear if RFC5626 should be applied. The RFC5626 defines that a UAC includes an “ob” parameter in the Contact field if it whishes further requests over the same flow. Also the RFC mandates a client to add a "reg-id=x" in the Contact field. Both are not the case here, so in short I think RFC5626 should NOT be applied. In which case conecting to the originating port (instead of the Contact port) would be a server error. So in short and if I interpret the RFCs correctly, the client is reachable and should be contacted on Transport: TCP IP: 212.126.160.92 Port: 6717 If anyone who lives and breathes SIP could enlighten me if the UAS is right to call back on 19091 instead of 6717 I would really appreciate it;-)) Best regards, Joachim

3 4

Issue with ACK behind NAT
by Alberto Sagredo 30 Jul '15

30 Jul '15

I have Kamailio in a local network with RTPPROXY and NAT MANAGE I have seen an issue with ACK as show when Asterisk sends ACK to Kamailio, it sends to advertised Address instead to Kamailio IP address. What could i add to solve this? route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails dlg_manage(); } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } BR

2 14

Re: [SR-Users] Memory Issue while accessing $rb (prakash)
by Surendra.P 30 Jul '15

30 Jul '15

Hi Prakash, Specify the more private memory with -M option, while running kamailio Like -M 64 :-) Regards @surendra

1 0

Planning to release v4.2.6
by Daniel-Constantin Mierla 30 Jul '15

30 Jul '15

Hello, I am planning to release v4.2.6 later this week, most likely on Thursday, July 30. If there is anything not reported that should be considered for this minor release of the branch 4.2, write to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

Carrierroute and dispatcher
by Nelson Migliaro 29 Jul '15

29 Jul '15

Hello everybody, I am currently using Kamailio in order to separate traffic based on dialed number. Some traffic goes to several Asterisk in a load balance strategy and other traffic goes to other SIP devices. I use carrierroute module in order to route traffic based on destination. I use scan_prefix to classify traffic and rewrite_host to route the traffic. A destination number usually go to one of 4 asterisk server. I do this in order to load balance the traffic. In order to distribute traffic I use prob (0.25 for each destination). Using this configuration, I have to include on carrieroute table a row for each destination node. Since the amount of destination is huge, I have to repeat configuration in tables. It also complicates maintenance. I was thinking of using a combination of carrierroute and dispatcher module. Carrierroute for classify traffic, strip numbers, and rewrite destination host and dispatcher to distribute the traffic over asterisk. Using this configuration I have to indicate asterisk nodes once in dispatcher table once. Also in carrierroute table I have to indicate each prefix once. I was thinking in something like this. if(!cr_route("carrier", "domain", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); drop; } if(!ds_select_dst("1", "4")) { sl_send_reply("403", "Not allowed"); drop; } t_relay; Does it make sense, or am I complicating everything and there is a magic way to achieve this. Thank you.

2 1

Re: [SR-Users] Memory Issue while accessing $rb
by Daniel-Constantin Mierla 29 Jul '15

29 Jul '15

Hello, you have to increase the private memory -- see -M command line parameter -- as well as pv_buffer_size core parameter -- see core cookbook in the wiki. Cheers, Daniel On 29/07/15 13:37, prakash wrote: > > Hello, > > > > I am getting this Error: > > > > *ERROR: <core> [rvalue.c:1706]: rval_str_add2(): rv eval out of memory* > > *ERROR: <core> [lvalue.c:389]: lval_assign(): rval expression > evaluation failed (881,17-881,29)* > > > > When I tried access *$rb* when sending large amount of data(*nearly > 4MB*) as a XML body. > > How can I sort out this issue? Is there anything to be configured > properly? > > > > NOTE: > > Version that I’m using kamailio 4.1.5 > > > > Thanks and Regards, > > Prakash P > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

Kamailio no audio
by goReeves Admin 29 Jul '15

29 Jul '15

Hi - I just installed Kamailio 4.3.0 on Ubuntu 14.04.2. I can connect using jitsi on windows and csipcimple on the android and do text messaging. However, when I try to do audio callis it seems that the mic is working but the audio out is not functioning. I used the tutorial to do the install so everything in the config files is pretty vanilla except for the database name, use and password. I also installed siremis and added my sip.xxxxx.com as a gateway. On the Ubuntu image I use VPN to a dedicated IP address on which all ports are open. I run web and cloud servers on this machine as well and they all work just fine. I am pretty new to this SIP stuff so it is probably my issue. Here is the network configuration: root@xxxxxxxx-VM:/var/www/siremis-4.2.0# ifconfig eth0 Link encap:Ethernet HWaddr xx:xx:xx:xx:xx:xx inet addr:192.168.x.x Bcast:192.168.x.255 Mask:255.255.255.0 inet6 addr: fe80::20c:29ff:fe84:447a/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1579455 errors:0 dropped:0 overruns:0 frame:0 TX packets:1444329 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:346579445 (346.5 MB) TX bytes:625010061 (625.0 MB) lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:65536 Metric:1 RX packets:98232 errors:0 dropped:0 overruns:0 frame:0 TX packets:98232 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:262176392 (262.1 MB) TX bytes:262176392 (262.1 MB) ppp0 Link encap:Point-to-Point Protocol inet addr:xxx.xxx.xxx.xxx P-t-P:xxx.xxx.xxx.xxx Mask:255.255.255.255 UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1400 Metric:1 RX packets:1105547 errors:0 dropped:0 overruns:0 frame:0 TX packets:1228003 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:3 RX bytes:184558026 (184.5 MB) TX bytes:235399612 (235.3 MB) -- Web Administrator <mail:webadmin@goreeves.net> goreeves <https://www.goreeves.net>

2 2

Memory Issue while accessing $rb
by prakash 29 Jul '15

29 Jul '15

Hello, I am getting this Error: ERROR: <core> [rvalue.c:1706]: rval_str_add2(): rv eval out of memory ERROR: <core> [lvalue.c:389]: lval_assign(): rval expression evaluation failed (881,17-881,29) When I tried access $rb when sending large amount of data(nearly 4MB) as a XML body. How can I sort out this issue? Is there anything to be configured properly? NOTE: Version that I'm using kamailio 4.1.5 Thanks and Regards, Prakash P

1 0

double rr and $fs
by Asgaroth 29 Jul '15

29 Jul '15

Hi All, I have kamailio setup and listening on logical interfaces (for failover purposes) and therefore need to force the sending socket on initial messages. We are testing sips+tls at the moment and what we are trying to achive is that the TLS connection occurs from the UAC to the edge proxy (also responsible for the domain) and then force UDP for internal communications to services. What I am comming up against is that when I force the sending socket on the internal interface to UDP, the record route header still shows up as a "sips" request. Then, at the presense server, when generating the NOTIFY it attempts to send it to the proxy using TLS. My understanding was that the record route would set the uri something like "sip:internal_ip" when forcing the sending socket to be "udp:internal_ip". Is my understanding correct, or am I doing something wrong here? An example subscribe message follows, the top most record-route header, from my understanding, should read sip:internal_ip, not sips:internal_ip. Is this the expected behaviour of double rr with these types of messages? How could I ensure that the top most record-route would always be a sip uri (never sips). SUBSCRIBE sips:subscriber@domain.com:5061 SIP/2.0. Record-Route: <sips:proxy_internal_ip;r2=on;lr;ftag=594055226>. Record-Route: <sips:proxy_external_ip:5061;transport=tls;r2=on;lr;ftag=594055226>. Max-Breadth: 60. Via: SIP/2.0/UDP proxy_internal_ip;branch=z9hG4bK6c9.d7dc4801e872ce9fb5730f9e09f1889e.0;i=1. Via: SIP/2.0/TLS 172.16.0.121:5060;rport=33273;received=78.143.152.30;branch=z9hG4bK708ea7ba. Max-Forwards: 69. Call-ID: 17212ff4-4421321c8feafd63bf800080f0808080@KX-HDV230X. From: <sips:subscriber@domain.com>;tag=594055226. To: <sips:subscriber@domain.com>. CSeq: 2 SUBSCRIBE. Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,MESSAGE,NOTIFY,REFER. Accept: application/dialog-info+xml,application/rlmi+xml,multipart/related. Contact: <sips:subscriber@172.16.0.121:5060;alias=78.143.152.30~33273~3>. Expires: 3600. Event: dialog. Any thoughts, tips, tricks would be greatly appreciated. Cheers

2 2

Kamailio - Missing INVITEs
by Ali Taher 28 Jul '15

28 Jul '15

Hello, I connected Kamailio to a postgresql database (through sqlops module) and fetching result from a table containing around 300K records , where time processing of the single query is taking around 12 ms. I noticed that on huge load (around 200 - 300 calls per second) Kamailio is not processing all INVITE packets. For instance , I initialized a counter in Kamailio that increments on every INVITE , and on the other hand I counted the number of INVITEs found in Wireshark. I got a big difference (from 30K INVITES in Kamailio , I got 15K in Kamailio), though no errors were raised in kamailio log. How can I proceed in order to detect the problem? :( Any help would be appreciated. Regards, Ali

2 1

RR module - Fail detecting strict routing
by Antonio Reale 28 Jul '15

28 Jul '15

Hi all, I have the following scenario: U1 (caller) ---> P1 (192.168.0.245, kamailio 4.3, loose-router) ----> P2 ----> (192.168.0.101, strict router) ----> .... ----> U2 (called) When U2 answers the call, at P1 arrives the 200 OK with: Record-Route: <sip:0000000011@192.168.0.101;transport=udp;pgw-call=call-28d2e> Record-Route: <sip:192.168.0.245;lr=on> The problem is that the ACK from U1 is forwarded from kamailio to P2, as if P2 is a loose router. P2 drops the ACK message. From kamailio logs I see: Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:88]: is_preloaded(): is_preloaded: No Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:783]: after_loose(): Topmost route URI: 'sip:192.168.0.245;lr=on' is me Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:869]: after_loose(): URI to be processed: 'sip:0000000011@192.168.0.101;transport=udp;pgw-call=call-289ce' Jul 14 10:07:44 P1 /usr/sbin/kamailio[15908]: DEBUG: rr [loose.c:878]: after_loose(): Next URI is a loose router Why after_loose function considers the next URI a loose router? The Route HF in the ACK with the URI of the next hop does not contain ;lr . It seems that the function is_strict fails detecting the strict router. Thanks. Regards. Antonio

2 7

CNXCC lock
by Kelvin Chua 28 Jul '15

28 Jul '15

after a call exhausted credits, the next call will fail with a timeout. found out it gets stuck in cnxcc_mod.c line 1137 static call_t *__alloc_new_call_by_money(credit_data_t *credit_data, struct sip_msg *msg, double credit, double cost_per_second, int initial_pulse, int final_pulse) { call_t *call = NULL; lock_get(&credit_data->lock); ... the next lines will never be executed Kelvin Chua

2 2

SIPCapture and Message Queues
by Grant Bagdasarian 27 Jul '15

27 Jul '15

Hello, Would it be possible to send the messages captured by the sipcapture module to a message queue instead of inserted directly into a database? Be it DMQ or SQL Server Service Broker Queues, etc. Or maybe another approach, executing a stored procedure instead of an insert statement? I couldn't find a way to accomplish this, but maybe someone has a brilliant idea how to accomplish any of the above. Regards, Grant

2 2

Problem Storing USRLOC Additional Attributes
by Luca Mularoni 27 Jul '15

27 Jul '15

Hi All, I'm using kamailio 4.2.5 (compiled from source and aligned up to commit [00a36ae07c587992d6486277ce2e9ae6c009685f]). In order to enrich users data collected during registering phase, I would like to store them in USRLOC Additional Attributes through the use of 'xavp_contact' structure, but I'm having problems storing them. Below relevant chunks of config: ... # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 0) modparam("usrloc", "xavp_contact", "ulvals") ... route[REGISTRAR] { ... # Just an example $xavp(ulvals=>A) = "A"; $xavp(ulvals[0]=>B) = "B"; $xavp(ulvals[0]=>C) = "C"; $xavp(ulvals[0]=>D) = "D"; ... if (!save("location")) sl_reply_error(); exit; } route[LOCATION] { ... if(!lookup("location") { # Handle Failure ... } # Print USRLOC Additional Attributes # Just an Example $var(i) = 0; $var(N) = $branch(count); while($var(i)<=$var(N)) { xlog("LOC - [$var(i)] - A [$xavp(ulvals[$var(i)]=>A)]\n"); xlog("LOC - [$var(i)] - B [$xavp(ulvals[$var(i)]=>B)]\n"); xlog("LOC - [$var(i)] - C [$xavp(ulvals[$var(i)]=>C)]\n"); xlog("LOC - [$var(i)] - D [$xavp(ulvals[$var(i)]=>D)]\n"); $var(i) = $var(i) + 1; } ... } And this is the relevant chunk of syslog when a local subscriber is found in 'location' table: ... LOC - [0] - A [A] LOC - [0] - B [<null>] LOC - [0] - C [<null>] LOC - [0] - D [D] ... Summing up, it seems that when xavp structure is stored in memory, just first and last elementsare inserted. In real config I'm using more than 4 attributes and behaviour is the same: just first and last are stored. I've tried also with modparam("usrloc", "db_mode", 0) but the problem remains. Where I'm wrong in using this structure? Or it could be a bug? Thank You in advance for your help Luca Mularoni

2 11

Check if user is registered after receiving reply
by Dmytro Bogovych 27 Jul '15

27 Jul '15

Greetings. I want to check if user is registered before kamailio will try to route reply to its user agent. IMHO it should be: onreply_route[MANAGE_REPLY] {

2 1

Re: [SR-Users] FROM number based on destination number
by Keith 27 Jul '15

27 Jul '15

Hi, I have a requirement to change the FROM number to a different number depending on the destination number. For example if I dial 01789 from 0800 I want to change the 0800 from number to a 01789 number. Ideally I want to be able to do a db lookup on the destination number to then set the from number. Thanks Keith

1 0

Changing acc_cdrs time and duration columns from varchar to int/float
by mayamatakeshi 27 Jul '15

27 Jul '15

Hello, I am planning to enable acc cdr in my kamailio 4.1 server. I am testing in my lab and it is working fine. However, when I check the table definition I see this: mysql> describe acc_cdrs; +------------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +------------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | start_time | varchar(32) | NO | MUL | | | | end_time | varchar(32) | NO | | | | | duration | varchar(32) | NO | | | | +------------+------------------+------+-----+---------+----------------+ 4 rows in set (0.00 sec) The *_time and duration columns are defined as varchar. Because of this, i need to perform a cast when doing queries: select * from acc_cdrs where cast(start_time as unsigned) >= 1437120622; which would cause extra processing in the db server. So I experimented changing these columns to "int unsigned" and to "float unsigned" and it worked for both without problems (records properly added to acc_cdrs and no errors in kamailio.log): alter table acc_cdrs modify start_time int unsigned; alter table acc_cdrs modify end_time int unsigned; alter table acc_cdrs modify duration int unsigned; alter table acc_cdrs modify start_time float unsigned; alter table acc_cdrs modify end_time float unsigned; alter table acc_cdrs modify duration float unsigned; So I am thinking in doing this in my production server but I would like to know if there is any potential risk of doing this (as I am not sure why it was decided for these fields to be varchar instead of a numeric type). Regards, Takeshi

2 3

New SIP INVITE from UAS (new dialogue)
by Joao Alves 27 Jul '15

27 Jul '15

Hi, What would be the best way to generate and send an INVITE message generated by the Kamailio server. This would be for a new dialog session, UAS initiated (e.g. to make a click 2 call like service). I've tried the $uac_req(method)="INVITE"; uac_req_send(); but didn't work. I gather something would be needed first to open the dialogue before doing it but couldn't find it on the dialogue module. Thanks, Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

4 10

Issues sending SIP UPDATE inside an existing dialogue.
by Joao Alves 24 Jul '15

24 Jul '15

Hi, I need to send an SIP UPDATE inside an existing dialogue (in order to update the SDP offer). I've been using the following uac functions from the command file. $uac_req(method)="UPDATE"; $uac_req(ruri)=$shv(userAcontact); $uac_req(turi)=$tu; $uac_req(furi)=$fu; $uac_req(callid)=$ci; $uac_req(hdrs)="Content-Type: application/sdp\r\n"; $uac_req(body)=$var(sdp); # uac_restore_from(); uac_req_send(); As a result, the SIP UPDATE is being sent but since the original From Tag is not being used (i.e. and a new tag is generated), the SIP client (sipML5) does not recognize it as part of the existing dialogue. Attach is a pcap of the test made. Subsequently, I've also tried to recover the from using uac_restore_from() function but got the following an error message. DEBUG: uac [replace.c:511]: restore_uri(): getting 'vsf' Route param DEBUG: uac [replace.c:515]: restore_uri(): route param 'vsf' not found Thanks, Joao Alves Solution Architect, Unified Communications +351 214094660 (desk) +351 912783702 (mobile) AMDOCS | EMBRACE CHALLENGE EXPERIENCE SUCCESS This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

1 0

Installation issues
by ngn zone 24 Jul '15

24 Jul '15

>Hello all, I am new to Kamailio. I Tried installing Kamailio 4.0 and I had the following error. I will appreciate any help >ERROR: could not load the script in /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL'

2 1

dialog keepalive timeout - which parameter rules it?
by Andrey Utkin 24 Jul '15

24 Jul '15

What happens if dialog parties don't respond to keepalive requests sent by dialog module? It looks it is not coded in modules/dialog, I guess it can be set in t_uac_prepare() from modules/tm/uac.c. Please explain what timeout or lifetime parameters rule - at which delay the unanswered keepalive request sent by Kamailio within dialog triggers dialog termination. Thanks a lot. -- Andrey Utkin

2 2

dmq_usrloc replication and max_contacts
by Asgaroth 24 Jul '15

24 Jul '15

Hi All, I have a situation where I have 2 devices registering with the same AOR, I have the registrar module's max_contacts parameter set to 1 and I use the 0x04 flag on the save function. When I use dmq_usrloc for replication, I can see 2 contacts registered for the AOR on the "client" nodes whereas the node where the actual registration took place only has 1 contact for the AOR. Is this expected behaviour of the dmq_usrloc module in terms of registration or have a come accross another bug? Thanks

2 5

"kamctl ul show"/xmlrpc ul.lookup AOR not found for valid registration
by Asgaroth 24 Jul '15

24 Jul '15

Hi All, I have a situation where a device has successfully registered and its location saved/stored in memory. I can see all the stored information when I issue a "kamctl ul show" However, if I perform a kamctl ul show subscriber@domain, where the 'subscriber@domain' is copied/pasted from the output of the "kamctl ul show" command, it results in an AOR not found error. However, with other AOR's on the same registrar, the "kamctl ul show subscriber@domain" works. I checked the usernames for the failed commands, and I dont see anything obvious, they are a combination of lower/upper-case leters and digits. String length is around 24 characters for the username (the username table for the subscriber is 64 characters). I'm not sure what elese to check to see why the AOR lookup for these particular clients are failing when they are indeed registered. All other functions around the lookup seem to work, for example, a location lookup for these AOR's are working. I wonder if I have come accross an issue with the kamctl function and/or the equivilient xmlrpc ul.lookup method. Any thoughts on this would be greatly appreciated. Thanks

1 3

sending INVTE with Digest values
by Al S 24 Jul '15

24 Jul '15

I am registering my client to kamailo successfully. client --> Register --> kamailioclient <-- 401 with nonce value <-- kamailioclient --> Register with nonce and md5 response values --> kamailioclient <-- 200ok <-- kamailio However, when the client sends an invite with the same digest values, kamailio sends a 407 request for another challenge: client --> Invite with the same nonce and md5 response values --> kamailioclient <-- 407 <-- kamailio I am thinking the same digest values from register could be used for making calls. Thanks,Al

2 1

Dispatcher module number of calls
by Alberto Sagredo 23 Jul '15

23 Jul '15

Hi Hi have read documentation but it seems dispatcher does not keep how many calls has been dispatched or currently are in any of dispatcher destinations I have take a look to code on: http://lists.sip-router.org/pipermail/sr-users/2012-July/073919.html But it seems to use calls limit per user, I would need to establish a limit per dispatcher destination i order to do not send more calls. Any idea where to look for? I have take a look to DMQ module, that maybe i could to use to spread information of how many calls to every destination but do not know how to extract calls from kamailio that have been dispatched to this destination using disptcher module. Best Regards Alberto

3 5

Kazoo like load balancing
by Jurijs Ivolga 23 Jul '15

23 Jul '15

Hi, First of all I would like to mention, that I'm very new to Kamailio. :) My target is to create simple load-balancing similar what have Kazoo, basically 2 servers will share same domain name. I installed 2 Kamailio 4.3 servers and configured this 2 servers with one domain address. I used almost default config with: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_MULTIDOMAIN #!define WITH_NAT #!define WITH_NATSIPPING Calls are going through if I register both clients to one server. But all my calls are failing, when clients are registered to different servers with error "404 Not Found". Maybe somebody can give me a hint how to make this work. Thank you! With kind regards, Jurijs

3 19

Re: [SR-Users] Kamailio mod_destroy() ERROR
by Ali Taher 23 Jul '15

23 Jul '15

Any help regarding the below would be appreciated. BR, Ali From: Ali Taher [mailto:ataher@vanrise.com] Sent: Wednesday, July 22, 2015 1:39 PM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Cc: 'Ali Taher~Vanrise Technical Support' <ataher(a)vanrise.com> Subject: Kamailio mod_destroy() ERROR Hello, I'm getting below error when starting Kamailio. ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Any advice ? Regards, Ali

2 1

Port 5061 Issue
by Ankhit Vivekananda 23 Jul '15

23 Jul '15

Hello, I have configured kamailio 4.0.4 long back. Now, I wanted to enable the usage of port 5061 with TLS. By doing so, the client(video camera) can place outgoing calls fine. But while receiving an incoming call, the message does not get sent out to the other client. Hence, the call fails. I can place and receive calls on port 5060 successfully without any issues. Can anyone please tell me why port 5061 is giving me a problem. Also, as a test, I set up kamailio on another machine(version 4.1) and 5061 works fine in all directions. It's only on the other machine that I have problems with. Appreciate any help! Thank you, Ankhit

2 1

htable replication between several kamailio's through dmq
by José Seabra 22 Jul '15

22 Jul '15

Hello there, I'm using the dmq and htable modules to limit the number of concurrent calls in several kamailio's servers. Every call that one kamailio server receives, kamailio increments a htable variable that is replicated to other kamailios through dmq, but if one of these kamailios fails or is restarted it looses the data from htable variable used to count the number of calls and it is not recovered again when kamailio starts again. Is there any way to recover the data from htable variable when kamailio starts, from another kamailio instances through dmq module? Best Regards -- Cumprimentos José Seabra

2 1

RTP Proxy to WAV Script conversion
by Alberto Sagredo 22 Jul '15

22 Jul '15

Using RTPBreak and sox im able to convert rtpproxy rtp files to wav. Later you could convert to mp3 if liked Just to share with list https://github.com/albersag/rtpproxy-utils Any comment is well appreciated Best Regards

2 1

dmq_is_from_node() issue
by Asgaroth 22 Jul '15

22 Jul '15

Hi All, I have an issue with the dmq_is_from_node() function, whereby it does not detect a node is on the bus if it has been removed from its "table" due to previously failed, i presume, ping requests. I have the following snippet in my main routing block: if(is_method("KDMQ")) { if (dmq_is_from_node()) { dmq_handle_message(); exit; } else { xlog("L_INFO", "KDMQ message recieved from node not on our bus!! ($si:$sp) Dropping."); exit; } } I have the following dmq module parameters enabled for dmq: modparam("dmq", "server_address", DMQ_ADDRESS) modparam("dmq", "notification_address", "sip:lb.domain.com:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) DMQ_ADDRESS is set to each of the addresses in the lb.domain.com for each local kamailio instance lb.domain.com resolves to the following addresses: 10.6.0.109 10.6.0.110 10.6.0.111 If I start up the 1st instance of kamailio and do not start the other 2 within the default ping timout then I get the expected log message of dmq removing the nodes from its table, for example: kamailio[26657]: ERROR: dmq [notification_peer.c:588]: notification_resp_callback_f(): deleting server sip:10.6.0.111:5060 because of failed request kamailio[26657]: ERROR: dmq [notification_peer.c:588]: notification_resp_callback_f(): deleting server sip:10.6.0.110:5060 because of failed request However, if I then start one, or both, of the other nodes, then the KDMQ requests are dropped due to my snippet above, I presume, because dmq_is_from_node() checks its "active" table. See messages below as an example of the above snippet in action: kamailio[26656]: INFO: <script>: KDMQ message recieved from node not on our bus!! (10.6.0.110:5060) Dropping. kamailio[26656]: INFO: <script>: KDMQ message recieved from node not on our bus!! (10.6.0.111:5060) Dropping. Is this expected behaviour from the dmq_is_from_node() function? I would have expected it to always check the addresses resolved in the notification address, or am I using this function in a manner not intended. Any suggestions/tips would be greatly appreciated. Thanks

3 5

RFC: about the self signed tls certificates
by Daniel-Constantin Mierla 22 Jul '15

22 Jul '15

Hello, (cross-posting as it impacts users as well) Currently we generate and install the TLS self-signed certificates when tls module is installed, including them in the debian packages (and I guess in rpms). Debian has a policy of reproducible builds, meaning that from same source tree snapshot the same binary packages should result. Even more important considering the impact on security, it would be better that the certificates are generated locally on the installation server, to be distinct. Right now, people relying on default installation config/certificates (and I guess there are many, at least in testing phase), are exposed to eavesdropping, because the private key is available in public packages. My proposal is to move generation of self signed certificates to kamctl. There can be a kamctl.tls file to be deployed by the tls package (same is done by kamctl.mysql, being part of mysql package), which should add a new group of commands, among them something like: kamctl tls generate-certificate The drawback is that before enabling tls and starting kamailio, one has to run the above command. We can document that in tls module readme and in kamailio.cfg in the comments related to WITH_TLS define. Anyone with comments, pros/cons? Other suggestions on how to address the reproducible builds as well as solve the security issue for the default installation? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

5 4

Kamailio mod_destroy() ERROR
by Ali Taher 22 Jul '15

22 Jul '15

Hello, I'm getting below error when starting Kamailio. ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Any advice ? Regards, Ali

1 0

replace From header username for AUTH
by Al S 22 Jul '15

22 Jul '15

Hi, I was wondering how can I replace From header username to perform the following authentication check with the new value: route[AUTH]......if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "1"); exit;}... Thanks,Al

2 1

kamailio 4.3v partition column name is reserved word in mysq ndbcluster engine
by José Seabra 21 Jul '15

21 Jul '15

Hello Daniel, After update my kamailio to version 4.3 and change the table location as required in migration process, kamailio start giving some errors: [root@ccp1 kamailio]# 35(7637) ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Unknown column 'partition' in 'where clause' 35(7637) ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query that's is because the column name "partition" is reserved word ( http://dev.mysql.com/doc/refman/5.6/en/keywords.html) in mysql ndbcluster engine, so what i did to fix this issue was add a new modparam in modules/usrloc/ul_mod.c file: {"partition_col", PARAM_STR, &partition_col}, with this i can configure what name i want for this column in kamailio module configuration and configure a diferent column name in table location(ex: modparam("usrloc", "partition_col", "partitionusrloc") ), Can you please check out if what i did is the correct way to fix it, in order to avoid damage something else. Best regards -- Cumprimentos José Seabra

2 6

Compiling only one module
by Loic Chabert 21 Jul '15

21 Jul '15

Hello, I'm writing a new module for influxdb stats. I've wrote my code and now i need to compile it. How can i compile it ? I don't want to compile all modules, only this one. Thanks in advance, Regards.

2 3

usage of t_flush_flags() in event_route
by Andrew Pogrebennyk 20 Jul '15

20 Jul '15

Hi Daniel and others, I'm having a problem with acc module if I'm using the event_route/ branch-failure: say, the call comes from the app server and goes to the registered user. We arm the the failure route and per-branch failure route for the 302 redirect from the UA. We explicitly reset the accounting flags because we don't want to account the calls from the app server. The transaction is created implicitly by the t_relay(). Now if the UA responds with the 302 response and we are going to process that, we want to create an acc record for the new target from 302 message because this call may incur additional costs. I'm setting the accounting flags and calling t_flush_flags() but that doesn't work (no accounting record). Any idea if I'm doing something wrong or maybe there's a bug when changing the flags and then calling t_flush_flags from the event_route? Here are the module parameters: modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "db_url", PAIR_URL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)") modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "cdr_log_enable", 0) FTR, we tried 4.1.6 and 4.1.8. And here is the event route (with flags defined like this: flags FLAG_ACC_DB:1, FLAG_ACC_MISSED:2, FLAG_ACC_FAILED:3, ...): event_route[tm:branch-failure:redirect] { route(ROUTE_STOP_RTPPROXY_BRANCH); if($T_rpl($rs) == 301 || $T_rpl($rs) == 302) { # initialise variables when entering failure route route(ROUTE_INITVARS); # these need to be avps because we need it in reply/failure-route $(avp(s:from_faxserver)[*]) = 0; $(avp(s:to_faxserver)[*]) = 0; $(avp(s:cf_from_pstn)[*]) = 0; $(avp(s:from_pstn)[*]) = 0; $(avp(s:proxylu_from_pstn)[*]) = 0; $(avp(s:lcr_flags)[*]) = 0; $(avp(s:em_call)[*]) = 0; $(avp(s:from_pbx)[*]) = 0; $(avp(s:p_to_device)[*]) = 0; $(avp(s:p_to_group)[*]) = 0; $(avp(s:is_primary)[*]) = 0; # now let's process a 30x $(avp(s:acc_state)[*]) = "cfb"; $(avp(s:orig_acc_caller_user)[*]) = $avp(s:acc_caller_user); $(avp(s:orig_acc_caller_domain)[*]) = $avp(s:acc_caller_domain); $(avp(s:acc_caller_user)[*]) = $avp(s:acc_callee_user); $(avp(s:acc_caller_domain)[*]) = $avp(s:acc_callee_domain); $(avp(s:caller_uuid)[*]) = $avp(s:callee_uuid); $(avp(s:callee_uuid)[*]) = $null; # the $var(no_acc) is 0 at this point but the flags may have been reset # if this is a call from PBX user - we do want accounting for the 302 redirect if(isflagset(FLAG_ACC_DB)) { xlog("L_NOTICE", "++++++ ACC flag is set - [% logreq -%]\n"); } else { xlog("L_NOTICE", "------ ACC flag is NOT set - [% logreq -%]\n"); } setflag(FLAG_ACC_FAILED); setflag(FLAG_ACC_DB); t_flush_flags(); # get last URI from destination-set and set it as R-URI $var(contact) = $T_rpl($ct); $var(contact) = $(var(contact){nameaddr.uri}); if($var(contact) == 0 || $var(contact) == $null) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - [% logreq -%]\n"); acc_db_request("480", "acc"); $var(announce_handle) = "callee_tmp_unavailable"; $var(announce_set) = $xavp(callee_real_prefs[0]=>sound_set); $(avp(s:announce_code)[*]) = 480; $(avp(s:announce_reason)[*]) = "Temporarily Unavailable"; route(ROUTE_EARLY_REJECT); } $ru = $var(contact); xlog("L_NOTICE", "Redirect from UAC intercepted - [% logreq -%]\n"); $(avp(s:forwarder_cli_userprov)[*]) = $T_rpl($tU); $(avp(s:forwarder_domain_userprov)[*]) = $T_rpl($td); $var(forward) = 1; $var(redirected_forward) = 1; route(ROUTE_LOAD_CALLER_PREF); route(ROUTE_FIND_CALLEE); } } Thanks. Andrew

2 5

File Transfer
by Abdul Hakeem 20 Jul '15

20 Jul '15

Hello, Is there a support for file transfer between registered SIP clients, If not, how can this be achieved ? Regards, Abdul Hakeem

1 0

Kamailio v4.3.1 Released
by Daniel-Constantin Mierla 20 Jul '15

20 Jul '15

Hello, Kamailio SIP Server v4.3.1 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.3.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.1. For more details about version 4.3.1 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/07/kamailio-v4-3-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Missing BYE at the far end
by Kliment Toshkov, Netfinity JSC 20 Jul '15

20 Jul '15

Dear colleagues, We have an issue with a couple of peers - when the called party hangs up, they (the peer) does not receive the appropriate BYE message and call appears as still connected at peers end. Then it drops after 30 seconds and sends us BYE. Current setup is the following: PEER IP > KAMAILIO 4.2.2 > (Asterisk PBX 1, 2, 3, …, N) We provision only termination services, which is: 1) peer calls us 2) Kamailio chooses destination using dispatcher module 3) forwards the call to it. Attached is captured trace. Clearly BYE is retransmitted a few times, then request for UPDATE is received but it is not possible to answer it because the call is already hung up (as far as I understand). Kamailio routing file is available upon request. Thank you in advance! Климент Тошков Изпълнителен директор 24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/> t: +359 2 4918888 f: +359 2 4815555 <http://www.netfinity.bg/> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.

2 1

Planing to release v4.3.1
by Daniel-Constantin Mierla 20 Jul '15

20 Jul '15

Hello, I am considering to release the first patch version in 4.3 series by beginning of next week (Monday or Tuesday, July 20 or 21). If you are aware of issues not reported yet, then open an item on the tracker to be taken care: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
by Ben Fitzgerald 20 Jul '15

20 Jul '15

Hi, I've been following this integration tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and have a successful registration and I can even make calls through my asterisk box. However what is unusual to me is that every time a phone registers with Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies with 401 Unauthorized. Oddly enough the phone registers and can still make calls. What worries me is that as we scale to 100's of cps, this seemingly erroneous message may slow down Asterisk because it's trying to handle authentication for users which have already been authenticated by Kamailio. If this behavior is expected, then that would be good to know as well. This is the sip debug from ASTERISK (I have replaced IP's with the names of the servers): <--- SIP read from TCP:kamailio:41205 ---> REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0 To: <sip:40081@asteriskIP> From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 CSeq: 10 REGISTER Call-ID: 0005ce130bcee5c4-26538@kamailio Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (4.3.0 (x86_64/linux)) Contact: <sip:40081@kamailio:5060> Expires: 3600 <-------------> --- (11 headers 0 lines) --- Sending to kamailio:5060 (no NAT) Sending to kamailio:5060 (no NAT) <--- Transmitting (no NAT) to kamailio:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= kamailio From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 To: <sip:40081@asteriskIP>;tag=as404bac9a Call-ID: 0005ce130bcee5c4-26538@ kamailio CSeq: 10 REGISTER Server: Asterisk PBX 11.6-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="262b338e" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '0005ce130bcee5c1-26536@ kamailio' Method: REGISTER ========================= sip.conf for kamailio trunk: [kamailio-inbound] type=friend dtmfmode=auto host=kamailioIP allow=all context=sipout insecure=port,invite canreinvite=no ======================== Asterisk version: 11.6-cert2 Kamailio version: 4.3 Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600

3 7

Double Record-Route to subscriber with multiple locations on different SIP transports
by Andrey Utkin 17 Jul '15

17 Jul '15

Hi! We found that we need double Record-Route when we call from one SIP transport to another (let's say, TCP -> TLS), because without |transport=" hint in Record-Route, mobile Linphone app sends messages (INVITE reply, ACK, BYE) via UDP, even if configured for another transport. record_route_advertised_address() is told to add double RR if needed, automatically, so we use it. But the subscriber (the receiver of call) can have multiple registered locations, let's say, one connected via TCP, and another via TLS. So we need to handle each call branch added by lookup(). But adding record_route_advertised_address() ONLY to branch_route attached by t_on_branch() doesn't execute if there's only one location. Having record_route_advertised_address() just in routing script after lookup() seems wrong because it seems unable to edit it for other branches (I don't remember if I have tested this and have proven a failure, or this is just my speculation). So we have added it in both places - right under lookup(), and in branch_route. But in this case we have an error logged: ERROR: rr [rr_mod.c:337]: w_record_route_advertised_address(): Double attempt to record-route Could anyone explain what would be the most correct solution? What is intended behaviour of branch_route? Is it omitted from execution on "primary" branch of call? Is there any parameter which could help? Thanks in advance. -- Andrey Utkin

3 7

Dialogue restore after initial processing
by Joao Alves 17 Jul '15

17 Jul '15

Hi, I'm trying to restore a SIP session dialogue context and found dlg_get(callid, ftag, ttag) function for that. However, I cannot access the To tag. I noticed on the 180 RINGING there was already an self-generated included on this response, but I also could not access its value. When receiving the SIP INVITE request I used the dlg_manage() and the context was initially stored. if(is_known_dlg()) { xlog("Request $rm from $ci is in-dialog\n"); xlog(">>From tag is $ft and To tag is $tt\n"); # Needed to later restore the dialog $shv(ftag)=$ft; $shv(ttag)=$tt; } 15(49100) ERROR: <script>: Request INVITE from 76589ZWQ5NzIyYjMzNDI1YjVjM2E5ZWU5NjhkYzEyNjE3MzQ is in-dialog 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1021 a=25 n=xlog 15(49100) DEBUG: pv [pv_core.c:376]: pv_get_xto_attr(): no Tag parameter 15(49100) ERROR: <script>: >>From tag is 8fd06474 and To tag is <null> Since there was a self-generated one (after the 180 RINGING being sent), I rechecked the $tt and $dlg(to_tag) values and both still returned null values. 15(49100) DEBUG: tm [t_lookup.c:1083]: t_check_msg(): DEBUG: t_check_msg: T already found! 15(49100) DEBUG: <core> [tcp_main.c:2196]: tcpconn_send_put(): send from reader (49100 (15)), reusing fd 15(49100) DEBUG: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... 15(49100) DEBUG: <core> [tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7effdf7da860 n=396 fd=5 15(49100) DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf= SIP/2.0 180 ringing Via: SIP/2.0/TCP 192.168.1.5:58751;branch=z9hG4bK-524287-1---a64e0a42199ee419;rport=61134;received=85.138.200.8 To: <sip:1001@decasa.website>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-f6a3 From: "1000"<sip:1000@decasa.website>;tag=8fd06474 Call-ID: 76589ZWQ5NzIyYjMzNDI1YjVjM2E5ZWU5NjhkYzEyNjE3MzQ CSeq: 1 INVITE Server: kamailio (4.3.0 (x86_64/linux)) Content-Length: 0 ..., shmem=0x7effdf7e1e00: SIP/2.0 180 ringingght(): DEBUG: reply sent out. buf=0x7effec023bf8: SIP/2.0 180 ringing 15(49100) DEBUG: tm [t_reply.c:657]: _reply_light(): DEBUG: _reply_light: finished 15(49100) DEBUG: sl [sl.c:272]: send_reply(): reply in stateful mode (tm) 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1031 a=25 n=xlog 15(49100) DEBUG: dialog [dlg_hash.c:640]: dlg_lookup(): ref dlg 0x7effdf7e1bf0 with 1 -> 2 15(49100) DEBUG: dialog [dlg_hash.c:642]: dlg_lookup(): dialog id=5769 found on entry 128 15(49100) DEBUG: dialog [dlg_hash.c:858]: dlg_unref(): unref dlg 0x7effdf7e1bf0 with 1 -> 1 15(49100) ERROR: <script>: Dialog To Tag: <null> <== xlog("Dialog To Tag: $dlg(to_tag)\n"); 15(49100) exec: *** cfgtrace:request_route=[MATRIX] c=[/etc/kamailio/kamailio.cfg] l=1033 a=25 n=xlog 15(49100) DEBUG: pv [pv_core.c:376]: pv_get_xto_attr(): no Tag parameter 15(49100) ERROR: <script>: >>From2 tag is 8fd06474 and To tag is <null> Many thanks, Joao Alves Solution Architect, Unified Communications This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

2 3

Configuring Kamailio as SIP Proxy
by Djamel Bahamid 16 Jul '15

16 Jul '15

Hi, I have terminals of video conference which are in public IP ( *DMZ*). I installed kamailio (4.2) which works correctly, it is also in IP pulique (*DMZ*). I wish configured the /kamailio.cfg/ file, so that kamailio works in *proxy SIP*. And declared my CODEC of visoconférence in URI, example: 'DR London(a)domaine.fr'. In what part of the file ( *kamailio.cfg* ) do I have to make these modifications ? This to avoid the flooding. How possibly write the rule of corresponding ACL. Thank you for your help. Best Regards,

3 4

interconnect two Kamailio server
by Kai Ohnacker 16 Jul '15

16 Jul '15

Hello, I have a setup with 2 Kamailio server and their own database. Is it possible to connect these 2 Kamailio server, that the client from the one can make a call to the other? If yes, how could this be realized? Many Thanks. Cheers, Karl --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com

2 1

Error forwarding to SCSCF
by Jan Devai 16 Jul '15

16 Jul '15

*Hi Experts,* *I would like to ask you if some of you have already come across this problem.* *We are integrating Kamailio all in one Solution (P/I/S-CSCF having all same IP only communicating between each other on different ports) taken from Carsten Bock example against an external HSS.* *The flow is as follows:* UE HSS P-CSCF I-CSCF S-CSCF --------------------------Sip Register--------------------------> ----------Sip Register---------> <---------------------------------UAR-------------------------------- ---------------------------------UAA--------------------------------> --------------SIP Register-------> *The last step "SIP Register" from I-CSCF to S-CSCF is not happenning for some reason.* *The Received AVP in UAA is "First Time Registration 2001 code".* *Now we do not know whether this UAA answer is OK, or some other URI is badly defined which could explain why I-CSCF has trouble contacting S-CSCF.If someone have some logs, traces which could be used as a reference for this scenario would be greatly appreciated.* *Find theses messages we get in the shown logs below:* *Thank you* *JanThis file /rtc/kamailio_icscf/kamailio.cfg is a function which sends message to SCSCF, see below:* route[REG_UAR_REPLY] { #xlog("$$avp(s:uaa_return_code) = $avp(s:uaa_return_code)\n"); #this is async so to know status we have to check the reply avp switch ($avp(s:uaa_return_code)){ case 1: #success if (I_scscf_select("0")){ t_on_failure("register_failure"); t_on_reply("register_reply"); #now relay to appropriate SCSCF if (!t_relay()) { t_reply("500", "Error forwarding to SCSCF"); } } else {#select failed I_scscf_drop(); t_reply("500", "Server error on SCSCF Select (UAR)"); } break; case -1: #failure xlog("L_ERR", "UAR failure - error response sent from module\n"); break; case -2: #error xlog("L_ERR", "UAR error - sending error response now\n"); t_reply("500", "UAR failed"); break; default: xlog("L_ERR", "Unknown return code from UAR, value is [$avp(s:uaa_return_code)]\n"); t_reply("500", "Unknown response code from UAR"); break; } } *Kamailio is getting an answer: "t_reply("500", "Error forwarding to SCSCF");"* *which means that th function I_scscf_select("0") is fine but the other function called "t_relay", is the one generating the error "Error forwarding to SCSCF".* *I-CSCF is not able to send the SIP message to S-CSCF, see this log below:* Jul 16 14:59:22 cw-aio /usr/sbin/kamailio[31604]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31599]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: ERROR: tm [ut.h:254]: uri2dst2(): ERROR: uri2dst: bad_uri: scscf.epc.mnc004.mcc324.3gppnetwork.org Jul 16 14:59:23 cw-aio /usr/sbin/kamailio[31605]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches

1 0

Purpose of table location_attrs
by mayamatakeshi 16 Jul '15

16 Jul '15

Hello, I need to prepare two kamailio instances accessing the same mysql db. They will write to different location tables so i use function calls like these: Server1: save("k1_location"); lookup("k1_location"); Server2: save("k2_location"); lookup("k2_location"); This is working fine. However, checking db, I can see there is also a table called location_attrs and I am not sure what is its purpose (is this documented?). Anyway, I suspect I should also set a separate copy of it for each kamailio instance, but I am not sure how to do it. Regards, Takeshi

2 2

Error: make modules include_modules ="db_mysql" cfg
by kumar m 16 Jul '15

16 Jul '15

I am a newbie trying out kamailio *kamailio version: *4.3.0 *Operating system:* CentOS 7 Following the instructions provided at http://www.kamailio.org/wiki/install/4.3.x/git While I was trying to include the module "db_mysql", I get the following error [root@localhost kamailio]# make modules include_modules="db_mysql dialplan app_mono" cfg Makefile:56: modules.lst: No such file or directory target architecture <x86_64>, host architecture <x86_64> saving modules list... target architecture <x86_64>, host architecture <x86_64> which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) make --no-print-directory -wC ../.. cfg-defs target architecture <x86_64>, host architecture <x86_64> making config... which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin) CC (gcc) [M db_mysql.so] km_dbase.o km_dbase.c:36:25: fatal error: mysql/mysql.h: No such file or directory #include <mysql/mysql.h> ^ compilation terminated. make[1]: *** [km_dbase.o] Error 1 make: *** [modules] Error 1 Please help me in fixing the error

2 1

Unclear documentation for "priority" field in Dispatcher module
by Sven Neuhaus 16 Jul '15

16 Jul '15

Hello, the documentation doesn't clearly state if a larger value for the priority field means that it gets used first before entries with smaller priority values are used - or vice versa. It currently states: “8” - select destination sorted by priority attribute value (serial forking ordered by priority). and priority: sets the priority in destination list (based on it is done the initial ordering inside the set) This is just ambiguous. It should clearly say wether the list is processed in ascending or descending order for the priority value when using algorithm 8 (serial forking aka failover). Regards, -Sven Neuhaus

2 1

Issue handling SRTP and RTP with rtpproxy and rtpengine
by Alberto Sagredo 15 Jul '15

15 Jul '15

Hi Im using an Asterisk cluster behind Kamailio and some services are running on Asterisk. Traffic always must go to Asterisk so RTPPROXY and RTPNGINE do the work and send traffic to Asterisk and traffic from Asterisk go to RTP Proxy I have been able to get rtpproxy working and later tried RTPengie to be able to manage SRTP I have been able to make SRTP To RTP to Asterisk Im able to call between RTP using rtpproxy as recording is a must and rtpengine does not record yet. But im not able to call between SRTP extensions, i understand also SRTP to RTP would work as im doing with Asterisk (Only the speak SRTP as rtpengine trasncode) I have this code on NATMANAGE but i do not know what could be wrong to handle SRTP Any idea? Thanks in advance if(src_ip==ASTERISK1_LAN || src_ip==ASTERISK2_LAN ){ if (sdp_get_line_startswith("$avp(mline)", "m=")) { if ($avp(mline) =~ "AVP") { add_rr_param(";rtp=AVP-kamtoast"); set_rtp_proxy_set("1"); rtpproxy_manage("rwei"); } if ($avp(mline) =~ "SAVP") { add_rr_param(";rtp=SAVP-kamtoast"); rtpengine_manage("trust-address direction=external replace-origin replace-session-connection ICE=remove RTP AVP"); } } } else{ if (sdp_get_line_startswith("$avp(mline)", "m=")) { if ($avp(mline) =~ "AVP") { add_rr_param(";rtp=AVP-asttokam"); set_rtp_proxy_set("1"); rtpproxy_manage("rwie"); } if ($avp(mline) =~ "SAVP") { add_rr_param(";rtp=SAVP-asttokam"); rtpengine_manage("trust-address direction=external replace-origin replace-session-connection ICE=remove "); } } } If you need any more info let me know.

4 12

Missing BYE at the far end
by Kliment Toshkov, Netfinity JSC 15 Jul '15

15 Jul '15

Dear colleagues, We have an issue with a couple of peers - when the called party hangs up, they (the peer) does not receive the appropriate BYE message and call appears as still connected at peers end. Then it drops after 30 seconds and sends us BYE. Current setup is the following: PEER IP > KAMAILIO 4.2.2 > (Asterisk PBX 1, 2, 3, …, N) We provision only termination services, which is: 1) peer calls us 2) Kamailio chooses destination using dispatcher module 3) forwards the call to it. Attached is captured trace. Clearly BYE is retransmitted a few times, then request for UPDATE is received but it is not possible to answer it because the call is already hung up (as far as I understand). Kamailio routing file is available upon request. Thank you in advance! Климент Тошков Изпълнителен директор 24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/> t: +359 2 4918888 f: +359 2 4815555 <http://www.netfinity.bg/> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.

1 0

ISUP decode and store to be able to append_body_part() later
by andre second 15 Jul '15

15 Jul '15

Hi, I am trying to extract and save ISUP from SIP-I packet for a later use. The problem is that data is stored in binary format and needs decoding before saving to a variable. Can Kamailio perform any decoding so that later I can use append_body_part() to append it back to a multipart message? Thanks,Andrei

1 0

License for SIP-Router server when used as part of a commercial product development
by Mahesh Kumaraguru 15 Jul '15

15 Jul '15

We wish to use SIP-Router server for our commercial product development. We plan to develop SIP clients for Android, iOS, Windows phone and PC browser to have voice and video calls. Please let us know the licensing terms under which SIP-Router server is available, especially in the context of us using it for our commercial product. Many Thanks and Best Regards, Mahesh Kumaraguru +(91)-942-417-4471 +(91)-835-782-9220 mahesh.kumaraguru(a)vectramind.com<mailto:mahesh.kumaraguru@vectramind.com> http://www.vectramind.com/

1 0

License for iptel SIP server when used as part of a commercial product development
by Mahesh Kumaraguru 15 Jul '15

15 Jul '15

We wish to use iptel SIP server for our commercial product development. We plan to develop SIP clients for Android, iOS, Windows phone and PC browser to have voice and video calls. Please let us know the licensing terms under which iptel SIP server is available, especially in the context of us using it for our commercial product. Many Thanks and Best Regards, Mahesh Kumaraguru +(91)-942-417-4471 +(91)-835-782-9220 mahesh.kumaraguru(a)vectramind.com<mailto:mahesh.kumaraguru@vectramind.com> http://www.vectramind.com/

1 0

Issues with Yealink SDP and sdp_with_transport()
by Alberto Sagredo 14 Jul '15

14 Jul '15

Im using if(sdp_with_transport("RTP/SAVP")) to detect with endpoint is send SAVP or not to divert to and rtp proxy or rtpengine, as you know rtpproxy supports recording and rtpengine does not yet. So when using if(sdp_with_transport("RTP/SAVP")) with Grandstream Phones all worked fine, but when configuring Optional or Compulsory SRTP in Yealink it seems to do not detect i have seen that crypto lines are not in the final SDP but do not know if thats the reason Did you have a similar issue with Yealink? If i could get traces in anyway to help let me know. BR Alberto INVITE sip:212@10.0.1.34:15060 SIP/2.0. Record-Route: <sip:x.x.x.x:8002;r2=on;lr=on;ftag=1072578853;nat=yes>. Record-Route: <sip:x.x.x.x:8001;transport=tls;r2=on;lr=on;ftag=1072578853;nat=yes>. Via: SIP/2.0/UDP x.x.x.x.:8002;branch=z9hG4bK24c2.948e5074172530002b3bfb131ba51de6.0;i=1. Via: SIP/2.0/TLS 10.0.1.111:11891 ;received=83.x.x.x;rport=11891;branch=z9hG4bK456460360. From: "214" <sip:214@1x.x.x.x:8001>;tag=1072578853. To: <sip:212@x.x.x.x:8001>. Call-ID: 0_1310998066(a)10.0.1.111. CSeq: 2 INVITE. Contact: <sip:214@83.x.x.x:11891;transport=TLS>. Content-Type: application/sdp. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE. Max-Forwards: 69. User-Agent: Yealink SIP-T21P_E2 52.80.0.3. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 553. . v=0. o=- 20143 20143 IN IP4 x.x.x.x. s=SDP data. c=IN IP4 x.x.x.x t=0 0. m=audio 8530 RTP/AVP 0 8 18 101. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:N2RjYzlhMjNmMzAwMDU5YzU2YjQ4ZTU1ODE4MzNm. a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NWQwYzgzMzhlYmU1OGY2NThmMzk2NjYwMTllZWI3. a=crypto:3 F8_128_HMAC_SHA1_80 inline:YjEyN2M5Nzk4YzRmZDQ5ZTYxZGUzNTI3Yzg1YTgw. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=nortpproxy:yes

2 2

Kamailio Load balancing Across Multiple Asterisk Servers -Chanspy
by Jonathan Hunter 14 Jul '15

14 Jul '15

Hi guys, I have a kamailio load balancing requests across 2 Asterisk Servers, and all is working well. Only problem now is when utilizing feature codes to look to listen into calls using the chanspy application I am having issues. I am passing calls to *2222 into Asterisk, where an IVR answers the call, then sends an INVITE back to kamailio to check to see if the extension I want to listen into is in an active call (hash table), and the call will then be forwarded to the Asterisk in question and chanspy applied so it can be listened to. The problem I have is when I send an INVITE from Asterisk Server, and then kamailio sends an INVITE back to the same Asterisk Server I get the following failure in asterisk console; Not accepting call completion offers from call-forward recipient I appreciate its potentially more an Asterisk question but I wondered if anyone had experienced the same issue? I have setup a completely load balanced environment so dont want to use Primary/Secondary Server routing. Thanks Jon

2 4

Nat Detect does not work with GS Phones
by Alberto Sagredo 14 Jul '15

14 Jul '15

I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on this phones Usual nat_uac_test with numbers 19 and 3 does not seems to detect is behind nat so NATMANAGE is not called Here its some trace. ANy clue how to handle this phones if they activate STUN for example U 80.26.x.x:52768 -> 192.168.0.170:8002 INVITE sip:2@x.x.x.x:8002 SIP/2.0. Via: SIP/2.0/UDP 80.26.x.x:52768;branch=z9hG4bK358742535;rport. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1478444456. To: <sip:2@x.x.x.x:8002>. Call-ID: 244257786-52768-56(a)IA.CG.BIE.BCH. CSeq: 550 INVITE. Contact: "Anonymous" <sip:212@80.26.x.x:52768>. X-Grandstream-PBX: true. Max-Forwards: 70. User-Agent: Grandstream GXP2140 1.0.4.23. Privacy: id. P-Preferred-Identity: <sip:212@x.x.x.x:8002>. Supported: replaces, path, timer. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 335. . v=0. o=212 8000 8000 IN IP4 80.26.x.x. s=SIP Call. c=IN IP4 80.26.x.x. t=0 0. m=audio 55422 RTP/AVP 0 8 18 9 2 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:9 G722/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19)) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }

4 6

Possible problem with multiple threads handling same SIP message
by Dirk Teurlings - SIGNET B.V. 14 Jul '15

14 Jul '15

Hello, Running kamailio 4.2.5 for a few weeks now, coming from 4.2.3. Since the upgrade we increased the logging substantially in our routing to get more insight into what is going on. We noticed that in a few cases the log showed a SIP message was being threated by multiple forks, each considering if it was a non-existing call. e.g. Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17170]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking service restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17170]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking international restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17171]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking service restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17171]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking international restrictions As you can see this shows different threads handling the exact same callid (printed by $ci). And both run the same checks in our routing plan. The main problem is, I'm not able to reproduce this on the testplatform, but only see it happen on the main systems. Also, it doesn't happen that often, in this case there are about maybe 10 similar calls like this over the run of a few weeks. This case show the call being handled by 2 threads, but we've even seen with one call it being handled by 4 threads, all with the same $ci and the same messages, just different PIDs. I've checked and made sure the routing isn't recursively calling itself. But even if that was true, I would imagine the same PID for all the log entries. Running production in debug mode isn't an option, as it will interfere too much with the day-to-day business. So I'm hoping to get a few pointers here maybe, I'm happy to look at the source code more in depth, I just need a little help in the right direction to not get lost in other code. Any help is much appreciated. Cheers, Dirk

2 2

Re: [SR-Users] code coverage in kamailio using gcov
by Daniel-Constantin Mierla 14 Jul '15

14 Jul '15

Hello, On 09/07/15 12:54, Surendra Pullaiah wrote: > > Hello > > > > I am trying to implement code coverage tool in > kamailio using gcov. I have added CFLAGS and LIBS in Makefile.defs and > it got compiled successfully. > > My problem is when I am running kamailio it is saying > lib64/kamailio/libkcore.so.1: undefined symbol: __gcov_merge_add. > > > > Can anyone help me on this. > Maybe those defines and libs need to be added to internal libraries -- see the Makefiles inside the folders of lib/ directory in kamailio source tree. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

Kamailio as pass through proxy
by kai.ohnacker＠cbc.de 14 Jul '15

14 Jul '15

Hello, I try to setup a SIP infrastructure with Kamailio server in private network and a pass through Kamailio proxy in the DMZ for NAT handling. (-> TEST.jpg) The pass through proxy uses a new routing algorithm and this works fine for external calls to internal calls. (->Kamailio.cfg.txt) The logic behind is if a packet on external interface arrives, the proxy change the URI and forward the packet to the internal SIP server. Otherwise, the proxy shall forward the packet to external interface if a packet arrives at the internal interface. Because the SIP server has a fix IP, it's easy to route to this server, but the way back is more complicated and I think my algorithm is not working. But the first problem is the routing behavior from the private server if I try to make a call from intern to extern. The server sends the request to the external IP address from the client and the IP address from the pass through Kamailio proxy is not considered. 8(2720) ERROR: <core> [udp_server.c:576]: udp_send(): sendto(sock,0x7f4724dd9238,1328,0,192.168.1.20:5060,16): Network is unreachable(101) 8(2720) ERROR: tm [../../forward.h:208]: msg_send(): udp_send failed 8(2720) WARNING: tm [t_fwd.c:1610]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed 8(2720) exec: *** cfgtrace:request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=553 a=24 n=sl_reply_error 8(2720) ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) 8(2720) exec: *** cfgtrace:request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=555 a=2 n=exit 7(2718) exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=474 a=5 n=route To change the IP address which is used from the private server as the destination address, I added the following lines to the Kamailio.cfg. modparam("registrar", "use_path",1) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received",1) modparam("path", "use_received", 1) But the routing behavior has not changed. (perhaps this configure was not the right) Is there a way to configure the private server that the last IP address from the record-route header is used? Cheers, Kai

2 1

Removing headers in a failure route on 3xx redirect
by Daniel Tryba 13 Jul '15

13 Jul '15

I'm adding and removing headers in the standard route[RELAY], but if there is a redirection involved headers aren't removed. route[RELAY] { #stdstuff route(ADDCHECKSUM); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } route[ADDCHECKSUM] { if($avp(dst_accountcode) || $avp(src_accountcode)) { #some magic remove_hf("X-rand"); remove_hf("X-csum"); append_hf("X-rand: $var(rand)\r\n"); append_hf("X-csum: $var(checksum)\r\n"); } return; } This works well unless the destination is a redirecting sip server. failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } if (t_check_status("3[0-9][0-9]")) { #... route(RELAY); exit; } -user calls destination. -dispatcher sets destination is send to redirectserver and routes to RELAY, addchecksums add some headers called from relay. -redirectserver replies 302. -failure route triggers and ends with a route to RELAY. Any existing old values should be removed and the newer ones added. But end the result is that I have two sets op X-rand/csum headers for the INVITE to the redirection contact. How do I get rid of every X-csum/rand header except the last added ones? Version tested: 4.2.5.

2 2

Pipelimit
by Fred Posner 13 Jul '15

13 Jul '15

I know in ratelimit, you can define a method within queues... for example, modparam("ratelimit", "queue", "3:INVITE") to match on INVITE method. With pipelimit, is the method called from the config, such as: if(is_method("INVITE")) { # Checking limit on INVITE if (!pl_check("$au", "traildrop", "$var(limita)")) { pl_drop(); exit; } } if(is_method("REGISTER")) { # Checking limit on REGISTER if (!pl_check("$au", "traildrop", "$var(limitb)")) { pl_drop(); exit; } } -- Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

2 1

msilo delete rows
by Evgeny Zhavoronkov 13 Jul '15

13 Jul '15

Hi All, I'm fighting with msilo module now. I recive an offline message, but msilo gets 408 status and think that a message wasn't sent succesfully. According to the log, the error somewhere in t_relay, but I couldn't find where exactly. Therefore, msilo doesn't delete rows from silo table in database and everytime the user reregisters he gets offline messages again and again. -- lifayk e: lifayk(a)gmail.com pgp: 0x84CAC40B

1 0

Does Kamailio support Cx interface Diameter over SCTP, somebody tested it?
by Jan Devai 13 Jul '15

13 Jul '15

Hi Team, Does anybody know if Kamailio support Cx interface Diameter over SCTP, somebody tested it? Thank you Jan

2 1

working config
by lifayk 13 Jul '15

13 Jul '15

Hello All! Can somebody share a working/example config with biderectional nathelper, tcp transport and push notifications (tsilo) for Kamailio >=4.3? Thanks in advance! -- lifayk e: lifayk(a)gmail.com pgp: 0x84CAC40B

2 1

sip: uri to tel: uri
by manuel espinoza 13 Jul '15

13 Jul '15

Hi, The arch looks like this SBC1 <-------- SIP-Router (Kamailio) <------------- Asterisk Asterisk sends an invite to kamailio using "sip:" as request uri scheme, but SBC1 expects that invite to arrive using tel:, is there a way to change this in kamailio and forward that invite? Something like: $ru = "tel: +" + $rU; Thanks!!!

2 1

sctp on Centos 7
by micunek＠gmail.com 13 Jul '15

13 Jul '15

Hi, I would like to enable and use sctp in kamailio but I do not know exactly how to do it. It seems that documentation is a little bit inconsistent. 1. I have Centos 7 and kamailio installed from repositories. 2. I set loadmodule "sctp" in kamailio.cfg 3. I got this error: Jul 9 08:08:54 kamailio: : sctp [sctp_mod.c:99]: mod_register(): sctp core support not enabled Jul 9 08:08:54 kamailio: ERROR: <core> [sr_module.c:600]: load_module(): /usr/lib64/kamailio/modules/sctp.so: mod_register failed Jul 9 08:08:54 cw-aio kamailio: : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio.cfg, line 230, column 12-17: failed to load module 4. I checked mod_register() function and found out that USE_SCTP should be set. 5. kamailio -V did not show USE_SCTP 6. Next I tried to compile kamailio sources from 4.3 branch make include_modules="db_mysql tls sctp" SCTP=1 cfg make all 7. Still the same problem "mod_register(): sctp core support not enabled" 8. Then I added USE_SCTP to C_DEFS and compile again. This attempt was successfull. I was able to see USE_SCTP in kamailio -V output. Questions: 1. What is the right procedure to enable sctp in kamailio? 2. Why the -DUSE_SCTP was not used when SCTP=1 was used. According to documentation: "The core Makefile variable SCTP must be set to 1 (which is by default set to 1 in Makefile.defs) and sources compiled with -DUSE_SCTP (automatically set when SCTP=1). In other words, if core Makefiles are not changed and SCTP variable is not overwritten from command line, then the SCTP support in core is enabled." Regards, Peter

2 1

nat_uac_test question
by Kristian Kielhofner 13 Jul '15

13 Jul '15

Hello, I'm testing some basic flags with nat_uac_test. I have the following (rudimentary) script: route[NATDETECT] { #!ifdef WITH_NAT if (has_body("application/sdp")) { if (nat_uac_test("18")) { xlog("SCRIPT: SDP NAT Detected\n"); setflag(6); } } else { if (nat_uac_test("2")) { xlog("SCRIPT: Other NAT Detected\n"); setflag(6); } } if (isflagset(6)) { xlog("SCRIPT: NAT detected\n"); force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } When presented with this REGISTER message (ngrep output, so first line is IP source address and port): U 192.168.113.17:6060 -> 192.168.113.11:5060 REGISTER sip:192.168.113.11;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.113.17:6060;branch=z9hG4bKOzvuzaIa;rport. From: <sip:user_1@192.168.113.11>;tag=3F785909-559E7E9F0009321B-B5BFFB40. To: <sip:user_1@192.168.113.11>. CSeq: 10 REGISTER. Call-ID: 1C193C6B-559E7E9F00093255-B5BFFB40. Max-Forwards: 70. User-Agent: FreeSWITCH. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Contact: <sip:gw+sip0@10.40.47.190:5080;transport=udp;gw=sip0>;expires=3600. Content-Length: 0. . U 192.168.113.11:5060 -> 192.168.113.17:6060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.113.17:6060;branch=z9hG4bKOzvuzaIa;rport=6060;received=192.168.113.17. From: <sip:user_1@192.168.113.11>;tag=3F785909-559E7E9F0009321B-B5BFFB40. To: <sip:user_1@192.168.113.11>;tag=b27e1a1d33761e85846fc98f5f3a7e58.1045. CSeq: 10 REGISTER. Call-ID: 1C193C6B-559E7E9F00093255-B5BFFB40. Contact: <sip:gw+sip0@10.40.47.190:5080;transport=udp;gw=sip0>;expires=3600;received="sip:192.168.113.17:6060". Server: Kamailio. Content-Length: 0. . Log output: Jul 9 10:00:41 kamailio /usr/sbin/kamailio[3886]: ERROR: <script>: SCRIPT: Other NAT Detected Jul 9 10:00:41 kamailio /usr/sbin/kamailio[3886]: ERROR: <script>: SCRIPT: NAT detected Log output, the presence of the added "received" parameter in the returned Contact, etc show that NAT was detected. However, the IP source and port match the one presented in the Via header. What am I missing here? Thanks! -- Kristian Kielhofner

2 1

pua_publish MWI message via xmlrpc
by Asgaroth 10 Jul '15

10 Jul '15

Hi All, I am having difficulty getting pua_publish to issue a NOTIFY when I send an XMLRPC command to generate a PUBLISH into the presence server. I have been reading the pua_mi module docs, but I cannot see where I am going wrong with the following xmlrpc command, please would someone point me in the correct direction. I have an XML file which is the basis of the XMLRPC request via curl, I have attached it to this email for you to view, please would you take a look and let me know where I am going wrong, I suspect it is my message body as the pua module complains about the message body being NULL: Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua [add_events.c:125]: pres_process_body(): while parsing xml memory Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua [send_publish.c:650]: send_publish(): while processing body Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua [send_publish.c:652]: send_publish(): NULL body Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua_mi [mi_func.c:290]: mi_pua_publish(): sending publish failed Kamailio verion is "version: kamailio 4.3.0 (x86_64/linux) c6aa95" I can issue other XMLRPC commands, urloc lookups method lists and those work, I just cant see where I am going wrong with this one. Any pointers would be greatly appreciated. Thanks

1 1

dmq/dmq_userloc crash on startup (kamailio 4.3.0)
by Asgaroth 10 Jul '15

10 Jul '15

Hi All, Another thing I noticed while playing with kamailio and dmq/dmq_usrloc is that 7 out of 10 times the kamailio process crashes when I initially start it up. Any tips/requests for further debugging would be greatly appreciated. Kamailio version: version: kamailio 4.3.0 (x86_64/linux) c6aa95 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: c6aa95 compiled on 16:14:27 Jun 23 2015 with gcc 4.4.7 I start kamailio as follows, from commandline: /usr/sbin/kamailio -P /var/run/kamailio_registrar01/kamailio.pid -f /shared/kamailio/etc/kamailio_registrar.cfg -l "udp:(10.6.0.173, 10.7.0.173)" -m 1024 -M64 -A RESOURCE_REGISTRAR01 -l tcp:10.6.0.173:80 -u root -g root -ddd DMQ related settings are as follows: # grep -i dmq module_definitions.cfg loadmodule "dmq.so" loadmodule "dmq_usrloc.so" # ----- dmq params ----- modparam("dmq", "server_address", DMQ_ADDRESS) modparam("dmq", "notification_address", "sip:sip-registrars.internal_domain.com:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) # ----- dmq_usrloc params ----- modparam("dmq_usrloc", "enable", 1) DMQ_ADDRESS is defined as "sip:10.6.0.173:5060" in this servers case. sip-registrars.internal_domain.com resolves to the following address list 10.6.0.173, 10.6.0.174, 10.6.0.175 Here is the debug log of the crash: Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [worker.c:82]: worker_loop(): dmq_worker [0 27445] getting lock Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [worker.c:84]: worker_loop(): dmq_worker [0 27445] lock acquired Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=66fabe782d71b7a9842d4a6a5eb93c4d-1de2 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:454]: dmq_notification_callback(): dmq triggered from dmq_notification_callback Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmqnode.c:153]: build_dmq_node(): build_dmq_node sip:10.6.0.173:5060;status=active with private memory Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmqnode.c:153]: build_dmq_node(): build_dmq_node sip:10.6.0.175:5060;status=active with private memory Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmqnode.c:153]: build_dmq_node(): build_dmq_node sip:10.6.0.174:5060;status=active with private memory Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:468]: dmq_notification_callback(): received 0 new or changed nodes Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:528]: build_notification_body(): body_len = 1024 - clen = 0 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:528]: build_notification_body(): body_len = 1024 - clen = 35 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:528]: build_notification_body(): body_len = 1024 - clen = 70 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:388]: usrloc_dmq_request_sync(): requesting sync from dmq peers Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:404]: usrloc_dmq_request_sync(): sending serialized data {"action":3} Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:226]: usrloc_dmq_send(): sending dmq broadcast... Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==10 && [10.6.0.173] == [10.6.0.173] Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:usrloc@10.6.0.174:5060> Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 24685 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==10 && [10.6.0.173] == [10.6.0.173] Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:usrloc@10.6.0.175:5060> Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 24686 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmq_funcs.c:163]: bcast_dmq_message(): skipping node sip:10.6.0.173:5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [msg_translator.c:158]: check_via_address(): (10.6.0.174, 10.6.0.174, 0) Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [worker.c:134]: worker_loop(): sent reply Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=66fabe782d71b7a9842d4a6a5eb93c4d-1ac2 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:250]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@10.6.0.174:5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:360]: usrloc_dmq_handle_msg(): Received DMQ_SYNC. Sending all contacts... And here is the backtrace from the core file: # gdb /usr/sbin/kamailio /core.27445 GNU gdb (GDB) Red Hat Enterprise Linux (7.2-75.el6) Copyright (C) 2010 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/sbin/kamailio...(no debugging symbols found)...done. [New Thread 27445] Reading symbols from /lib64/libdl.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libdl.so.2 Reading symbols from /lib64/libresolv.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libresolv.so.2 Reading symbols from /lib64/libc.so.6...(no debugging symbols found)...done. Loaded symbols for /lib64/libc.so.6 Reading symbols from /lib64/ld-linux-x86-64.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/ld-linux-x86-64.so.2 Reading symbols from /usr/lib64/kamailio/modules/db_unixodbc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/db_unixodbc.so Reading symbols from /usr/lib64/libodbc.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libodbc.so.2 Reading symbols from /usr/lib64/kamailio/libsrdb1.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libsrdb1.so.1 Reading symbols from /usr/lib64/kamailio/libkcore.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libkcore.so.1 Reading symbols from /usr/lib64/libltdl.so.7...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libltdl.so.7 Reading symbols from /lib64/libpthread.so.0...(no debugging symbols found)...done. [Thread debugging using libthread_db enabled] Loaded symbols for /lib64/libpthread.so.0 Reading symbols from /usr/lib64/kamailio/modules/mi_fifo.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/mi_fifo.so Reading symbols from /usr/lib64/kamailio/libkmi.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libkmi.so.1 Reading symbols from /usr/lib64/kamailio/modules/tm.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/tm.so Reading symbols from /usr/lib64/kamailio/modules/sl.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/sl.so Reading symbols from /usr/lib64/kamailio/modules/pv.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/pv.so Reading symbols from /usr/lib64/kamailio/libsrutils.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libsrutils.so.1 Reading symbols from /lib64/libm.so.6...(no debugging symbols found)...done. Loaded symbols for /lib64/libm.so.6 Reading symbols from /usr/lib64/kamailio/modules/kex.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/kex.so Reading symbols from /usr/lib64/kamailio/modules/usrloc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/usrloc.so Reading symbols from /usr/lib64/kamailio/modules/registrar.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/registrar.so Reading symbols from /usr/lib64/kamailio/modules/xlog.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/xlog.so Reading symbols from /usr/lib64/kamailio/modules/auth.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/auth.so Reading symbols from /usr/lib64/kamailio/modules/auth_db.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/auth_db.so Reading symbols from /usr/lib64/kamailio/modules/rr.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/rr.so Reading symbols from /usr/lib64/kamailio/modules/path.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/path.so Reading symbols from /usr/lib64/kamailio/modules/nathelper.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/nathelper.so Reading symbols from /usr/lib64/kamailio/modules/siputils.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/siputils.so Reading symbols from /usr/lib64/kamailio/modules/textops.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/textops.so Reading symbols from /usr/lib64/kamailio/modules/xmlrpc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/xmlrpc.so Reading symbols from /usr/lib64/libxml2.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libxml2.so.2 Reading symbols from /lib64/libz.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libz.so.1 Reading symbols from /usr/lib64/kamailio/modules/ctl.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/ctl.so Reading symbols from /usr/lib64/kamailio/modules/cfg_rpc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/cfg_rpc.so Reading symbols from /usr/lib64/kamailio/modules/dmq.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/dmq.so Reading symbols from /usr/lib64/kamailio/modules/dmq_usrloc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/dmq_usrloc.so Reading symbols from /lib64/libnss_files.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libnss_files.so.2 Reading symbols from /usr/lib64/gconv/ISO8859-1.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/gconv/ISO8859-1.so Reading symbols from /usr/lib64/psqlodbc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/psqlodbc.so Reading symbols from /usr/lib64/libssl.so.10...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libssl.so.10 Reading symbols from /usr/lib64/libpq.so.5...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libpq.so.5 Reading symbols from /usr/lib64/libodbcinst.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libodbcinst.so.2 Reading symbols from /lib64/libgssapi_krb5.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libgssapi_krb5.so.2 Reading symbols from /lib64/libkrb5.so.3...(no debugging symbols found)...done. Loaded symbols for /lib64/libkrb5.so.3 Reading symbols from /lib64/libcom_err.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libcom_err.so.2 Reading symbols from /lib64/libk5crypto.so.3...(no debugging symbols found)...done. Loaded symbols for /lib64/libk5crypto.so.3 Reading symbols from /usr/lib64/libcrypto.so.10...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libcrypto.so.10 Reading symbols from /lib64/libcrypt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libcrypt.so.1 Reading symbols from /lib64/libldap_r-2.4.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libldap_r-2.4.so.2 Reading symbols from /lib64/libkrb5support.so.0...(no debugging symbols found)...done. Loaded symbols for /lib64/libkrb5support.so.0 Reading symbols from /lib64/libkeyutils.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libkeyutils.so.1 Reading symbols from /lib64/libfreebl3.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libfreebl3.so Reading symbols from /lib64/liblber-2.4.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/liblber-2.4.so.2 Reading symbols from /usr/lib64/libsasl2.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libsasl2.so.2 Reading symbols from /usr/lib64/libssl3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libssl3.so Reading symbols from /usr/lib64/libsmime3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libsmime3.so Reading symbols from /usr/lib64/libnss3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libnss3.so Reading symbols from /usr/lib64/libnssutil3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libnssutil3.so Reading symbols from /lib64/libplds4.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libplds4.so Reading symbols from /lib64/libplc4.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libplc4.so Reading symbols from /lib64/libnspr4.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libnspr4.so Reading symbols from /lib64/libselinux.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libselinux.so.1 Reading symbols from /lib64/librt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/librt.so.1 Reading symbols from /lib64/libnss_dns.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libnss_dns.so.2 Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio_registrar01/kamailio.pid -f /shared/kam'. Program terminated with signal 11, Segmentation fault. #0 0x00007fdad84d76a5 in memcpy () from /lib64/libc.so.6 Missing separate debuginfos, use: debuginfo-install kamailio-4.3.0-0.el6.bf.x86_64 (gdb) bt #0 0x00007fdad84d76a5 in memcpy () from /lib64/libc.so.6 #1 0x00007fdacfeff03d in usrloc_get_all_ucontact () from /usr/lib64/kamailio/modules/dmq_usrloc.so #2 0x00007fdacff02dfd in usrloc_dmq_handle_msg () from /usr/lib64/kamailio/modules/dmq_usrloc.so #3 0x00007fdad012d4a7 in worker_loop () from /usr/lib64/kamailio/modules/dmq.so #4 0x00007fdad011232c in ?? () from /usr/lib64/kamailio/modules/dmq.so #5 0x0000000000592675 in ?? () #6 0x00000000005923b0 in ?? () #7 0x000000000059297f in init_child () #8 0x00000000004a7680 in main_loop () #9 0x00000000004acd36 in main () (gdb) ^C(gdb) Quit

2 4

dialog: use keepalive & timeout features together
by Andrey Utkin 09 Jul '15

09 Jul '15

Would be nice if it would be possible to - make timer of "default_timeout" reset to 0 when keep-alive reply is received from remote part of call leg, so that the timeout expires (triggering call termination) only if some call party gets offline (we don't want just to limit the duration of normal call, even if call parties don't send any new SIP requests); OR - make unresponsiveness of call parties to keep-alive messages (controlled with ka_timer, ka_interval parameters) lead to some action, like call termination. Please advise if I miss something, and how to implement this better. Thanks in advance. -- Andrey Utkin

2 3

Re: [SR-Users] Kamailio deployment
by Daniel Tryba 09 Jul '15

09 Jul '15

On Thursday 09 July 2015 17:46:22 Srujan Kumar K wrote: > I am connecting my kamailio server to MSC for my solution which has both > inbound and outbound. ... > Kamailio server IP : 172.22.14.12 > MSC IP : 172.28.0.68 Why does you kamailio config bind to the msc ipadress?

1 0

[Kamailio-Users] HA+DRBD setup
by Siddhardha Garige 09 Jul '15

09 Jul '15

Hello all, Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio. Especially creating partitions, DRBD devices and mount points. Thanks in advance, -Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

11 24

Kamailio deployment
by Srujan Kumar K 09 Jul '15

09 Jul '15

Hai, I am connecting my kamailio server to MSC for my solution which has both inbound and outbound. The SIP trunk is not UP and we don't see any packets transmitted between the 2 nodes. We are also trying to load SIPI/SIPT module Kamailio server IP : 172.22.14.12 MSC IP : 172.28.0.68 I have also attached configuration file with this mail. Requesting your help on this. Regards ~Srujan

1 0

Moving from haproxy to db_cluster
by Jan Hazenberg 09 Jul '15

09 Jul '15

Hi, I have a question about the db_cluster module. We are currently using a galera mysql cluster from severalnines, in this setup haproxy is installed on the kamailio nodes handling the database failover. The question is if a database node fails and the TCP connection breaks the sql query will probably fail when using haproxy since it is not aware of the query. Will the db_cluster module send the query to the next available mysql node in the cluster without a failing query? If thats the case i should not notice the failing dbnode in my kamailio script. In other words, will there be a advantage in using the db_cluster module instead of haproxy? Thanks, Jan Hazenberg

2 3

code coverage in kamailio using gcov
by Surendra Pullaiah 09 Jul '15

09 Jul '15

Hello I am trying to implement code coverage tool in kamailio using gcov. I have added CFLAGS and LIBS in Makefile.defs and it got compiled successfully. My problem is when I am running kamailio it is saying lib64/kamailio/libkcore.so.1: undefined symbol: __gcov_merge_add. Can anyone help me on this. Regards Surendra

1 0

Function sdp_remove_codecs_by_id seems to be not working
by José Seabra 09 Jul '15

09 Jul '15

Hello, I'm using the function sdp_remove_codecs_by_id from sdpops module in order to remove some codecs in INVITE request before send it to freeswitch, but the function doesn't remove the codec, and it doesn't give any error message. I'm using this function in request route. Kamailio version is 4.2.2. INVITE that kamailio receives from phone: INVITE sip:401@teste.d <sip%3A401(a)teste.itcenter.com.pt>emo.pt;user=phone SIP/2.0 Record-Route: <sip:10.0.20.102:5062 ;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> Via: SIP/2.0/UDP 10.0.20.102:5062 ;branch=z9hG4bKecf3.3ff3f7e77d2abc0fd3f74c61eeb68a0b.0 Via: SIP/2.0/UDP 192.168.10.147:5060 ;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060 From: "301" <sip:301@teste.demo.pt <sip%3A301(a)teste.itcenter.com.pt> >;tag=oztyflbzbx To: <sip:401@teste.demo.pt <sip%3A401(a)teste.itcenter.com.pt>;user=phone> Call-ID: 3c3a58a25d63-ghfc5xdg1sn0 CSeq: 1 INVITE Max-Forwards: 69 Contact: <sip:301@192.168.10.147:5060 ;alias=100.64.250.254~5060~1;line=c1r2c8u6>;reg-id=1 X-Serialnumber: 000413262FA0 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/8.4.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:teste.demo.pt <http://teste.itcenter.com.pt> >;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 391 v=0 o=root 24935823 24935823 IN IP4 192.168.10.147 s=call c=IN IP4 192.168.10.147 t=0 0 m=audio 19410 RTP/AVP 0 8 9 99 3 18 4 101 a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv INVITE that kamailio send to freeswitch after execute sdp_remove_codecs_by_id("18"): INVITE sip:401@teste.demo.pt;user=phone SIP/2.0. Record-Route: <sip:10.0.20.100;lr=on;ftag=zvjgcz9zs9;proxy=yes;did=441.0eb2>. Record-Route: <sip:10.0.20.102:5062 ;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. Via: SIP/2.0/UDP 10.0.20.100;branch=z9hG4bK8711.bb31396197409170b2c1bd05b24e7f36.0. Via: SIP/2.0/UDP 10.0.20.102:5062 ;branch=z9hG4bK8711.07ffcc13fb96f90f6b4dbe4b2dfd0fa5.0. Via: SIP/2.0/UDP 192.168.10.147:5060 ;received=100.64.250.254;branch=z9hG4bK-aq7e0puz8p6o;rport=5060. From: "301" <sip:301@teste.demo.pt>;tag=zvjgcz9zs9. To: <sip:401@teste.demo.pt;user=phone>. Call-ID: 3c3a7c84e065-pr2hm0uk9yfz. CSeq: 2 INVITE. Max-Forwards: 68. Contact: <sip:301@192.168.10.147:5060 ;alias=100.64.250.254~5060~1;line=ttnfv9c7>;reg-id=1. X-Serialnumber: 000413262FA0. P-Key-Flags: resolution="31x13", keys="4". User-Agent: snom370/8.4.35. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Call-Info: <sip:teste.itcenter.com.pt>;appearance-index=1. Session-Expires: 3600;refresher=uas. Min-SE: 90. Content-Type: application/sdp. Content-Length: 403. . v=0. o=root 228603317 228603317 IN IP4 100.64.250.4. s=call. c=IN IP4 100.64.250.4. t=0 0. m=audio 49404 RTP/AVP 0 8 9 99 3 18 4 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:9 G722/8000. a=rtpmap:99 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:49405. SDP body has no changes related with codecs. Anyone call help please. Thank you BR José Seabra -- Cumprimentos José Seabra

3 6

Textops and Multipart Body - adding ISUP segfaults Kamailio
by andre second 09 Jul '15

09 Jul '15

Hi, I am using Kamailio version: kamailio 4.3.0 (x86_64/linux) c6aa95 on CentOS 6 I am trying to encapsulate ISUP in the INVITE: ...if(has_body("application/sdp")){ set_body_multipart(); if(msg_apply_changes()) { $var(acm) = "7e Od 04 55 75 69 20 4d 61 6b 65 43 61 6c 6c"; append_body_part("$var(acm)","application/isup; version=itu-t92+","signal; handling=optional"); xlog("L_INFO", "ISUP Changes Applied Succesfully"); }} ...And getting the following error: Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1691]: get_boundary(): boundary is <--unique-boundary-1>...Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1808]: check_boundaries(): last bondary without -- at the end Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1612]: replace_body(): old size body[365] actual[534]Jul 7 14:59:12 ukrcld1000 kamailio: CRITICAL: <core> [data_lump.c:463]: free_lump(): non free-able lump: 0x7f7a33d5ae40 flags=1 and then Kamailio crashes. Can you please recommend? Thanks,Andrei

2 3

usleep with variable
by Antanas Masevicius 09 Jul '15

09 Jul '15

Hello, been struggling to make a random sleep with cfgutils functions. ex: $avp(i:444) = 123; usleep("$avp(i:444)"); gives an error: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio-sp1new.cfg:723 Kamailio version: 4.3 If usleep accepts only numbers, is there any other way to sleep by passing variable? best regards, Antanas -- # <#> Antanas Masevičius Technikos Direktorius UAB Nacionalinis telekomunikacijų tinklas Panerių g. 51, LT-03160 Vilnius *Tel.* 8 5 2056000 *Tiesiog. tel.* 8 700 00031 *Mob. tel.* 8 685 77730 *Faks.* 8 700 00034 *El. p.:* antanas.masevicius(a)ntt.lt <#> # www.ntt.lt <http://www.ntt.lt> | www.spykas.com <http://www.spykas.com>

1 1

MetaSwitch ignores Contact: header
by Markus 08 Jul '15

08 Jul '15

Dear list, I'm running kamailio-4.2.3-3.1.x86_64 in a very simple setup: one Asterisk server behind kamailio. But I'm using dispatcher, the other servers are just down currently. :) Different vendors route DIDs to kamailio's IP and kamailio handles the communication towards Asterisk. Now I have a new vendor who uses MetaSwitch, and there is a problem. All calls get dropped with a BYE from the Asterisk side after 32 seconds because Asterisk never receives the ACK, because kamailio doesn't forward it, because MetaSwitch sends it as ACK sip:number@KAMAILIO-IP instead of ACK sip:number@ASTERISK-IP That means MetaSwitch ignores what we tell it in the Contact: header. Is this MetaSwitch'es "fault" or kamailio's? All other vendors work fine without problems. Therefore I have compared an "old" vendor and the new one. Here is the result that shows the difference: Old vendor - works: OLDVENDOR-IP.5060 > KAMAILIO-IP.5060: INVITE sip:number@KAMAILIO-IP KAMAILIO-IP.5060 > OLDVENDOR-IP.5060: SIP/2.0 100 trying KAMAILIO-IP.5060 > ASTERISK-IP.5060: INVITE sip:number@KAMAILIO-IP ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 100 Trying ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > OLDVENDOR-IP.5060: SIP/2.0 200 OK OLDVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@ASTERISK-IP:5060 KAMAILIO-IP.5060 > ASTERISK-IP.5060: ACK sip:number@ASTERISK-IP:5060 New vendor with MetaSwitch - doesn't work: NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: INVITE sip:number@KAMAILIO-IP:5060 KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 100 trying KAMAILIO-IP.5060 > ASTERISK-IP.5060: INVITE sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 100 Trying ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ... and so on ... Can I do something about this or is a change on MetaSwitch'es side in order? Thank you! Markus

3 3

BYE request with Sipp refused
by Loic Chabert 08 Jul '15

08 Jul '15

Hello everyone, I'm a newbie in kamailio world and make some tests to dispatch call over several asterisk instances. This part works, my call has been distributed over two instances (if i make a call with jitsi for example). Now, i want to test performance of my cluster and use Sipp to chalenge it. All method is working fine, except the BYE method, where kamailio send me back a 404 Not here. This is my SIPP scenario: ... <pause/>  <send retrans="500"> <![CDATA[ BYE sip:[service]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@loicchabert.fr>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> ... I use the classic file of kamailio, with some adjustement for dispatching. Any ideas concerning this problem ? This BYE is not in transaction so he is drop by kamailio ? Thank in advance, Regards.

3 6

Sending a message form within kamailio
by Shouvanik Chakrabarti 08 Jul '15

08 Jul '15

Hello, I am writing a module for kamailio and as part of its functionality I need to generate and send SIP messages. How would I go about sending an SIP message from within kamailio? Any help would be deeply appreciated. Thanks, Shouvanik

3 5

Kamailio as radius server
by Ali Taher 08 Jul '15

08 Jul '15

Hello, I'm wondering if kamailio can act as radius server who's responsible of authentication,authorization and accounting, where kamailio receive accounting record types (start-stop-interim) and write them in a database. Thanks Ali

3 3

CNXCC cnxcc_set_max_time Version 4.2
by Jonathan Hunter 08 Jul '15

08 Jul '15

Hi Guys, I have just moved from kamailio 4.1, to 4.2 and have installed via git version 4.2 I am testing with the function cnxcc_set_max_time, as I want to set the maximum call duration. Having moved to 4.2, i cant get the module to terminate the call. I have set dlg_flag, and cc_flag, following examples on the wiki but I cant get it to work. Does anyone have any suggestions for me please? My same code on 4.1 works ok. Thanks Jon

1 1

Potential ip_addr_match_net bug
by Kyle Kurz 07 Jul '15

07 Jul '15

I'm using allow_address function of the permissions module and I'm seeing strange behavior with network masks not divisible by 8. I have 172.16.0.0/12 in my address table and Kamailio is marking packets from 172.56.16.232 as being part of that local subnet. If I change the subnet to 172.16.0.0/16, I no longer have an issue. I know there was a bug several years ago related to this ( http://git.kamailio.org/gitlist/index.php/sip-router/commit/0ead7ab31bf0df6…), but that code is very different, so I'm wondering if a regression has crept in? I'm using Kam 4.1.6. Thanks, Kyle -- *Kyle Kurz* Digium, Inc. · Senior Software Engineer 9115 Brown Deer Rd · San Diego, CA 92121 · US direct +1 858.537.1166 Check us out at: http://digium.com · http://asterisk.org <http://www.asterisk.org/>

3 3

dmq_replicate deserializing?
by Asgaroth 07 Jul '15

07 Jul '15

Hi All, I am playing with the new dmq_replicate module and am banging my head against an issue I have come accross. I can see the kamailio registrars sending the replication messages to the other node(s) in the dmq bus, and the payload looks to be of type json. However, on the recieving nodes, when I try to issue a kamctl ul show --brief, the output of the AOR's are garbled. I had a look at the dmq/dmq_ursloc moduled, and the dmq module docs mention that we may need to load our own deserializers, is this the case when using the dmq_usrloc module as well? I am only using dmq to replicate registration messages. Here is a sample of the dmq workers on a recieving node applying the update, and you can see that it thinks the contact to add is 'p÷#031#002', is was expecting the actual AOR of the subriber to show up here. Am I missing something simple here or is there something more sinister at play. Any pointers would be greatly appreciated. Kamailio Version: version: kamailio 4.3.0 (x86_64/linux) c6aa95 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: c6aa95 compiled on 16:14:27 Jun 23 2015 with gcc 4.4.7 Debug log below: Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:606]: parse_msg(): SIP Request: Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): method: <KDMQ> Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): uri: <sip:usrloc@10.6.0.174:5060> Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): version: <SIP/2.0> Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK9b3c.fe993716000000000000000000000000.0>; state=16 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [receive.c:134]: receive_msg(): After parse_msg... Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [receive.c:177]: receive_msg(): preparing to run routing scripts... Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=450 global id=449 T start=0xffffffffffffffff Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [30]; uri=[sip:usrloc@10.6.0.174:5060] Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:usrloc@10.6.0.174:5060>#015#012] Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <10> <KDMQ> Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=471 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=50105, isACK=0 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=450 global id=450 T end=(nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: dmq [message.c:53]: dmq_handle_message(): dmq_handle_message [KDMQ sip:usrloc@10.6.0.174:5060] [ ] Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: dmq [message.c:65]: dmq_handle_message(): dmq_handle_message peer found: usrloc Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq [worker.c:84]: worker_loop(): dmq_worker [0 23014] lock acquired Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=390c95c339281829d3cea6f43c8512cb-7e62 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:250]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@10.6.0.173:5060 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:353]: usrloc_dmq_handle_msg(): Received DMQ_UPDATE. Update contact info... Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:60]: add_contact(): 'p÷#031#002' found in usrloc Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:62]: add_contact(): get_ucontact = 0 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:72]: add_contact(): Found contact Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: usrloc [ucontact.c:1688]: update_ucontact(): exists callback for type= UL_CONTACT_UPDATE Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: usrloc [ul_callback.h:84]: run_ul_callbacks(): contact=0x7ffe86eabad8, callback type 2/15, id 0 entered Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:494]: dmq_ul_cb_contact(): Callback from usrloc with type=2 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:517]: dmq_ul_cb_contact(): Contact recieved from DMQ... skip Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:85]: add_contact(): Release record Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:87]: add_contact(): Unlock udomain Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: <core> [msg_translator.c:158]: check_via_address(): (10.6.0.173, 10.6.0.173, 0) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [receive.c:278]: receive_msg(): cleaning up Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq [worker.c:134]: worker_loop(): sent reply Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq [worker.c:82]: worker_loop(): dmq_worker [0 23014] getting lock

2 6

SRTP to RTP with Kamailio
by Alberto Sagredo 07 Jul '15

07 Jul '15

Hi I have a Kamailio in front of an Asterisk Cluster and moved to rtpengine to give support for SRTP from rtpproxy What i want is to have SRTP on Kamailio-rtpengine leg but RTP inside network for Asterisk. I have found that config https://loadmultiplier.com/sites/default/files/kamailio/pcscf/kamailio.cfg But could not be able to make it work? Anyone did this scenareo and could give me some tips? I have enabled sdops and be able to remove SRTP and crypto lines but not be able to make it work as i mention Thanks in advanced Alberto

1 1

IMS server with a core network
by test testt 07 Jul '15

07 Jul '15

hello, I posted a message in the kamailio forum but it wasn't in the good place, so I rebroadcast at this e-mail as it was say. Hello, I have installed Kamailio 4.2 and I would like to use it as IMS server but I don't have any ideas how to do this. I have a LTE core network which includes HSS ans PCRF. I saw on this page ( http://www.kamailio.org/w/2013/05/ims-kamailio/ ) that is apparently easy, just need to take the P-CSCF, S-CSCF and I-CSCF Kamailio and Diameter config files in the examples folder and run kamailio but I don't know how to use them and I didn't find any tutorials about it. Could you help me please ? Marc link: https://github.com/kamailio/kamailio/issues/227#issuecomment-118073500 I tried the 2 tutorials that it has given but it's not really working I am using ubuntu 15.04 Thanks Marc

1 0

Dispatcher, Failure Route and fr_timeout
by Asgaroth 07 Jul '15

07 Jul '15

Hi All, I would like to reduce the time it takes for kamailio to timeout an initial message to a dispatcher destination so that it will select the next destination in the set, before, hopefully, a restransmission. In this case I am testing registration messages. I came accross the t_set_fr() function in the TM module documentation today and have been experimenting with it for some hours. I have an issue where a 408 timout is sent back to the UAC before the last destination in the dispatcher set is selected, I suspect I am hitting the timeout somewhere, but I cannot see where, I presume the fr_timeout is set every time I issue the t_set_fr() call and the associated t_relay. I'm just wondering if I am doing this correctly, if one of the guru's out there wouldnt mind casting an eyeball over this config and let me know if I am on the right track, or if I am going about this in the wrong way. Any suggestions are most welcome. What appears to be happening is, I have 3 destinations in the dispatcher set, it looks like the 1st 2 are tried, which correctly timeout and are set inactive-probing, but on the last destination, which is working, I appear to be getting a 408 timeout being sent back from kamailio. Here is a snip of my configuration: #!define NO_RESPONSE_TIMER 250 request_route { t_on_failure("DISPATCHER_FAILURE"); t_set_fr(0, NO_RESPONSE_TIMER); if(!ds_select_dst("1", "0")) { xlog("L_ERR", "No dispatcher destinations available for $ru\n"); sl_send_reply("500", "No destination available"); exit; } route("RELAY") } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } failure_route[DISPATCHER_FAILURE] { if (t_is_canceled()) { exit; } # select next destination only for local timeout if (t_branch_timeout() && !t_branch_replied()) { xlog("L_INFO", "route[DISPATCHER_FAILURE] : $rm : timeout and no reply ($si:$sp->$Ri:$Rp->$du)\n"); xlog("L_INFO", "route[DISPATCHER_FAILURE] : $rm : setting $du to probing state"); ds_mark_dst("ip"); if(ds_next_dst()) { xlog("L_INFO", "route[DISPATCHER_FAILURE] : $rm : next destination selected ($du)\n"); t_set_fr(0, NO_RESPONSE_TIMER); t_relay(); exit; } else { xlog("L_INFO", "route[DISPATCHER_FAILURE] : $rm : No destinations available for $rd"); send_reply("500", "No destination available"); exit; } } }

2 2

How to configure Cx - diameter running on SCTP
by Jan Devai 07 Jul '15

07 Jul '15

Hi guys, I would like to configure Cx interface, i.e. Diameter on SCTP between I/S-CSCF and external HSS. Can somebody Guide me how to achieve it or if somebody who already did that, can he share his configurations files with me? Thank you Jan

1 0

clarification on tcp keepalive
by Kelvin Chua 07 Jul '15

07 Jul '15

using kamailio 4.3 on ubuntu server 14.04 i noticed tcp socket will be killed by kamailio after around 11 minutes with or without tcp_keepalive=no. tcpdump shows that by default (tcp_keepalive=yes) there is indeed keepalive packets going back and forth. if set to no, there would be nothing in between. if i have clients who have registration timeout of 1700, they would send re-registration around half of that, or around 14 minutes after registration. then the socket would be long dead and they would be unreachable for 3 minutes. based on procfs, ubuntu have settings for 9 probes with 75 seconds interval so that means around 11 minutes of sending keepalives. so i tried changing these values, on procfs. and tried playing around with these kamailio cfg settings as well: tcp_keepcnt tcp_keepidle tcp_keepintvl whatever i do, the interval between keepalives are always 2 minutes and after 4-5 probes, kamailio will send a FIN to the client. i checked the source code tcp_main.c and it looks alright. could this be an ubuntu issue? anybody seen this before? I created a workaround by setting expire 900 for all registrations but this is not ideal. Kelvin Chua

2 1

error in kamctlrc line 13 not sure of "CHROOTED DIRECTORY"
by Joseph Zimmer 07 Jul '15

07 Jul '15

Sorry, this is my problem: have included the kamctlrc, kamailio.cfg, and modus.lst files if that helps, run create command, not sure of path for chrooted directory in kamctlrc file so I have not been able to create my initial database. I know that is holding up creation of MySQL database. can you assist, sent contribution to this email address, could not find any other. thanks in advance, joe zimmer +1 703-957-0186 USA root@telephonepros:~# /usr/local/sbin/kamdbctl create /usr/local/etc/kamailio//kamctlrc: line 13: =/path/to/chrooted/directory: No such file or directory MySQL password for root: INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found Usage: grep [OPTION]... PATTERN [FILE]... Try 'grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database kamailio ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not found ERROR: Creating database kamailio failed! root@telephonepros:~#

2 1

DIAMETER based on TCP requires a FIXED I/S-CSCFSOURCE PORT in order to make Cx work towards a External HSS
by Jan Devai 07 Jul '15

07 Jul '15

Hi experts, from the current trace we can see that the I/S-CSCF always changes SOURCE port when contacting our eternal HSS. In order to make the Cx work HSS expects the Source port to be fixed on the TCP layer and in our I/S-CSCF configuration we did not find any setting for SOURCE Port. Can somebody help us please ;-) ? We appreciate your support, Br, Jan

3 2

Looking for kamailio service supplier.
by ZhuYue Zhang 07 Jul '15

07 Jul '15

Hi Sir, This is a Chinese company from ABB group, We're looking for a VOIP system solution, As our understanding, kamailio is the best choice, Does any Chinese company take this business? Thank you and Best Regards. [cid:image001.png@01D02128.0536F2C0] Zhuyue.zhang ABB Genway Xiamen Electrical Equipment CO.,LTD 7F,NO.23 Wanghai Road,Software Park 2,Lvling Road Xiamen SEZ,Fujian,China 361008 Phone: (86)592-2954819 Mobile: (86)15260200863

1 0

mongodb and jansson
by Kelvin Chua 06 Jul '15

06 Jul '15

when using mongodb with jansson, intermittent failures will be seen because jansson would change the order of keys inside a json object. mongodb would expect the first key to be a command, for example "insert". this is disastrous! below is a fix for jansson which will preserve the order on which the json object was created. --- modules/jansson/jansson_utils.c.old 2015-07-06 17:41:25.852770681 +0800 +++ modules/jansson/jansson_utils.c 2015-07-06 17:42:27.384769579 +0800 @@ -33,7 +33,7 @@ val->flags = 0; if(json_is_object(v) || json_is_array(v)) { - const char* value = json_dumps(v, JSON_COMPACT); + const char* value = json_dumps(v, JSON_COMPACT|JSON_PRESERVE_ORDER); *freeme = (char*)value; val->rs.s = (char*)value; val->rs.len = strlen(value); Kelvin Chua

2 1

Re: [SR-Users] dispather
by Slava Bendersky 06 Jul '15

06 Jul '15

Hello Daniel, Thank you for reply, based on you example. Can I do some thing like that. modparam("dispatcher", "dstid_avp", "$avp(dsdst_id)") switch ($fd) { case "domain1.tld": $avp(dsdst_id)=1; break; case "domain2.tld": $avp(dsdst_id)=2; break; } ds_select_dst($avp(dsdst_id),4); Slava. From: "Daniel Tryba" <d.tryba(a)pocos.nl> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, July 3, 2015 6:18:03 AM Subject: Re: [SR-Users] dispather On Thursday 02 July 2015 09:12:41 Slava Bendersky wrote: > Is dispatcher module can do call routing based on domain ? > This only inbound traffic. > So if call come from @domain1.tld -----> asterisk box 1 @domain2.tld -----> > asterisk box 2. Sure, just make the dispatcher target dynamic with an avp: if($fd=="domain1.tld") { $avp(dispatcher_id)=1; } ... ds_select_dst($avp(dispatcher_id),4); _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] Kamailio multiple registration
by Fadi Hawari 06 Jul '15

06 Jul '15

Dear Sir , i need to register 10k users in sequance could you please let me know how to do it . Regards

5 4

modules.lst
by Joseph Zimmer 06 Jul '15

06 Jul '15

Hey guys, new to kamailio, keep getting stuck on step 7 - creating first database.Can you assist? see below: root@telephonepros:~# /usr/local/sbin/kamdbctl create /usr/local/etc/kamailio//kamctlrc: line 13: =/path/to/chrooted/directory: No such file or directory MySQL password for root: INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found Usage: grep [OPTION]... PATTERN [FILE]... Try 'grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database kamailio ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not found ERROR: Creating database kamailio failed! root@telephonepros:~#

2 1

Siremis 4.2 and Kamailio 4.3
by Alberto Sagredo 03 Jul '15

03 Jul '15

Hi Im installing Siremis 4.2 and getting this error when login. INstallation Wizard was fine and all dependencies are installed following Asipto Guide for Siremis. Any idea what could be wrong? Popup i get is.. {"target":"ERROR","content":" \n[2015-07-03 10:04:05 (GMT)] An exception occurred while executing this script: \nError message: #8192, Non-static method BizSystem::getConfiguration() should not be called statically, assuming $this from incompatible context<\/font> \nScript name and line number of error: \/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php:268<\/font> \n *function:<\/b> errorHandler ( 8192, \"Non-static method BizSystem::getConfiguration() should not be ca...\", \"\/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 268, Array(0) ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/sysheader.inc 117\nfunction:<\/b> userErrorHandler ( 8192, \"Non-static method BizSystem::getConfiguration() should not be ca...\", \"\/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 268, Array(0) ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php 268\nfunction:<\/b> setQuoteIdentifiers ( ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php 177\nfunction:<\/b> __construct ( Array(1) ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj.php 37\nfunction:<\/b> __construct ( Array(1) ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/ObjectFactory.php 166\nfunction:<\/b> constructObject ( \"system.do.UserDO\" ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/ObjectFactory.php 56\nfunction:<\/b> getObject ( \"system.do.UserDO\", 0 ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/BizSystem.php 252\nfunction:<\/b> getObject ( \"system.do.UserDO\" ) @ \/var\/www\/siremis-4.2.0\/siremis\/modules\/service\/authService.php 92\nfunction:<\/b> authDBUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.2.0\/siremis\/modules\/service\/authService.php 66\nfunction:<\/b> authenticateUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.2.0\/siremis\/modules\/user\/form\/LoginForm.php 92\nfunction:<\/b> Login ( ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/BizController.php 310\nfunction:<\/b> invoke ( ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/BizController.php 110\nfunction:<\/b> dispatchRequest ( ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/BizController.php 32\nfunction:<\/b> include_once ( \"\/var\/www\/siremis-4.2.0\/openbiz\/bin\/BizController.php\" ) @ \/var\/www\/siremis-4.2.0\/siremis\/bin\/controller.php 6<\/div>\n------------------------------Please ask system administrator for help...<\/div>\n"}]* *BR*

2 5

subst + insert_hf issue
by Igor Potjevlesch 03 Jul '15

03 Jul '15

Hello, I'm doing a subst on the From header for some particular cases, like this one: subst('/^(From|f):(.*)<sip:(.*)@/\1: "Anonymous" <sip:\3@/g'); In the same route (REQUEST_ROUTE called "INVITE"), I do the following: insert_hf("X-CarrierName: $avp(s:carrier_id)\r\n"); For many User-Agent, the X-CarrierName header is inserted after the @ into the user part of the From. So the From: looks like this: From:"Anonymous" <sip:0123456789@X-CarrierName: ABCDE\r\n The domain name part of the From is on the next line: <domain>:5060>;tag=3fdce0-a658bfa-13c4-50029-55941ccd-5eab9369-55941ccd\r\n I can't reproduce. The only potential explanation is that this impact the User-Agent that have the From at the top of the headers. If someone know what could going wrong here. Regards, Igor

2 3

Diameter sorce/destination port
by micunek＠gmail.com 03 Jul '15

03 Jul '15

Hi All, I am trying to use Cx (Diameter) interface via TCP towards our HSS according to Carsten Bock example. However kamailio client obtain TCP RST from HSS because a kamailio source port is different than port where kamailio is listen. I do not know if this is Diameter/Cx requirement or not. I did not find it in specifications. Is there any way how to ensure that kamailio use same port number for source and destination (3869)?. <?xml version="1.0" encoding="UTF-8"?> <DiameterPeer FQDN="...3gppnetwork.org" Realm="...3gppnetwork.org" Vendor_Id="10415" Product_Name="CDiameterPeer" AcceptUnknownPeers="1" DropUnknownOnDisconnect="1" Tc="30" Workers="4" QueueLength="8" TransactionTimeout="5" SessionsHashSize="128" DefaultAuthSessionTimeout="3600" MaxAuthSessionTimeout="3600" > <Peer FQDN="hss....3gppnetwork.org" Realm=".3gppnetwork.org" port="3868"/> <Acceptor port="3869" /> <Auth id="16777216" vendor="10415"/>  <Realm name="....3gppnetwork.org"> <Route FQDN="hss...3gppnetwork.org" metric="10"/> </Realm> <DefaultRoute FQDN="hss.....3gppnetwork.org" metric="10"/> </DiameterPeer> Thanks Peter

2 1

Segment fault
by Nick Raper 03 Jul '15

03 Jul '15

Hi, Can anyone help with a segment fault we had today please? Our Kamailio has been running for months with no problem but today we received this error 3 times and Kamailio stopped. Jul 1 12:48:43 sipa kernel: kamailio[2853]: segfault at 7f516819f6fc ip 00007f516a51c4a6 sp 00007fffd0fbe420 error 7 in pv.so[7f516a510000+36000] The full log files are attached for the three seg faults. Regards, [cid:image001.jpg@01CFEF9D.A1F537A0] Nick Raper Infrastructure Engineer 4th Dimension Technology | South Africa Office: +27 21 506 6260 | 24 hour helpline: +27 21 506 6260 | Fax: +27 86 610 5792 Email: nick(a)4dt.co.za<mailto:nick@4dt.co.za> | Web: www.4dt.co.za<http://www.4dt.co.za/> P Please consider the environment before printing this email

3 9

dispather
by Slava Bendersky 03 Jul '15

03 Jul '15

Hello Everyone, Is dispatcher module can do call routing based on domain ? This only inbound traffic. So if call come from @domain1.tld -----> asterisk box 1 @domain2.tld -----> asterisk box 2. Or which way is better to achieve it ? Thank you.

2 1

start-up crash if wdir is set and WITH_AUTH is defined (Kamailio 4.2)
by jack wang 02 Jul '15

02 Jul '15

Hello, If the 'wdir' (or 'workdir') is not set in kamctl.cfg or WITH_AUTH feature is not defined, kamailio can normally be started. Actually , the purpose is thatI want to add the 'wdir' to set the path coredump file... And ther error logs like following: (modified for hiding private info) ./kamailio[3565]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3565]: ERROR: <core> [pt.c:341]: fork_process(): init_child failed for process 2, pid 3565, "udp receiver child=0 sock=10.100.90.62:5060 (10.100.90.62:5060)" ./kamailio[3565]: CRITICAL: <core> [main.c:1621]: main_loop(): Cannot fork ./kamailio[3568]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3564]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3568]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3568]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3568]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3568]: ERROR: <core> [mi.c:128]: init_mi_child(): failed to init proc rpc for sip handling ./kamailio[3564]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3564]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3568]: CRITICAL: mi_fifo [mi_fifo.c:233]: fifo_process(): Failed to init the mi process ./kamailio[3564]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3564]: ERROR: <core> [pt.c:341]: fork_process(): init_child failed for process 1, pid 3564, "udp receiver child=0 sock=[2010:0:0:0:0:0:0:1]:5060" ./kamailio[3567]: CRITICAL: <core> [main.c:1693]: main_loop(): cannot fork timer process ./kamailio[3569]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3569]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3564]: CRITICAL: <core> [main.c:1621]: main_loop(): Cannot fork ./kamailio[3569]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3569]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3569]: ERROR: <core> [pt.c:341]: fork_process(): init_child failed for process 6, pid 3569, "ctl handler" ./kamailio[3569]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module ctl (/xxx/kamailio/lib/kamailio/modules/ctl.so) ./kamailio[3569]: ERROR: <core> [main.c:1707]: main_loop(): error in init_child ./kamailio[3563]: ALERT: <core> [main.c:781]: handle_sigs(): child process 3565 exited normally, status=255 ./kamailio[3563]: ALERT: <core> [main.c:781]: handle_sigs(): child process 3566 exited normally, status=255 ./kamailio[3572]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3572]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3572]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3572]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3572]: ERROR: <core> [pt.c:341]: fork_process(): init_child failed for process 9, pid 3572, "RTIMER SEC EXEC" ./kamailio[3572]: ERROR: rtimer [rtimer_mod.c:164]: child_init(): failed to start timer routine as process ./kamailio[3572]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module rtimer (/xxx/kamailio/lib/kamailio/modules/rtimer.so) ./kamailio[3572]: ERROR: <core> [main.c:1707]: main_loop(): error in init_child ./kamailio[3573]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3573]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3573]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3573]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3571]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3573]: ERROR: <core> [pt.c:479]: fork_tcp_process(): init_child failed for process 10, pid 3573, "tcp receiver (generic) child=0" ./kamailio[3571]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3573]: ERROR: <core> [tcp_main.c:4869]: tcp_init_children(): fork failed: No such file or directory ./kamailio[3571]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3571]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3571]: ERROR: <core> [pt.c:341]: fork_process(): init_child failed for process 8, pid 3571, "Dialog Clean Timer" ./kamailio[3571]: ERROR: dialog [dialog.c:742]: child_init(): failed to start clean timer routine as process ./kamailio[3571]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module dialog (/xxx/kamailio/lib/kamailio/modules/dialog.so) ./kamailio[3571]: ERROR: <core> [main.c:1707]: main_loop(): error in init_child Thanks in advance. :)

2 1

Problem Storing USRLOC Additional Attributes
by Luca Mularoni 02 Jul '15

02 Jul '15

Hi All, I'm using kamailio 4.2.5 (compiled from source and aligned up to commit [00a36ae07c587992d6486277ce2e9ae6c009685f]). In order to enrich users data collected during registering phase, I would like to store them in USRLOC Additional Attributes through the use of 'xavp_contact' structure, but I'm having problems storing them. Below relevant chunks of config: ... # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 0) modparam("usrloc", "xavp_contact", "ulvals") ... route[REGISTRAR] { ... # Just an example $xavp(ulvals=>A) = "A"; $xavp(ulvals[0]=>B) = "B"; $xavp(ulvals[0]=>C) = "C"; $xavp(ulvals[0]=>D) = "D"; ... if (!save("location")) sl_reply_error(); exit; } route[LOCATION] { ... if(!lookup("location") { # Handle Failure ... } # Print USRLOC Additional Attributes # Just an Example $var(i) = 0; $var(N) = $branch(count); while($var(i)<=$var(N)) { xlog("LOC - [$var(i)] - A [$xavp(ulvals[$var(i)]=>A)]\n"); xlog("LOC - [$var(i)] - B [$xavp(ulvals[$var(i)]=>B)]\n"); xlog("LOC - [$var(i)] - C [$xavp(ulvals[$var(i)]=>C)]\n"); xlog("LOC - [$var(i)] - D [$xavp(ulvals[$var(i)]=>D)]\n"); $var(i) = $var(i) + 1; } ... } And this is the relevant chunk of syslog when a local subscriber is found in 'location' table: ... LOC - [0] - A [A] LOC - [0] - B [<null>] LOC - [0] - C [<null>] LOC - [0] - D [D] ... Summing up, it seems that when xavp structure is stored in memory, just first and last elementsare inserted. In real config I'm using more than 4 attributes and behaviour is the same: just first and last are stored. I've tried also with modparam("usrloc", "db_mode", 0) but the problem remains. Where I'm wrong in using this structure? Or it could be a bug? Thank You in advance for your help Luca Mularoni

1 0

(no subject)
by Nelson Migliaro 01 Jul '15

01 Jul '15

Hello everybody, My SIP vendor request me to replace FROM before sending the traffic. In order to achieve this I use uac_replace_from. UAC module is setup in restore_mode = auto. In my insfrastructure I have an Asterisk and then a Kamailio that connects to vendor via internet. Softphone -> Asterisk -> Kamailio -> Internet -> SIP vendor If caller ID is setup in Asterisk using CALLERID(num)=34888888888 and then INVITE is forwarded to Kamailio, the call is established and finished correctly but the URI in TO field in BYE request from Kamailio to Asterisk contains garbage. In the scenario the callee hangs up the call. Example of TO Field with garbage: 34888888888 <sip:50026896@no{soy,ns^^> What I do see is that the number "50026896" that is part of the URI is the same I use in: uac_replace_from("50026896", "sip:50026896@sip.vendor.es"); Something else that I have found is that vsf field is the same in the INVITE and in the BYE. ---------------------------------------------------------------------------------------------------------------- 2015/06/23 17:48:38.552442 192.168.0.2:5060 -> 192.168.0.1:5060 BYE sip:34888888888@192.168.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bKfb49.73c6517609cdb6f7ec00b1f40a05dbe9.0 Via: SIP/2.0/UDP 8.8.8.8.8;rport=5060;branch=z9hG4bKfcdf.3767c59c2d0e3d8ab695669845ce4cea.0 branch=z9hG4bK04boo6104o5hcso0c2a1sd0000g00.1 Call-ID: 4bf8effb45b0ae8e049366297924cbba@192.168.0.1:5060 From: <sip:28999999999@192.168.0.2;tag=k0eci3x3-CC-30 To: 34888888888 <sip:50026896@no{soy,ns^^>;tag=as041b7d84 CSeq: 1 BYE Reason: Q.850;cause=16;text="normal call clearing" Max-Forwards: 67 Content-Length: 0 ------------------------------------------------------------------------------------------------------------------ DEBUG: uac [replace.c:525]: restore_uri(): getting 'vsf' Route param DEBUG: uac [replace.c:533]: restore_uri(): route param is 'AAAAAAEECQkCAgsNAXBeL0NGQUsfVl02Ni44Mw--' (len=40) DEBUG: uac [replace.c:607]: restore_uri(): decoded uris are: new=[sip:50026896@no{soy,ns#005#007] old=[sip:34888888888@8.8.8.8] DEBUG: uac [replace.c:525]: restore_uri(): getting 'vst' Route param DEBUG: uac [replace.c:533]: restore_uri(): route param is 'AAAAAAQPAw8MDgsAAHZBKRVdAhoVHQ4XH1BdYWJhbnRlLmVz' (len=48) DEBUG: uac [replace.c:607]: restore_uri(): decoded uris are: new=[ sip:28999999999@192.168.0.1] old=[sip:999999999@sip.vendor.es] -----------------------------------------------------------------------------------------------------------------------------------

2 1

Kamailio no recieved 480/486 messages
by Yuriy Gorlichenko 01 Jul '15

01 Jul '15

At my confiig I have route like this onreply_route[REPLY_FROM_WS] { if(status=~"[12][0-9][0-9]") { xlog("L_INFO", "Manage_Reply from webrtc client {$si:$sp} for method {$rm}: $rs"); rtpengine_manage("force trust-address replace-origin replace-session-connection DTLS=passive ICE=remove RTP/AVP"); route(NATMANAGE); } } This route successfully handling replies 180 183 and 200 and then (after natmanage route) resend to mediaserver. When callee client reject call it send 480 ot 486 message that not resended by kamailio nowhere. First of all I thought that 480 and 486 replies are failure replies but at log I see its at onreply route. So Ok . I added this to onreply_route if (status=~"48[06]"){ xlog("L_INFO", "Manage_Reply from webrtc client {$si:$sp} for method {$rm}: $rs"); rtpengine_manage("force trust-address replace-origin replace-session-connection DTLS=passive ICE=remove RTP/AVP"); route(NATMANAGE); } But message just going through natmanage route and goes nowhere My NATMANAGE route is xlog("L_INFO","NATMANAGE reply {$rm}"); if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } xlog("L_INFO","reply {$rm}"); } I see at my log only first NATMANAGE xlog... no more... thnks for answer

2 3

dialog flags vs dialog variables vs dialog profiles
by Muhammad Shahzad 01 Jul '15

01 Jul '15

Hi, Lasting night i was doing some tests and identified a strange behavior from dialog module. I am using Kamailio v4.3.0 rev. c6aa95. I basically wanted to identify if an incoming sequential ACK (from caller for an answered call) or a BYE (from callee who wants to hangup the call) belong to an existing dialog. First I tried to use dialog profiles as they were already enabled in my kamailio.cfg for every call. Calling the function "is_in_profile" after loose_route in WITHINDLG route, does not works for ACK, but works for BYE. Then I added dialog flags to initial INVITE (using function "dlg_setflag") and check them for ACK and BYE after loose_route (using "dlg_isflagset" method), i have the same behavior. They worked/matched only for BYE, not for ACK. Lastly I set dialog variables (e.g. dlg_var(my_call) = "1") in initial INVITE and checked for them for ACK and BYE requests after loose_route. Only they worked perfectly, i.e. i can match them for both incoming ACK and BYE requests. Is this correct and expected behavior? Why dialog profile and flags do not match for a dialog in unconfirmed state (i.e. ACK from caller after successful call answer)? They only work for confirmed dialogs (e.g. BYE from callee).On the other hand, dialog variables seems to work everywhere (i.e. all sequential requests). Thank you.

2 1

TLS certificate verification failed
by Austin Einter 01 Jul '15

01 Jul '15

Hi All I am having a sip client, I am using GNU tls for transport layer security. I am using Kamailio (TLS enabled) to test all the functionalities. I did registration over TLS. It was fine. I tried to make call. I was able to send INVITE. Kamaiio trying to connect to called party (which is again my sip endpoint). That time certificate verification failed. *Error log* * 8(14180) ERROR: tls [tls_server.c:1193]: tls_read_f(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed 8(14180) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading 8(14180) DEBUG: <core> [tcp_read.c:1481]: release_tcpconn(): releasing con 0x7f962f6b6db8, state -2, fd=5, id=3* How do I debug and fix this issue. Can somebody help me out. Thanks Austin

2 1

Media timeout using RTPProxy
by Sharath Kumar 01 Jul '15

01 Jul '15

Hello, Is there an API that can notify Kamailio of media inactivity in the RTPProxy? The documentation has something called "timeout_socket" . I don't understand how to use it though. Any help will be greatly appreciated. Thanks Sharath

2 1

How can I figure out the problem why call is not establish?
by zhangkai.gis 01 Jul '15

01 Jul '15

Hello. When I installed kamailio I add users then I can make a call.But When I install asterisk I found the asterisk take over the kamailio and I can register users.But When I make a call.It shows me can not establish the call.How can find the problem?Any logs?Or tools? Thanks, zhangkai

2 1

Fwd: websocket devices keepalive not working
by Yuriy Gorlichenko 01 Jul '15

01 Jul '15

As I read at README to Nathelper it may not work with usrloc table, Is it write? now I use NAT traversal module and Websocket module to keepalive mechanisms, that uses UDP or WS. with UDP it is wery simple method - OPTIONS replies. With WebSocket module keepalive mechanism uses thomething else. Not a SIP methods, I see at TCPDUMP packets that marks with modparam("websocket", "ping_application_data", "WebSockets rock") I need to handle replies of keepaliwe mechanism. How I may handle replies form this mechanism? 2015-06-30 18:37 GMT+03:00 Yuriy Gorlichenko <ovoshlook(a)gmail.com>: > thnks for response. I thing - nathelper is what I need. If I resolve issue > above (why it not work with setbflag(7) ) I thing this questions will be > closed. > > P.S may be it help - my kamailio version is > > version: kamailio 4.4.0-dev1 (x86_64/linux) 1dbd53 > flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, > DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, > F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, > USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > id: 1dbd53 > compiled on 06:24:22 Jun 23 2015 with gcc 4.8.4 > > 2015-06-30 18:18 GMT+03:00 Camille Oudot <camille.oudot(a)orange.com>: > >> Le Tue, 30 Jun 2015 15:49:26 +0300, >> Yuriy Gorlichenko <ovoshlook(a)gmail.com> a écrit : >> >> > 4) I tried to do it with tcpops >> > >> > Only for ws and tcp (WS is use TCP as i know) >> > tcp_keepalive_enable("15", "5", "15"); >> > >> > same result... >> >> Hi Yuriy, >> >> the tcp_keepalive_enable() function will send keepalive packets at TCP >> level, not at application level (SIP OPTIONS). It can be used to check >> if the remote peer connection is still valid, and close the TCP >> connection on the server side if it is no more valid. If you enable the >> TCP keepalives, you should see these packets using tcpdump or wireshark. >> >> Along with "handle_lost_tcp" option on usrloc, this can be used to >> maintain the usrloc table as consistent as possible. >> >> -- >> Camille >> > >

1 0

WWW-Authenticate header Query
by Anil Kumar 01 Jul '15

01 Jul '15

Hi, I am using Kamailio(4.2.5) as my SIP proxy also enabled authentication. Added two users xyz_0 and xyz_1 to the kamailio database. While Registration I am getting 401 Unauthorized as shown below SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.1.107:48601;branch=z9hG4bK858339335 From: <sip:xyz_0@192.168.1.107>;tag=251958751 To: <sip:xyz_0@192.168.1.107 <sip%3Axyz(a)192.168.1.107> >;tag=ca5a1db24813f1937965921e320b98e4.6232 Call-ID: 1808202998(a)192.168.1.107 CSeq: 2 REGISTER WWW-Authenticate: Digest realm="192.168.1.107", nonce="VZJ3hlWSdlr0HS/jGQljEKmY20ZrO0Qz" Server: kamailio (4.2.5 (x86_64/linux)) Content-Length: 0 The WWW-Authenticate header is not complete here, May I know the reason for this ? I am expecting algorithm, domain and other stuff in the WWW-Authenticate header. I am not able to proceed further , Please do needful. Thanks

2 1

REST API
by Marek Cervenka 01 Jul '15

01 Jul '15

hello, is there REST API for kamailio for basic functions like subscribers, dispatcher, blacklist/whitelist, ACL, dial plan table, ... ? thank you -- --------------------------------------- Marek Cervenka =======================================

2 1

websocket devices keepalive not working
by Yuriy Gorlichenko 30 Jun '15

30 Jun '15

I try to send keepalive requests (options) to clients at usrloc I tried 4 mechanisms 1) nathelper modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@mydomain") but it didn't send keepalives to noone client 2)nat_traversal I tried to trigger keepalives with nat_keepalive() when REGISTER coming to my service It works fine with UDP packets modparam("nat_traversal", "keepalive_interval", 15) modparam("nat_traversal", "keepalive_method", "OPTIONS") modparam("nat_traversal", "keepalive_from", "sip:pinger@mydomain") modparam("nat_traversal", "keepalive_state_file", "/var/run/keepalive_state") 3)So I also tried to do it with WebSocket keepalive mechanism, but it still not work for (off course I checked only WS devices) modparam("websocket", "keepalive_mechanism", 15) modparam("websocket", "keepalive_timeout", 15) modparam("websocket", "keepalive_interval", 15) 4) I tried to do it with tcpops Only for ws and tcp (WS is use TCP as i know) tcp_keepalive_enable("15", "5", "15"); same result... So I need mechanism to check all devices for keepalive. I think nathelper works with all protos (As I read at docs) Can somebody explain to me what I doing wrong?

3 3

REST API
by Marek Červenka 30 Jun '15

30 Jun '15

hello, is there REST API for kamailio for basic functions like subscribers, dispatcher, blacklist/whitelist, ACL, dial plan table, ... ? thank you -- --------------------------------------- Marek Cervenka =======================================

1 0

interconnect 2 Kamailio server
by kai.ohnacker＠cbc.de 30 Jun '15

30 Jun '15

Hello, after some trying with SEMS and Blox as SBC, I changed my idea of SIP infrastructure from Kamailio (private) -SBC in DMZ (Private & Public) - www (Public) to Kamailio (private) - Kamailio with rtpproxy in DMZ (Private & Public) - www (Public). Now, my questions are: How can I connect 2 Kamailio SIP server together? Which modules I need? Is there somewhere a good howto? Cheers, Kai Ohnacker

2 5

BYE from CANCLEd path gets forwarded to caller
by Daniel Tryba 30 Jun '15

30 Jun '15

I was testing a scenario whith parallel forking (multiple registers) where 2 endpoints answer near simultaniously. When phone that was just a bit slower hangs up, the BYE gets forwarded to the caller and thus the connection is terminated. This is as far as I know for stateful transactions (t_relay()) I always assumed in this scenario the state machine in kamailio would prevent this kind of behavior by itself. Is there a way to prevent these messages being forwarded?

2 2

Which Open Source IVR can be used with Kamailio and how?
by zhangkai.gis 30 Jun '15

30 Jun '15

Hello, Which Open Source IVR can be used with Kamailio and how? Thanks, zhangkai

2 8

invalid char in json text
by Kelvin Chua 30 Jun '15

30 Jun '15

Playing around with ndb_mongodb getting this error: ERROR: ndb_mongodb [mongodb_client.c:262]: mongodbc_exec_cmd(): Failed to run command: lexical error: invalid char in json text query is very simple: db.cdrs.insert({time:new Date(),src_user:"kelvin",term_ani:"14112",src_domain:"domain.com ",src_ip:"0",dst_ouser:"18188",dst_user:"18112",dst_domain:"domain.com"}) which is inside a pv mongodb_cmd("mgs1", "datab", "cdrs", "$var(MNG_MISSED)", "mgr1"); tried this: insert({time:new Date(),src_user:"kelvin",term_ani:"14112",src_domain:" domain.com",src_ip:"0",dst_ouser:"18188",dst_user:"18112",dst_domain:" domain.com"}) and other combinations, same error any ideas? i guess we need some more examples on the documentation Kelvin Chua

3 7

Siremis Register Page problem.
by Sven Juhkam 29 Jun '15

29 Jun '15

Hi! In SIREMIS I am getting the info after trying to register, that "Public registration is not enabled!" but i have set siremis/modules/ser/config/common.Main.php to $cfg_siremis_public_registrations = true; What is the problem? Thanks and Greetings, Sven

1 0

answer back from wrong IP / Interface
by Mihaly Zachar 27 Jun '15

27 Jun '15

Hi Gents, We have a redirect server based on Kamailio (4.2.3). The server has multiple IPs, but Kamailio is bound to only one address: http://view.xscreenshot.com/9dfdf449a387bdc6f9a719eae3620416 *Sometimes* it answers back from a wrong IP address, here is the proof: http://view.xscreenshot.com/ac661d6b303eda6653b37be7cf65aa65 The interesting thing is that if I do: root@tralab-route-01:~# ip ro get 10.121.24.51 10.121.24.51 via 10.121.43.30 dev vlan486 src 10.121.43.17 cache ipid 0x05ad root@tralab-route-01:~# So the 10.121.43.17 IP is selected, but I *do* expect that Kamailio send back the answer from the same socket where it got the request. What could be the issue ? Why does it do that *sometimes* ? In the majority of the time it works well.. I am not sure why this happens... Thanks, Mitya

1 1

don't fragment flag in rtpengine request
by Juha Heinanen 27 Jun '15

27 Jun '15

i was wondering why sometimes requests to rtpengine don't get through and icmp unreachable destination error message is received. it turned out that in these error cases ip packet containing the rtpengine request has 0x02 (don't fragment) flag set even when the packet was bigger than 4 KB. in this case kamailio was running in an amazon container which may be doing something weird. is it somehow possible from the application (= kamailio) to control setting of that flag? -- juha

1 0

deb building system
by Victor Seva 26 Jun '15

26 Jun '15

Hello, I would like to inform that the deb nightly builds are going to be disabled. I'm having problems with the actual configuration and until I found the time to fix them the service will be out of order. No new debs will be produced until further notice. Cheers, Victor

2 1

limit calls to incoming only?
by Peter Villeneuve 26 Jun '15

26 Jun '15

Hi, Is it possible for Kamailio to restrict certain users (like in a group) to only receive calls but never dial out or send SIP Simple messages outbound, while allowing all other "regular" users to make outgoing calls and text messages as well as receive them? If so, which module do you guys recommend I look into? Cheers, Peter

2 3

fix_nated_contact NAT rtpproxy problems
by Vik Killa 25 Jun '15

25 Jun '15

Hello, We are trying to get Kamailio working as an SBC using rtpproxy(); The config seems to work except with some carriers, kamailio does not recognize the ACK because it does not match the contact header of the OK I've been trying to figure out how to get kamailio to send the OK back using the public IP address but I cannot figure out how. I've tried using various deployments of the nathelper module such as fix_nated_contact() set_contact_alias() No matter what though, Kamailio will not change the contact header it receives from FS before sending it back to the carriers. This has become frustrating as I read tutorials that showed this is possible, yet I am unable. Thank you. /V

1 0

"00" prefix record problem on ACC
by Yasin CANER 25 Jun '15

25 Jun '15

3 2

linking math.h while compiling a module
by Shouvanik Chakrabarti 25 Jun '15

25 Jun '15

Hello, I have developed a Kamailio module to handle group messaging and am using certain functions from the libm.so library. Compilation happens fine, but Kamailio subsequently fails to start because the math.h symbols are unlinked/undefined. Could someone please suggest a workaround for this. Thanks, Shouvanik.

2 4

How to disable retransmits via TCP connection? etc.
by Andrey Utkin 25 Jun '15

25 Jun '15

Hi! We meet such an issue with some kinds of client internet connections (e.g. reproduces always on local wifi, but doesn't repr. on 3g inet). We don't know what are exact network characteristics, but what happens from perspective of traffic on SIP server is the following: 1. the SIP client registers successfully to Kamailio (which is recent git master HEAD) with TCP transport for SIP; 2. another client calls this user; 3. Kamailio relays INVITE, it gets transmitted fully and ACK-ed on TCP level; 4. then, within 0.1-0.2 second, Kamailio fires first retransmit of that INVITE, which doesn't get fully transmitted (it is shown trunkated in sniffer output, no TCP ACK replies for it); 5. Kamailio tries to retransmit INVITE again, with the same result as in #5. 6. The caller gets SIP/2.0 408 Request Timeout. A piece of traffic from "ngrep -t -e -d any -W byline port 5060": https://gist.githubusercontent.com/krieger-od/219f9975e5efb980ff5b/raw/096a… The called side app is based on mobile Linphone app. Switching to UDP is not an option (in some networks SIP messages get delivered trunkated which breaks calls), we will check how it works with TLS transport a bit later when there's technical possibility. I have two questions: 1. How would I completely disable retransmissions to TCP connections? 2. Any ideas what can be the reason for this issue? Retransmissions by themselves? -- Andrey Utkin

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Kamailio Server Maintenance - Wed-Thu Night
by Daniel-Constantin Mierla 25 Jun '15

25 Jun '15

Hello, during the night between Wed (June 24) and Thu (June 25), planned to start not early than 00:00 GMT+1, there will be some schedule maintenance work to the infrastructure that is hosting some of the kamailio.org servers. The main affected services will be: - main website (www.kamailio.org) - wiki portals - mailing lists (lists.sip-router.org) - git mirror (git.kamailio.org) It is expected to have short downtimes of few minutes. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

Kamailio + RTPEngine + Asterisk
by Javier Aristizábal 24 Jun '15

24 Jun '15

Hello folks, I am trying to configure this on the same server: Kamailio v4.2 port 5060 RTPEngine v2.3 port 22222 Asterisk v11.18.0 port 5080 I am follow the Asipto tutorials, for the Kamailio and the real time configuration. Now, I can talk between extension and PSTN without issues. But the trouble is with the Status registration with Asterisk. And I need that Status OK because I am using queues in asterisk for call centre. I got this from with: sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 101 (Unspecified) D Yes Yes 0 UNREACHABLE Cached RT the status is UNREACHABLE and the asterisk table sipregs won't update after each registration. To be sure, I tried first all config only with asterisk to check that is not a problem with MySQL or something. On my cfg file I have this for the asterisk registration: route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } Any clue? Thanks! -- Javier Aristizábal

2 16

Kamailio proxy at Amazon EC2 (behind NAT)
by Alexandru Covalschi 24 Jun '15

24 Jun '15

Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP. In case of WebRTC I get lot's of erros: Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) The call reaches Asterisk, but not vice-versa. No media is being transferred. Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 15

Kannot start kamailio 4.3
by Yuriy Gorlichenko 23 Jun '15

23 Jun '15

Try to start kamailio on Ubuntu 14.04.02 Get this errors ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: ctl [ctl.c:273]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: <core> [sr_module.c:945]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) my params for ctl # ----- ctl params ----- modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") version: kamailio 4.4.0-dev1 (x86_64/linux) 1dbd53 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 1dbd53 compiled on 06:24:22 Jun 23 2015 with gcc 4.8.4

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