sr-users September 2016

sr-users@lists.kamailio.org

87 participants
128 discussions

Re: [SR-Users] help with kamailio rpm made from source
by Dmitry 18 Sep '16

18 Sep '16

Hello I tested a package http://download.opensuse.org/repositories/home:/kamailio:/v4.3.x-rpms/CentO… ( I downlowaded 4.3.6 version rpm and installed it. Kamailio restarts well. The behaviour is the same. the phone registeres without nonce. I install kamailio, kamailio-kazoo - 2 rpm зфслфпуы On Thursday, September 15, 2016 9:34 PM, Dmitry <mbike2000ru(a)yahoo.com> wrote: I made loadmodule and modparam("debugger", "cfgtrace", 1) but anyway - no logs when I register. As … [View More]I understand - it's like no config file. On Thursday, September 15, 2016 6:01 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I am not familiar with kazoo configs, maybe asking on their mailing list can help you more. From Kamailio point of view, you can load debugger module and set its cfgtrace parameter to 1, then see what actions from config are executed and why is not getting to the authentication part. Cheers, Daniel On 15/09/16 14:09, Dmitry wrote: /etc/kazoo/kamailio/default.cfg - which containes all routes. 2600hz/kazoo-configs | | | | | | | | | | | 2600hz/kazoo-configs kazoo-configs - Kazoo Configuration Files for Software We Use | | | | I test on a working server (testing one) and a working config -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda On Thursday, September 15, 2016 3:56 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Are you using default kamailio.cfg or another one? Cheers, Daniel On 15/09/16 12:39, Dmitry wrote: Hello I took this spec from suse. It generates no errors. When I installed from the RPM I had made - the phone register, but The phone sends a REGISTER and the KAmailio sends 200ok back to the phone (so no NONCE authorization) and no logs during it. In default.cfg I set L_DBG but no logs are generated during registering a phone. And I see no AMQP messages (I set to 1 Kazoo_enable in the SPEC file). Why there are no logs during register (I saw register in tcpdump)? On Wednesday, September 14, 2016 7:42 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Then you just need add those files in various packages inside the spec file, so they are not detected to be orphaned. Maybe you can inspire from: - https://build.opensuse.org/package/view_file/home:kamailio:v4.3.x-rpms/kama… Cheers, Daniel On 14/09/16 16:35, Dmitry wrote: 4.3.4 version is for Kazoo It is on production server currently. I need to rebuild the current RPM so as to apply patches. But first I want to get a working Kamailio and only after it I will apply the patches. I think I may take a list of modules from the production Kazoo-kamailio and rearchive the tar.gz because now the Fedora 18 rpmbuild gives a lot of files: RPM build errors: Installed (but unpackaged) file(s) found: /usr/lib64/kamailio/modules/auth_xkeys.so /usr/lib64/kamailio/modules/dmq_usrloc.so /usr/lib64/kamailio/modules/ims_charging.so /usr/lib64/kamailio/modules/jsonrpc-s.so /usr/lib64/kamailio/modules/mohqueue.so /usr/lib64/kamailio/modules/nosip.so /usr/lib64/kamailio/modules/rtjson.so /usr/lib64/kamailio/modules/rtpengine.so /usr/lib64/kamailio/modules/statsd.so /usr/lib64/kamailio/modules/tcpops.so /usr/lib64/kamailio/modules/tsilo.so /usr/share/doc/kamailio/modules/README.auth_xkeys /usr/share/doc/kamailio/modules/README.dmq_usrloc /usr/share/doc/kamailio/modules/README.ims_charging /usr/share/doc/kamailio/modules/README.ims_registrar_scscf /usr/share/doc/kamailio/modules/README.jsonrpc-s /usr/share/doc/kamailio/modules/README.mohqueue /usr/share/doc/kamailio/modules/README.nosip /usr/share/doc/kamailio/modules/README.rtjson /usr/share/doc/kamailio/modules/README.rtpengine /usr/share/doc/kamailio/modules/README.statsd /usr/share/doc/kamailio/modules/README.tcpops /usr/share/doc/kamailio/modules/README.tsilo /usr/share/snmp/mibs/KAMAILIO-MIB /usr/share/snmp/mibs/KAMAILIO-REG-MIB /usr/share/snmp/mibs/KAMAILIO-SIP-COMMON-MIB /usr/share/snmp/mibs/KAMAILIO-SIP-SERVER-MIB /usr/share/snmp/mibs/KAMAILIO-TC On Wednesday, September 14, 2016 5:05 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, any reason not to use series 4.4.x? Iirc, the latest spec that got update on 4.4 are those for oracle enterprise linux, perhaps is something that you can reuse a lot for upgrading to the centos flavour. On the other hand, you can use opensuse build service if you want to build yourself, the spec from the kamailio project there is up to date and the ones for 4.3 should still be there. Anyhow, as I said, I would recommend to go with 4.4.x version. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda [View Less]

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avp_db_query
by Angel Elena 18 Sep '16

18 Sep '16

Hi, good moorning. I have a Kamailio 4.0 with avp module working. Now, I need write some values in my database and I don't find and example for this. if (avp_db_query("update subscriber set bloqueado=0 where name=xxxxx")) { } Is possible make this operation with avp_db_query ? Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ --------------------------------

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Replacing Asterisk with Kamailio
by Valter Nogueira 17 Sep '16

17 Sep '16

I use Asterisk for SIP and Media Proxy. Despite the fact that Asterisk is not a SIP Proxy at all. Customer registers in a SIP account, sends the invite and thru de context Asterisk dials out thru a SIP Trunk. Asterisk does the media proxy, since customer can't route directly to the SIP Trunk (altough it has a valida address, it don't have a public route allowed to it). I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, … [View More]

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Commercial SBC or Kamailio
by Linux Vince 16 Sep '16

16 Sep '16

We are trying to setup VOIP infrastructure, mainly wholesale and retail. What is best option if money is not the problem? Developing our own infrastructure using Kamailio and other open source packages to act as SBC and switch or buy a commercial solution like GenBand/Sonus/Sansay. We are looking for high performance with scalability to handle thousands of call setups per second over the period of few years. I have no idea on how/if can kamailio outperform commercial solutions or not. … [View More]

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Kamailio fails to start when MySQL down
by Daniel Pocock 16 Sep '16

16 Sep '16

I recently set up a Kamailio instance using the default configuration for HOMER with MySQL[1] The database was not running and Kamailio refused to start. Would it be better for Kamailio to start anyway and go into a loop trying to connect to the database, just as if auto_reconnect was set? This is useful for people who start Kamailio from their init script and can't be sure if their database host is up before their Kamailio host boots 1. https://github.com/sipcapture/homer-api/blob/master/examples/sipcapture/sip…

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Integration with Broadsoft and Config Question
by Daryn Johnson 16 Sep '16

16 Sep '16

I’m looking to briefly check the sanity of my configuration. I have been working w/Kamailio the last few months, and have read the SIP router ebook. Thank you for your efforts and the documentation. It has been a GREAT HELP! We are attempting to replace an Acme Packet SBC, and our setup is as follows: |ACCESSUSERS| ⇔ |KAMAILIO/SEMS| ⇔ |BROADSOFT| ⇔ |GENBAND| ⇔ |PSTN| Kamailio & SEMS are (currently) on the same box, and communicate using localhost via different ports. They also listen on … [View More]the same private (core) IPs, using different ports. Since Broadsoft doesn’t support the PATH header, I manipulate the contact header w/the SEMS privateIP & port in the contact header. This causes it to be stored in the Broadsoft Registration string, and calls from PSTN route directly to SEMs to start calls etc… Registraton code looks like this: route[REGISTRATION] { remove_hf("Route"); add_path_received(); $avp(contact) = $ct; xlog("L_INFO", "found contact header $avp(contact)"); setflag(FLAG_REGISTRATION); if (nat_uac_test("20")){ xlog("L_INFO", "[REGISTRATION_ATTEMPT] [FROM_NAT] R=$ru ID=$ci UA='$ua'\n"); force_rport(); remove_hf("Contact"); append_hf("Contact: <sip:$fU" + "@" + "!SEMSPRIVATEIP!:5060;" + "alias=$si~$sp~1>\n"); setflag(FLT_NATS); } else { xlog("L_INFO", "[REGISTRATION_ATTEMPT] [PUBLIC_IP] \r\n[REGISTRATION_INFO] R=$ru UA='$ua'\r\n"); remove_hf("Contact"); append_hf("Contact: <sip:$fU@" + "!SEMSPRIVATEIP!:5060;" + "alias=$si~$sp~1>\n"); } $du = "sip:R_IP:REGISTRAR_PORT"; t_on_reply("REG"); route(RELAY); exit; } Then in the relay route, I force the registration out of the private interface & port of Kamailio if (isflagset(FLAG_REGISTRATION)){ $fs = "LISTEN_UDP_PRIVATE"; } Finally, if a registration is successful from BROADSOFT, I store it in the location table, to allow me to screen calls from access to ensure they are ‘registered’ onreply_route[REG]{ if(status=="200") { xlog("L_INFO", "REGISTRATION BE GOOD FOR $fu"); save("location", "0x04"); } } Calls from ACCESS, I check USRLOC for screening, then pass the call to SEMS via the localhost:port. From there, sems enacts the B2BUA/SBC module and routes the call to BROADSOFT, and along to our GENBAND Gateway… I could never get calls to work properly without SEMS, just using RTP PROXY/ENGINE, possibly because the RTP for the GENBAND & KAMAILIO/SEMS are on different private subnets. Our setup is fairly legacy ‘heavy’ and many things can’t be changed. Calls from PSTN route to GENBAND, then BROADSOFT, then SEMS, then KAMAILIO. Since I have manipulated the contact header, BROADSOFT maintains that in the INVITE and I can use the ‘handle_ruri_alias’ to route the call properly. SEMS sets the PUBLIC IP of the KAMAILIO/SEMS box in the SDP, and facilitates rtp relaying. if (isflagset(FLAG_TOUSER)){ xlogl("L_INFO", "TOUSERFLAGSET"); if(is_method("INVITE|BYE")){ handle_ruri_alias(); } /*if(has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); xlog("L_INFO", "SDP CONTAINS NAT"); }*/ $fs = "LISTEN_UDP_PUBLIC"; } I’ve gone headlong into this setup, and it works! But, I wanted to get a ‘Sanity Check’ to determine if there is a better way to do this. We are at the point where I’m doing load testing and attempting to validate this for a ‘production’ ready state. If possible, I’d just like to ask the community if they have encountered a similar setup, and how they solved it? Thanks in advance!!! Daryn Johnson Senior VoIP Engineer 248.485.1109 www.telnetww.com 1175 W. Long Lake Rd. | Suite 101 | Troy, MI 48098 [View Less]

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Kamailio Authentication failing
by Eric Koome 16 Sep '16

16 Sep '16

Hi all - my Kamailio - 4.1.6 is receiving this particular structured INVITES from multiple IPs, and for some reason it is not requesting authentication. I have AUTH & IPAUTH modules in use for two years now, but this is bypassing that and actually forwarding the invite to asterisk servers behind Kamailio. I notice from the invite that the contact (c=IN IP4 10.10.10.10) is unusual and in the private range. Is this what is bypassing Authentication? Any Pointers on how to stop this. This is … [View More]

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http_async_query - How to Allow SIP transaction to continue
by Jonathan Hunter 16 Sep '16

16 Sep '16

Hi Guys, I am just testing with jansson and http_async_query. Essentially I sent an INVITE into kamailio ,it hits CALL_START_REQUEST (see below) and the query allows the http site I send to, to provide routing details for kamailio. Logging the $rm that hits CALL_START_ REQUEST, it shows as INVITE, however when CALL_START_REQUEST_REPLY is hit, the message type is OPTIONS, is this expected behaviour? Essentially I want it to continue in the script with the same $ci,$rm so I can onwards route it … [View More]

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help with kamailio rpm made from source
by Dmitry 16 Sep '16

16 Sep '16

Hello I obtained the source from https://www.kamailio.org/pub/kamailio/4.3.4/src/kamailio-4.3.4_src.tar.gz I found a SPEC file there and I prepared a RPM with this spec file. When I start a Kamailio (from this rpm) - it starts but the LOG file gives only 1 line: INFO: <core> [tcp_main.c:4657]: init_tcp(): using epoll_lt as the io watch method (auto detected) I want to test this rpm and if successfull i willl add patches and rebuild the rpm. How to get correct RPM?

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Branch variables in branch_failure_route
by Paul Smith 14 Sep '16

14 Sep '16

Hi, I am struggling to figure out variable scope in branches and branch_failure_routes. Is there a way to store a variable in a branch route so that I can then read it from the branch failure route if that branch fails? Also does writing $ru in branch_failure_route append a branch? For example I have multiple SIP devices registered to Kamailio on the same account credentials. The devices are behind same NAT, so same public IP. Some devices have SRTP required and some cannot use SRTP at all,… [View More] Kamailio does not know in advance of the call whether the call will use SRTP or RTP. When I send a call to that user Kamailio correctly parallel forks to all the registered devices. I use rtpengine offer to send SDP with RTP/SAVP in the initial invites. If one of the devices sends a 488 due to not being able to handle the RTP/SAVP I then want to run rtpengine_offer with RTP/AVP and send another invite to the same device. In the branch failure route I seem to have to read the current $ru and then write it back into $ru before calling t_relay() or I get Sep 14 14:56:06 registrar-secure /usr/sbin/kamailio[10168]: ERROR: tm [t_fwd.c:1771]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding This code seems to work, but it feels like I’m doing something very wrong…. in particular $avp(triedrtpout) is set for the transaction, I should be setting it separately for each branch. event_route[tm:branch-failure:SRTP] { # Handle failure response xlog("L_INFO", "Handling $T_reply_code response to $rm to ru: <$ru> and du: $du\n"); if(t_check_status("488") && is_method("INVITE")) { if ($avp(triedsrtpout)==1) { # set $ru seems to create a new branch ... even when we set it to its current value. # we want to generate a new SIP INVITE to the same AOR but with different SDP $avp(buffer)=$ru; $ru=$avp(buffer); xlog("L_INFO","488 caught. Resending to $ru"); # try with unencrypted RTP/AVP rtpengine_delete(); rtpengine_offer("to-tag trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); if (!t_relay()) { sl_reply_error(); } } } } I am using kamailio 4.2 debian package for this registrar. Sorry for the rambling question … I don’t understand enough to write the short version! Paul [View Less]

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sr-users September 2016