- sr-users - kamailio.org

Kamailio and openLDAP integration.
by Safdar Khan 22 Jan '16

22 Jan '16

Hi guys, I have tried to integrate kamailio and ldap.It seems like both are taking to each other. Here the purpose of ldap is for back-end support for sip user authentication rather than kamailio database.However i am able to log in but could not make call between sip users.I could not figure out what i am missing. Here is the routing block for ldap in my kamailio.cfg > request_route { > route(LDAPAUTH); > > } --- route[LDAPAUTH] > { > if(is_method("REGISTER")) > { > if(is_present_hf("Authorization")) > { > if > (!ldap_search("ldap://sipaccounts/ou=MyUsers,dc=example,dc=com?cn,userPassword?one?(cn=$fU)")) > { > switch ($retcode) > { > case -1: > sl_send_reply("404", "User Not Found"); > exit; > case -2: > sl_send_reply("500", "Internal server error"); > exit; > default: > exit; > } > } > ldap_result("cn/$avp(username)"); > ldap_result("userPassword/$avp(password)"); > route(REGISTRAR); > if (!pv_www_authenticate("$td", "$avp(password)", "4")) > { > www_challenge("$td", "1"); > exit; > } > sl_send_reply("200", "ok"); > exit; > } > else > { > www_challenge("$td", "1"); > exit; > } > } > }

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SIP Trunking using Kamailio IMS
by indranil chowdhury 21 Jan '16

21 Jan '16

Hello, Is SIP trunking possible with the free version of Kamailio or we need to upgrade to business license? Kindly get back to us with the details of using SIP trunking using Kamailio. We have a requirement in our organisation regarding the same which uses IMS and SIP trunking. In this setup we would initially like to test with 2 LTE EPCs connected to two Kamailio IMS with a SIP trunk between them. Thanks and Regards, Indranil

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topoh quirk with 200 OK for CANCEL
by Alex Balashov 21 Jan '16

21 Jan '16

Hi, I'm using topoh and am running into lots of these errors: Jan 19 20:34:52 csrpus1 /usr/local/sbin/kamailio[3641]: ERROR: [R-DEFAULT-STATELESS-REPLY:nmhfrms2ib9Cnm9kS00Qtbtot5PMiQ0Ni1QKS5iKibuotel14Ygc] !> Received stateless reply 200 (CANCEL) from xxx.xxx.xxx.xxx:5060 Jan 19 20:34:53 csrpus1 /usr/local/sbin/kamailio[3639]: ERROR: tm [t_lookup.c:932]: t_reply_matching(): matching transaction found but callids don't match (received: nmhfrms2ib9Cnm9kS00Qtbtot5PMiQ0Ni1QKS5iKibuotel14Ygc stored: 51899BC6-569EE4330005CE1B-7D47C700_bleg) The call topology is: UAC --> Kamailio (+ topoh) --> PSTN GW Topoh is behaving precisely as expected, but it looks like we might have hit a TM case that topoh does not accommodate somehow: 1. A CANCEL comes in from the UAC and Kamailio sends a CANCEL for its branch to the PSTN GW. 2. Kamailio sends a '200 cancelling' for the UAC's branch backward. 3. The PSTN GW replies with a 200 OK for the CANCEL (PSTN GW branch) that contains the topoh-spoofed Call-ID, but for some reason Kamailio does not match this reply, as indicated by the above error. 4. Because the 200 OK reply was stateless and not matched to a CANCEL transaction, this causes Kamailio to retransmit the CANCEL to the PSTN GW, at which point it receives: 481 Call Leg/Transaction Does Not Exist 5. #4 repeats once more. I don't see any real harm from this situation, but it seems like something that needs fixing. The 487 Request Terminated final reply is processed and relayed correctly to the UAC, and the dialog ends appropriately everywhere. FWIW, I thought this might have to do with async processing of the initial INVITE (through mqueue + rtimer), so I turned it off, but it didn't make any difference. My CANCEL handling in the main request route looks like this: if(is_method("CANCEL")) { set_rtpengine_set("1"); rtpengine_delete(); if(!t_relay_cancel()) { sl_send_reply("500", "Internal Server Error"); exit; } xlog("L_ERR", "[R-MAIN:$ci] !> " "Corresponding INVITE transaction for CANCEL " "does not (or, no longer) exists\n"); exit; } So, it just looks like TM is not matching, and therefore not absorbing the 200 OK reply to its CANCEL to the upstream gateway -- when topoh is enabled. Any insights would be appreciated! -- 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/

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RFC: removing (disabling) fork=no
by Daniel-Constantin Mierla 21 Jan '16

21 Jan '16

Hello, wondering if anyone is using fork=no -- some old docs suggest it is suitable for debugging, but actually kamailio doesn't work properly in this mode, leading to more troubles than benefits (e.g., having reports of invalid issues, like tcp not working in this mode). In first phase I would disable setting this value, with a warning if set to no, because most of the configs out there have fork=yes. Removing it could be considered in the future. Note that this fork=no is different than don't daemonize controlled with -D, which will stay being useful for some init.d systems. Comments or other suggestions? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

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Re: [SR-Users] Fwd: Kamailio and NAT
by Nelson Migliaro 20 Jan '16

20 Jan '16

I am thinking that I cat get the port during the first call and then use that port for the rest of calls. Maybe that first call will fail but after that, all calls will go fine. Example of a Trying showing received port: 52548 ¿Do you know what pseudovariable represent that value? I tested with all variables I could find but I could not find anyone. Thank you --------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/01/19 17:38:38.981987 VENDOR-IP:5060 -> KAMAILIO-IP:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP PUBLIC-IP:52548;branch=z9hG4bK8b5c.b122371ac0ac2f3ef3204b0f192cb16c.1;rport=52548 Via: SIP/2.0/UDP ASTERISK-IP:5060;received=ASTERISK-IP;branch=z9hG4bK4082d124;rport=5060 From: 888888888 <sip:888888888@ASTERISK-IP>;tag=as34f971fe To: <sip:999999999@VENDOR-DOMAIN> Call-ID: 0b5307fa290674a97b47970643fce42a@ASTERISK-IP:5060 CSeq: 102 INVITE Server: kamailio Content-Length: 0 2016-01-15 9:50 GMT+01:00 Nelson Migliaro <eng.migliaro(a)gmail.com>: > Hello Daniel, > > Yes, I am registered to the vendor. > > Regards, > > Nelson.- > > 2016-01-15 7:58 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Ahh, I thought Asterisk is in the public internet, but actually you >> connect to a provider (vendor), which seems to run Kamailio as well. >> >> Using information from 100 trying is too late, as the INVITE was already >> sent... so one more question before trying to propose a solution. Do you >> have to register to the provider? >> >> Cheers, >> Daniel >> >> >> On 14/01/16 18:51, Nelson Migliaro wrote: >> >> Yes it is possible, but is there an easy way to workaround the issue >> using Kamailio. >> >> Because I have the port because vendor is sending that info in Trying: >> >> 2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060 >> SIP/2.0 100 trying -- your call is important to us >> Via: SIP/2.0/UDP PUBLIC-IP:52548;branch= >> z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=*52548* >> >> I need to find the way to discover the port used by firewall (maybe >> getting that info from Trying) and then advertise that port. >> >> >> >> 2016-01-14 18:32 GMT+01:00 Daniel-Constantin Mierla < <miconda(a)gmail.com> >> miconda(a)gmail.com>: >> >>> Not really up to date with all Asterisk features -- do you know if you >>> can append a custom header to a SIP response that is going to be generated >>> by Asterisk? Eventually the reply for an OPTIONS request. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 14/01/16 17:19, Nelson Migliaro wrote: >>> >>> Yes, I manage all devices, even the internet router but it does not >>> allow static pat. >>> >>> 2016-01-14 16:07 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>> >>>> Do you control the Asterisk? If yes, depending on Asterisk capabilities >>>> of building replies, you may be able to do some automation to detect the >>>> external port. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro < >>>> <eng.migliaro(a)gmail.com>eng.migliaro(a)gmail.com> wrote: >>>> >>>>> There is not a public Kamailio, only one Kamailio behind NAT, >>>>> >>>>> Right now the configuration is: >>>>> >>>>> Asterisk <-> Kamailio (Private IP + advertise public IP + RTP Proxy ) >>>>> <-> Internet router (public IP + symmetric na) <-> Internet >>>>> >>>>> Regards, >>>>> >>>>> 2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla < >>>>> <miconda(a)gmail.com>miconda(a)gmail.com>: >>>>> >>>>>> Is the kamailio behind nat communicating with another kamailio on a >>>>>> public IP? >>>>>> >>>>>> Cheers, >>>>>> DAniel >>>>>> >>>>>> On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro < >>>>>> <eng.migliaro(a)gmail.com>eng.migliaro(a)gmail.com> wrote: >>>>>> >>>>>>> Thank you Daniel for your answer, >>>>>>> >>>>>>> As you mention, there is a symmetric nat and router does not allow a >>>>>>> static NAT. >>>>>>> >>>>>>> By sniffing traffic I can see the port is using new but in case it >>>>>>> change, how can automate the process of advertising the correct port? >>>>>>> >>>>>>> Cheers! >>>>>>> >>>>>>> >>>>>>> ---------- Forwarded message ---------- >>>>>>> From: Daniel-Constantin Mierla < <miconda(a)gmail.com> >>>>>>> miconda(a)gmail.com> >>>>>>> Date: 2016-01-13 23:28 GMT+01:00 >>>>>>> Subject: Re: [SR-Users] Kamailio and NAT >>>>>>> To: "Kamailio (SER) - Users Mailing List" < >>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org> >>>>>>> >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> it looks like you have a symmetric nat router, so the allocated port >>>>>>> is randomly selected. >>>>>>> >>>>>>> If you don't control the nat router to set a static forwarding rule >>>>>>> or it doesn't provide the option to set static forwarding, then you are >>>>>>> pretty much left with sniffing the traffic to discover the external port >>>>>>> and advertise it. >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On 13/01/16 20:31, Nelson Migliaro wrote: >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> I finally were able to run my Kamailio behind NAT but in order to >>>>>>> accomplish that I included: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 >>>>>>> >>>>>>> 52548 is the port my internet router change when doing NAT >>>>>>> (5060->52548). I found this port sniffing traffic >>>>>>> >>>>>>> Conclusions at this point are: >>>>>>> >>>>>>> >>>>>>> ---------------------------------------------1-------------------------------------------------------------------------------------------------- >>>>>>> If I use this line: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work >>>>>>> :( >>>>>>> >>>>>>> When I dial a call, INVITE / ACK / Trying / OK goes fine because >>>>>>> they are part of the same transaction >>>>>>> When remote party disconnects the call, BYE goes to PUBLIC-IP port >>>>>>> 5060 and router blocks de request. I assume vendor sends BYE to 5060 >>>>>>> because it is a new transaction >>>>>>> >>>>>>> -----------------------------------------------2-------------------------------------------------------------------------------------------------- >>>>>>> >>>>>>> If I use this line: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!! >>>>>>> >>>>>>> When I dial a call, INVITE / ACK / Trying / OK goes fine because >>>>>>> they are part of the same transaction >>>>>>> When remote party disconnects the call, BYE goes to PUBLIC-IP port >>>>>>> 52548 and router forward the request to Kamailio. Since there is an open >>>>>>> connection. >>>>>>> >>>>>>> I need to find the way to find the way to advertise the public port >>>>>>> internet router is doing NAT (PAT). >>>>>>> >>>>>>> >>>>>>> --------------------------------------------------------------------------------------------------------------------------------------------------- >>>>>>> This trace is a call that worked fine because I included line: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 >>>>>>> >>>>>>> >>>>>>> This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060 >>>>>>> advertise PUBLIC-IP:52548 >>>>>>> 2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060 >>>>>>> INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0 >>>>>>> Record-Route: < >>>>>>> sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB >>>>>>> A-;nat=yes> >>>>>>> Via: SIP/2.0/UDP >>>>>>> PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0 >>>>>>> Via: SIP/2.0/UDP >>>>>>> PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060 >>>>>>> Max-Forwards: 69 >>>>>>> From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453 >>>>>>> To: <sip:NUM-DESTINATION@sip.VENDOR-IP> >>>>>>> Contact: >>>>>>> <sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1> >>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060 >>>>>>> CSeq: 102 INVITE >>>>>>> User-Agent: Kamailio >>>>>>> Date: Wed, 13 Jan 2016 19:10:15 GMT >>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >>>>>>> INFO, PUBLISH, MESSAGE >>>>>>> Supported: replaces, timer >>>>>>> Content-Type: application/sdp >>>>>>> Content-Length: 255 >>>>>>> >>>>>>> >>>>>>> Trying..... >>>>>>> >>>>>>> 2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060 >>>>>>> SIP/2.0 100 trying -- your call is important to us >>>>>>> Via: SIP/2.0/UDP >>>>>>> PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548 >>>>>>> Via: SIP/2.0/UDP >>>>>>> PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060 >>>>>>> From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453 >>>>>>> To: <sip:NUM-DESTINATION@VENDOR-IP> >>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060 >>>>>>> CSeq: 102 INVITE >>>>>>> Server: kamailio >>>>>>> Content-Length: 0 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> And finally a BYE >>>>>>> >>>>>>> 2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060 >>>>>>> BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1 >>>>>>> SIP/2.0 >>>>>>> Via: SIP/2.0/UDP >>>>>>> VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0 >>>>>>> From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93 >>>>>>> To: "NUM-SOURCE" <sip:NUM-SOURCE@VENDOR-IP>;tag=as3b72a453 >>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060 >>>>>>> CSeq: 28731 BYE >>>>>>> Max-Forwards: 69 >>>>>>> Route: < >>>>>>> sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na >>>>>>> yes> >>>>>>> Reason: Q.850;cause=16 >>>>>>> Content-Length: 0 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------- >>>>>>> >>>>>>> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------- >>>>>>> >>>>>>> Finally, It is finally working because I hardcoded NAT´d port. >>>>>>> I would like to find a way to avoid setting the port in "hard". >>>>>>> >>>>>>> Thank you >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>>> list >>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org >>>>>>> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>>> list >>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org >>>>>>> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierla - <http://www.asipto.com> >>>>>> http://www.asipto.com >>>>>> <http://twitter.com/#%21/miconda>http://twitter.com/#!/miconda - >>>>>> <http://www.linkedin.com/in/miconda>http://www.linkedin.com/in/micond >>>>>> >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>> list >>>>>> sr-users(a)lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>> sr-users(a)lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>> >>>>> >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla - <http://www.asipto.com>http://www.asipto.com >>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond >>>> <http://www.linkedin.com/in/miconda> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu >>> >>> >>> _______________________________________________ >>> 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 Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu >> >> >

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Registration to sip provider
by Evren Akbulut 19 Jan '16

19 Jan '16

Hi, I am a newbie. As I read I understand I need to use UAC module for registration but cannot figure how to trigger the registration request. I enable the UAC module and it seem to be working. How can I trigger the REGISTER to a SIP provider account? And is it possible to use multiple accounts for different provider registrations? Can someone help about UAC usage please. Regards Evren Akbulut

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rtpengine to do DTLS-SRTP handshake, but media should not pass through it. the media should go through media server, which is running on the same machine
by riko nir 19 Jan '16

19 Jan '16

Hi, We have a media server and (opensips + rtpengine). How can I configure opensips+rtpengine to just do DTLS handshake with the client, but the media should not go through RTPEngine. The media should go to the Media server. Thanks. Riko

1 0

Kamailio Load Balancing Cluster
by sandeep goje 19 Jan '16

19 Jan '16

Hi, I am using Kamailio as a load balancer with call load distribution to Freeswitch Gateways. I need a HA cluster of Kamailio. I have seen instructions to setup Kamailio HA cluster using KeepAlive but I also want them to share the load balancing information. I have seen OpenSIPS supports this. Is there a way in Kamailio to make both load balancers also share the load balancing information. Thank you. Regards, Sandeep The most profound statements are often said in silence. -Lynn Johnston

2 1

Get value of RPID privacy param
by Grant Bagdasarian 18 Jan '16

18 Jan '16

Hello, How do I select the value of the privacy parameter in the Remote-Party-ID header? Remote-Party-ID: "John Doe" <sip:jdoe@foo.com>;party=calling;id;privacy=full;screen=yes The following doesn't work, and I believe because it's not a valid select class: $sel((a)hf_value.remote_party_id.privacy) I'm using kamailio 4.1. Regards, Grant

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IMS terminating dialog does not re-run through all iFCs after calling t_newtran()
by Russell McConnachie 17 Jan '16

17 Jan '16

Hi, My CSCF environment has an application server acting as a B2BUA. It's originating a new session, and creating a new terminating session back to the S-CSCF (two separate sessions per-call). When I perform a re-INVITE with a diversion to transfer the call to another URI it is all good until the S-CSCF attempts to call the call control logic: # Start new transaction: t_newtran(); if (isc_match_filter("orig", "location")) { t_on_failure("ISC_ORIG_FAILURE"); exit; The t_newtran() method from what I understand will create a new transaction. Inside of this new transaction, I would think that the isc_match_filter should re-run through all of the iFCs defined on the HSS service profile, but it does not seem to be doing so. It seems to be continuing at where the original transaction ISC mark skip point was set. Thanks

2 1

Different rtpproxy for different media type
by Koray Vatansever 15 Jan '16

15 Jan '16

Hi everybody, Is there a way to use different rtpproxies for different media types? I want to use one rtpproxy set for audio, and use another one for video. Is this possible in kamailio? Regards, Koray

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Kamailio replies with SIP/2.0 500 Service Unavailable Under load
by sandeep goje 15 Jan '16

15 Jan '16

Hi, I am doing load testing with of Kamailio. I have sipp (version 3.3) as the load generator,Kamailio(4.2.7) as Load Balancer and Freeswitch (1.6.5) as Gateways. I am using the following command to generate load ./sipp -sf uac.xml -d 2000 -m 4000 -l 1000 -r 200 -trace_stat -i SIPP_IP -p 15060 --trace_err -trace_error_codes -trace_calldebug -trace_screen KAM_IP When the rate of generation is below 80 (-r 80), Kamailio behaves fine.But if we go beyond 80, Kamailio starts sending "500 Service Unavailable" 2016-01-12 11:18:15.375103 1452577695.375103: Aborting call on unexpected message for Call-Id '96-32493@SIPP_IP': while expecting '180' (index 2), received 'SIP/2.0 500 Service Unavailable^M Via: SIP/2.0/UDP SIPP_IP:15060;branch=z9hG4bK-32493-96-0^M From: sipp <sip:sipp@SIPP_IP:15060>;tag=96^M To: sut <sip:service@KAM_IP :5060>;tag=55f576f507e822fa6633cf4bc22740e6-6660^M Call-ID: 96-32493@SIPP_IP^M CSeq: 1 INVITE^M Server: kamailio (4.2.7 (x86_64/linux))^M Content-Length: 0^M With the above command, sipp output is ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2016-01-12 11:26:07.723764 1452578167.723764 Last Reset Time | 2016-01-12 11:27:16.280901 1452578236.280901 Current Time | 2016-01-12 11:27:16.282737 1452578236.282737 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:001000 | 00:01:08:558000 Call Rate | 0.000 cps | 58.345 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 0 | 4000 Total Call created | | 4000 Current Call | 0 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 1192 Failed call | 0 | 2808 -------------------------+---------------------------+-------------------------- Response Time 1 | 00:00:00:000000 | 00:00:15:102000 Call Length | 00:00:00:000000 | 00:00:10:614000 ------------------------------ Test Terminated -------------------------------- Kamailio is run with command /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 8 -u kamailio -g kamailio version: kamailio 4.2.7 (x86_64/linux) 727746 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, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 727746 compiled on 15:59:03 Jan 11 2016 with gcc 4.6.3 System : Ubuntu 12.04.4 LTS Linux devops10-60-20-169 3.11.0-15-generic #25~precise1-Ubuntu SMP Thu Jan 30 17:39:31 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux I am not seeing any errors in Kamailio Logs. Any pointers on how to get it to serve more calls/sec Regards, Sandeep The most profound statements are often said in silence. -Lynn Johnston

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Re: [SR-Users] Parser error when enabling siptrace with HEP functionality
by Sergey Basov 15 Jan '16

15 Jan '16

And one more thing. In captured packet from Homer, I see original SIP packets, without topoh modifications. But in dump from network interface by wireshark i see that packets was modified... -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com SATB-RIPE SATB-UANIC tel: (+38067) 403-62-54

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rtpengine graphite statistics server
by Juha Heinanen 15 Jan '16

15 Jan '16

rtpengine README.md mentions "graphite statistics server". is that server openly available somewhere? -- juha

3 3

Kamailio and NAT
by Nelson Migliaro 15 Jan '16

15 Jan '16

Hello, I am running Kamailio behind NAT. Kanailio has a private IP and I am relaying NAT to internet router. I am using: - #!define WITH_NAT - listen=udp:PRIVATE-IP:5060 advertise PUBLIC-IP:5060 - Patched RTP proxy including the advertise option And everything goes fine. I can make calls and have two way audio. The problem begins when the callee ends the call. BYE is not received in Kamailio (caller) I included the public IP using "add_contact_alias" because "set_contact_alias" was not adding the public IP. I included this in in NATDETECT (pre loaded router) if(is_first_hop()) { xlog("L_NOTICE","Metodo: $rm \n"); xlog("L_NOTICE","is first hop\n"); #set_contact_alias(); if (!add_contact_alias("PUBLIC-IP", "$Rp", "udp")) { xlog("L_ERR", "Error in aliasing contact $ct\n"); send_reply("400", "Bad request"); exit; } } I think the problem is related to destination that BYE is sent by the vendor. From what I see IP and port is taken from advertised in contact (PUBLIC-IP and 5060). The problem is that internet router changes the source port. Contact: <sip:999999999@PRIVATE-IP:5060;alias=PUBLIC-IP~5060~1> --- Is it correcto to add_contact_alias("PUBLIC-IP", "$Rp", "udp") in order to received new transactions or should I follow a different procedure??? Thank you

4 14

dispatcher state x and the return of the probing mode
by Sven Neuhaus 14 Jan '16

14 Jan '16

Hello, when trying to set a destination to inactive using dispatcher.set_state after setting the state to "i", when I do a "dispatcher.list" I see the state as "ix". The "x" state appears not to be documented. After a short time (seconds) the flags return to state "ip" (probing). According to the module documentation I should be able to set the "probing" mode. 1. Why doesn't probing mode remain disabled? 2. What is state "x"? Cheers, -Sven

3 4

using re.subst with an $avp or $var
by Vik Killa 13 Jan '16

13 Jan '16

Hello, Is it possible to use $avp() or $var() inside re.subst? Example: $var(PrefixMatch) = "00"; $var(destnumber1) = $(ru{re.subst,/^sip:$var(PrefixMatch)(.*)@(.*)/\1/}); $var(destnumber2) = $(ru{re.subst,/^sip:00(.*)@(.*)/\1/}); xlog("L_INFO", "destnumber1 $var(destnumber1)\n"); xlog("L_INFO", "destnumber2 $var(destnumber2)\n"); In the above, destnumber2 works, yet destnumber1 does not. Thanks! /V

3 13

RTP Engine multiplexing and De-multiplexing
by riko nir 13 Jan '16

13 Jan '16

Hello, A call from a remote webrtc client is coming to (opensips+rtpengine). The media streams from the webrtc client is multiplexed. Can I use rtpengine to demultiplex the multiplexed streams and send it to other end as de-multiplexed SRTP traffic . This is because, the other end handle the srtp traffic only in demultiplxed manner. Also, the streams from the remote end is a separate SRTP and SRTCP traffic and the rtpengine has to add required ICE information to make the streams to go in a single port and it has to multiplex the streams and send it to webrtc client. It is mentioned that rtpengine can support multiplex/demultiplex. So, how can I configure rtpengine for this purpose from opensips? Your suggestions would be very helpful for us. Thanks. Riko

2 1

Regarding rtpengine dtls handling and fetching crypto keys
by riko nir 13 Jan '16

13 Jan '16

Hi all, I have a media server and it is able to handle SRTP, provided the crypto key. We are planning to give webrtc support to the media server. We are using opensips+rtpengine for that. For dtls, we are using rtpengine. The rtpengine just needs to do the dtls handshake and it needs to fetch the crypto key and it should provide the key to media server, so that the media server will be able to handle the SRTP traffic. We are bit struggling, how to get the crypto keys from rtpengine and send it to opensips and opensips will send it to media server. how this can be done? your suggestions will help us a lot. Thanks, Riko

2 4

siptrace retransmission's
by Grant Bagdasarian 12 Jan '16

12 Jan '16

Hello All, I'm load testing Kamailio with siptrace (duplicate) enabled, but for some reason not all SIP messages are arriving at the sip capture server (kamailio instance with sipcapture). The load is an average of 50 CPS. I enabled debug 9 and I get the following messages. <core> [mem/q_malloc.c:369]: qm_malloc(): qm_malloc(0x7fefde4da010, 1274) called from siptrace: siptrace.c: trace_send_hep_duplicate(1615) <core> [mem/q_malloc.c:439]: qm_free(): qm_free(0x7fefde4da010, 0x7fefde618a88), called from siptrace: siptrace.c: trace_send_hep_duplicate(1659) <core> [mem/q_malloc.c:439]: qm_free(): qm_free(0x7fefde4da010, 0x7fefde61ee18), called from siptrace: siptrace.c: trace_send_hep_duplicate(1660) <core> [mem/q_malloc.c:474]: qm_free(): qm_free: freeing frag. 0x7fefde61ede8 alloc'ed from siptrace: siptrace.c: trace_send_hep_duplicate(1615) <core> [mem/q_malloc.c:369]: qm_malloc(): qm_malloc(0x7fefde4da010, 1271) called from siptrace: siptrace.c: trace_send_hep_duplicate(1615) <core> [mem/q_malloc.c:439]: qm_free(): qm_free(0x7fefde4da010, 0x7fefde61b488), called from siptrace: siptrace.c: trace_send_hep_duplicate(1659) <core> [mem/q_malloc.c:439]: qm_free(): qm_free(0x7fefde4da010, 0x7fefde61e880), called from siptrace: siptrace.c: trace_send_hep_duplicate(1660) <core> [mem/q_malloc.c:474]: qm_free(): qm_free: freeing frag. 0x7fefde61e850 alloc'ed from siptrace: siptrace.c: trace_send_hep_duplicate(1615) DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission DEBUG: siptrace [siptrace.c:1040]: trace_onreq_out(): retransmission In siptrace.c I see the following line of code: if(ps->flags&TMCB_RETR_F) { LM_DBG("retransmission\n"); return; } Does this in fact retransmit the message to the capture server? Has anyone else load tested kamailio with siptrace on? I'm using Kamailio 4.1. I'm also using the setflag(..) function in siptrace module to duplicate all forwarded messages to the capture server. Regards, Grant

2 3

Panning next major release - v4.4
by Daniel-Constantin Mierla 11 Jan '16

11 Jan '16

Hello, sketching the road to the next major release, so people can plan their goals for it and discuss, if needed, before the start of winter holidays: - development freezing by end of January 2016 - test for one to one and a half month - release in the first part of March 2016 If there are important topics to decide on, we can organize an IRC meeting sometime in January 2016. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

4 10

behaviour of exit();
by Grant Bagdasarian 11 Jan '16

11 Jan '16

Hello, I have the following statements in my config script. When either of the IF statements is true, what does the exit function do? Will it still pass the message to the next hop or will it just abort and drop the sip message? failure_route[DISPATCH_FAILURE] { xlog("L_INFO", "[R-FAILURE-ROUTE-DISPATCH-FAILURE:$ci] !>" "Route:DISPATCH_FAILURE;UnixTime:$TV(un);RequestURI:$rU;RequestMethod:$rm;Status:$rs;FromTag:$ft;ToTag:$tt;CallID:$ci;From:$fU;To:$rU\r\n"); t_on_branch("MSG_MOD"); if(t_is_expired()) { xlog("L_INFO", "[R-FAILURE-ROUTE-DISPATCH-FAILURE:$ci] !>" "Transaction expired - aborting\r\n"); exit; } if (t_is_canceled()) { xlog("L_INFO", "[R-FAILURE-ROUTE-DISPATCH-FAILURE:$ci] !>" "Transaction cancelled - aborting\r\n"); exit; } Thanks. Regards, Grant

2 2

Ways to reload Kamailio configuration file without restart
by Amit Patkar 11 Jan '16

11 Jan '16

Hi What are different ways to reload Kamailio configuration file without restart? I need to make configuration file changes on production server without bringing down kamailio service. *Thanks & Regards,* Amit Patkar

2 3

Re: [SR-Users] RTPPorxy -> RTPEngine migration issue
by Arsen Hovhanissian 10 Jan '16

10 Jan '16

Richard that was a champion’s answer thank you so much! Maybe it should be noted kamailio's rtpengine module documentation.

1 0

Re: [SR-Users] RTPPorxy -> RTPEngine migration issue
by Arsen Hovhanissian 10 Jan '16

10 Jan '16

Hi Richard! Here’s the output at log level 7 [1452404737.610730] WARNING: Failed to properly parse UDP command line '11683_0 d7:command4:pinge' from 10.0.0.10:54602, using fallback RE [1452404737.619006] WARNING: Failed to properly parse UDP command line '11683_1 d7:command4:pinge' from 10.0.0.10:59567, using fallback RE [1452404737.621995] WARNING: Failed to properly parse UDP command line '11694_2 d7:command4:pinge' from 10.0.0.10:57256, using fallback RE [1452404737.623204] WARNING: Failed to properly parse UDP command line '11689_1 d7:command4:pinge' from 10.0.0.10:52176, using fallback RE [1452404737.629991] WARNING: Failed to properly parse UDP command line '11687_1 d7:command4:pinge' from 10.0.0.10:55951, using fallback RE [1452404737.630268] WARNING: Failed to properly parse UDP command line '11686_1 d7:command4:pinge' from 10.0.0.10:51267, using fallback RE [1452404737.630392] WARNING: Failed to properly parse UDP command line '11690_1 d7:command4:pinge' from 10.0.0.10:55452, using fallback RE [1452404737.634052] WARNING: Failed to properly parse UDP command line '11688_1 d7:command4:pinge' from 10.0.0.10:49329, using fallback RE [1452404737.635731] WARNING: Failed to properly parse UDP command line '11693_1 d7:command4:pinge' from 10.0.0.10:43723, using fallback RE [1452404737.636118] WARNING: Failed to properly parse UDP command line '11685_1 d7:command4:pinge' from 10.0.0.10:55434, using fallback RE [1452404737.636293] WARNING: Failed to properly parse UDP command line '11692_1 d7:command4:pinge' from 10.0.0.10:34893, using fallback RE Could it be that I am starting it up wrong? Thanks a million!

2 1

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