- sr-users - kamailio.org

kamailio 5.2.5 core WARNING
by Andrew Chen 03 Mar '20

03 Mar '20

Hi all, Can someone help shed some light as to what this core WARNING means? root@ashmaintpsg51:/etc/kamailio # kamailio -c kamailio.cfg 0(2192) WARNING: <core> [core/ppcfg.c:220]: pp_ifdef_level_check(): different number of preprocessor directives: N(#!IF[N]DEF) - N(#!ENDIF) = 2 Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Dynamically change the transport protocol from UDP to TCP to avoid fragmentation
by supreeth herle 03 Mar '20

03 Mar '20

Hello Everyone, I have an IMS setup using Kamailo and is working as expected when the 4G Core Network and IMS is running in a single VM. However, when running in two different VMs registration is successful but calling fails. As per the scripts in kamailio ims example folder for P-CSCF, uac module is used to generate OPTIONS message to achieve a kind of NATPINGING to UEs. The ping achieved in this way times out with 408 when IMS and Core Network is run in different VMs. I suspect that the increasing packets size could be the issue for UE not responding to OPTIONS from P-CSCF (I have verified that OPTIONS message from P-CSCF is reaching the eNB). In this aspect, I would like to switch the transport protocol to TCP when MTU is greater than 1300 (please guide me if my thinking of switching to TCP to tackle NAT issue is wrong). I have explored the following options: tcp_reuse_port=yes udp_mtu=1300 udp_mtu_try_proto=TCP But, when I use these options, any request message which hit 1300 MTU mark are not sent out at all (in my case it is the NOTIFY request from SCSCF) and Kamailio has a warning saying that binding to address failed as the address is already in use. Following is the small part of the log: Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19712]: DEBUG: ims_registrar_scscf [registrar_notify.c:2275]: notification_event_process(): About to free notification Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: PCSCF MO_reply: Destination URI: <null> Request URI: <null> Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header present. Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Source IP and Port: (10.4.128.21:6060) Route-URI: Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Received IP and Port: (10.4.128.21:5060) Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Contact header: <sip:scscf.ims.mnc001.mcc001.3gppnetwork.org:6060> Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: ERROR: <script>: NOTIFY (tel:0198765432100 (172.24.15.30:6060) to tel:0198765432100, 4125089758_4268630712(a)192.168.101.2) Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Within DLG Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Within loose route Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: PCSCF MO_indialog: Destination URI: sip:172.24.15.10:5060 Request URI: sip: 192.168.101.2:5060 Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Source IP and Port: (172.24.15.30:6060) Route-URI: sip:mo@172.24.15.30;lr=on;ftag=4125089761 Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Received IP and Port: (10.4.128.21:5060) Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Contact header: <sip:scscf.ims.mnc001.mcc001.3gppnetwork.org:6060> Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: WARNING: <core> [core/tcp_main.c:1298]: tcp_do_connect(): binding to source address 10.4.128.21:5060 failed: Address already in use [98] I could be assuming things here about huge packet sizes causing the issue, if anyone has experienced similar issue in NATed scenario please let me know some hints or pointer to solve this. Thanks in advance. Best Regards, Supreeth

1 1

push notification via lua script on tsilo module
by max 03 Mar '20

03 Mar '20

I am expert in voip solution such as Asterisk and familiar with Kamailio .in one of our project we use the linphone app as a client mobile and we have to solved registering issue when extension has been unregistered in background mode. so i try to implement this scenario according federico document and enable extra module like this: loadmodule "tsilo.so" loadmodule "htable.so" #loadmodule "app_lua.so" loadmodule "cfgutils.so" I must disable app_lua.so because i haven't suitable lua script .i think most parts is OK and i can see ts_store parameters: ts.dump { Size: 1024 R-URIs: { R-URI: sip:1003@192.168.1.169:25378;rinstance=6cb3b1cc5efabdfb Hash: -238368019 Transactions: { Transaction: { Tindex: 9977 Tlabel: 1825563922 } } } Stats: { RURIs: 1 Max-Slots: 1 Transactions: 1 } } by the way push notification feature is enable in linphone app . Now i need a script (lua) for handling this issue (push notification for app) when we fetch SEND PUSH route. Could you please advice me ? do you have any example of this script? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 2

Kamailio - Force UAC to re register
by Sharad Kumar 03 Mar '20

03 Mar '20

Hey guys, I am implementing one scenario where I want the help of very smart people. So let’s say one UAC 999 registers to Asterisk 1 server via kamailio and asterisk 1 server goes down. Is there any way I can move that registration to another asterisk server or is there any way I can force UAC to re register so that I can route the new Register to the available asterisk ? Any help will be really appreciated. Thank you

2 1

Siremis web interface issue -- siptrace module active
by Bugaian A. Vitalie 02 Mar '20

02 Mar '20

Hi list, there is an issue in latest git version of Siremis with showing sip method content. If I select it from mysql I see it all good: select * from sip_trace where method = 'INVITE'; If I click it from web interface and select needed method I get only first register from sip_trace table. You can reproduce by /siremis/sipadmin/sip_trace_list <http://sbc.pride.md/siremis/sipadmin/sip_trace_list> then click on the method to see details. I tested on opera and chrome. Same result, something wrong with javascript, its not passing correctly parameter for selecting. Please have a look let me know if I can help. Thanks. Regards Vitalie A. Bugaian

2 9

Strange behaviour on $tu
by Duarte Rocha 02 Mar '20

02 Mar '20

Greetings, I have a cenario with the following header To = "sip:111234567890;npdi@111.222.333.444" If i print $tu I got "sip:111234567890". If i print $hdr(To) I get "sip:111234567890;npdi@111.222.333.444". If i don't include the "npdi" or any other user parameters, it works fine Is this the intended behaviour? Am i doing something wrong here? Best Regards,

2 3

How to detect NAT during authenticated registration of clients which adjust the headers
by Awal Junanto 02 Mar '20

02 Mar '20

Hi, We are building a service where we need to detect NAT when the clients register to our server. We are struggling in analyzing NAT status of some clients which modify their IP addresses/ports in the headers according to the value of "received" parameter sent during "401 Unauthorized" response. Here's the flow: Client->Server REGISTER sip:... Via: SIP/2.0/TLS 192.168.0.1:41157 ;rport;branch=z9hG4bKPj30093e5d-550d-4d4c-a9a2-22c3bd1cda7e;alias Contact: <sip:user@192.168.0.1:42251;transport=TLS;ob> ... Server->Client SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.0.1:41157 ;rport;branch=z9hG4bKPj30093e5d-550d-4d4c-a9a2-22c3bd1cda7e;alias;received=1.2.3.4 WWW-Authenticate: ... ... Client->Server REGISTER sip:... Via: SIP/2.0/TLS 1.2.3.4:6201 ;rport;branch=z9hG4bKPj30093e5d-550d-4d4c-a9a2-22c3bd1cda7e;alias Contact: <sip:user@ 1.2.3.4:6201;transport=TLS;ob> Authorization: ... ... By the time the client is authenticated, there is no way to detect whether the request was coming from a natted device or not by just analysing the Via or Contact headers. Thanks in advance.

3 6

using kamailio as sip server
by Mohammad Reza Keshavarzianpoor 02 Mar '20

02 Mar '20

Hi, I'm new to Kamailio I need to have a kamalio as sip server which do a trunk to NGN Center(like a friend peer trunk in freepbx). I need to outgoing calls go through kamailio and forwarded to ngn(multi did values, a range of numbers ...) extensions will register on kamailio I need incoming calls forwarded to an extension or a pbx server, doesn't matter and both are good ... that server will have heavy traffic about 1000 concurrent calls... I tried a successful kamailio installation but I don't know how to config that for my purpose. even I tried installation of dsiprouter over a fresh debian 9 but stopped on pyspark installation. please give me your helps, advises, hints or ideas! best regards

2 1

Re: [SR-Users] dlg.end_dlg on RTP timeout
by Miteshkumar Thakkar 28 Feb '20

28 Feb '20

This is really helpful development. I am facing one issue, when I use this. RTPEngine is sending two "dlg.terminate_dlg" events for single call to kamailio with same call-id but reversed to-tag and from-tag in second event. I believe this is due to two separate RTP streams. Problem here is that - in first event kamailio terminates the call, hence when second event comes, kamailio says - call-id not found and send response back to RTPEngine. But still RTPEngine keeps sending that event message for three more times. Kamailio has to respond to it with call-id not found. Which is overhead. Not recommended for production environment. How can I avoid this? Any help is appreciated. Thank you Mitesh That's great, thank you! Cheers, Daniel On 06.09.18 11:15, Richard Fuchs wrote: >* This is now supported as per * >* https://github.com/sipwise/rtpengine/commit/89084da8d820919b44a0244e16e6701… <https://github.com/sipwise/rtpengine/commit/89084da8d820919b44a0244e16e6701…> * > >* Cheers * > >* On 2018-09-05 05:39, Daniel-Constantin Mierla wrote: * >> >>* There is the dlg.terminate_dlg rpc command that requires callid, * >>* from-tag and to-tag as parameters: * >> >>* * * >>* https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dlg.r.terminate… <https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dlg.r.terminate…> * >> >>* So it expects something like: * >> >>* <?xml version="1.0" encoding="UTF-8"?> * >>* <methodCall> * >>* <methodName>dlg.terminate_dlg</methodName> * >>* <params> * >>* <param><value><string>_CALLID_VALUE__</string></value></param> * >>* <param><value><string>_FROM_TAG_VALUE__</string></value></param> * >>* <param><value><string>_TO_TAG_VALUE__</string></value></param> * >>* </params> * >>* </methodCall> * >> >>* I planned to make the from-tag and to-tag optional for quite some * >>* time, but didn't get the time for it yet. * >> >>* Cheers, * >>* Daniel * >> >>* On 05.09.18 08:52, Richard Fuchs wrote: * >>>* Yup that's exactly right. * >>> >>>* It would be fairly simple to implement an additional XMLRPC format * >>>* if there's a particular one that's more friendly towards Kamailio. * >>> >>>* Cheers * >>> >>>* On 2018-09-05 02:42, Daniel-Constantin Mierla wrote: * >>>> >>>>* Looking quickly at the readme of rtpengine application and digging * >>>>* a bit with google, it is something like rtpengine has to be started * >>>>* with * >>>> >>>>* -b http(s)://myrpcserver.ip/path -x 1 * >>>> >>>>* and then the xmlrpc request is going to be sent to that url, having * >>>>* a format like: * >>>> >>>>* <?xml version="1.0" encoding="UTF-8"?> * >>>>* <methodCall> * >>>>* <methodName>teardown</methodName> * >>>>* <params> * >>>>* <param><value><string>_CALLID_VALUE__</string></value></param> * >>>>* </params> * >>>>* </methodCall> * >>>> >>>>* Is it right? If yes, then I can try to make a sample config that * >>>>* could handle it using xmlops, xhttp and jsonrpcs modules. * >>>> >>>>* Cheers, * >>>>* Daniel * >>>> >>>> >>>>* On 05.09.18 08:24, Richard Fuchs wrote: * >>>>>* It does an XMLRPC callback. Currently there's two formats for it, * >>>>>* one is a sems sbc teardown request (using the from-tag), the other * >>>>>* is a generic "teardown" command using the call ID. * >>>>> >>>>>* Cheers * >>>>> >>>>>* On 2018-09-04 07:52, Daniel-Constantin Mierla wrote: * >>>>>> >>>>>>* Hello, * >>>>>> >>>>>>* what do you get from rtpengine on rtp timeout? An RPC call back * >>>>>>* or an http request? * >>>>>> >>>>>>* Cheers, * >>>>>>* Daniel * >>>>>> >>>>>> >>>>>>* On 04.09.18 12:48, Igor Olhovskiy wrote: * >>>>>>>* Hi all! * >>>>>>> >>>>>>>* Is there any way to end dialog in Kamailio on RTP Timeout event * >>>>>>>* on RTPEngine? * >>>>>>> >>>>>>>* Or only look at logs/redis database with external tool? * >>>>>>> >>>>>>>* Regards, Igor * >>>>>>> >>>>>>> >>>>>>>* _______________________________________________ * >>>>>>>* Kamailio (SER) - Users Mailing List * >>>>>>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>>>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>> >>>>>>* -- * >>>>>>* Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com> * >>>>>>* www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> * >>>>>>* Kamailio World Conference -- www.kamailioworld.com <http://www.kamailioworld.com> * >>>>>>* Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com <http://www.asipto.com> * >>>>>> >>>>>> >>>>>>* _______________________________________________ * >>>>>>* Kamailio (SER) - Users Mailing List * >>>>>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>> >>>>> >>>>>* _______________________________________________ * >>>>>* Kamailio (SER) - Users Mailing List * >>>>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>> >>>>* -- * >>>>* Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com> * >>>>* www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> * >>>>* Kamailio World Conference -- www.kamailioworld.com <http://www.kamailioworld.com> * >>>>* Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com <http://www.asipto.com> * >>> >>> >>>* _______________________________________________ * >>>* Kamailio (SER) - Users Mailing List * >>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >> >>* -- * >>* Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com> * >>* www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> * >>* Kamailio World Conference -- www.kamailioworld.com <http://www.kamailioworld.com> * >>* Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com <http://www.asipto.com> * > > >* _______________________________________________ * >* Kamailio (SER) - Users Mailing List * >* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

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Teams integration
by Bugaian A. Vitalie 28 Feb '20

28 Feb '20

Hello list, its been about one month for me playing with kamailio and I need some help to sort out a real life situation. I followed this guide https://skalatan.de/en/blog/kamailio-sbc-teams; great article, also got some inspiration from here https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/..articles look very similar. I manged to get my calls out to kamailio from TEAMS it works perfectly call gets connected and codec negotiation is fine, but I am facing issue geting the call inbound to TEAMS. My invite looks like this: ======================== U 217.26.163.205:5060 -> 52.114.75.24:5061 #22 INVITE sip:+37360844269@sip.pstnhub.microsoft.com:5061 SIP/2.0. Record-Route: <sip:sbc.pride.md:5061;transport=tls;lr>. Record-Route: <sip:217.26.163.205:5061;nat=yes;lr>. Via: SIP/2.0/UDP 217.26.163.205;branch=z9hG4bK7838.7417f50ed201fcada9609f5b7c4e520f.0. Via: SIP/2.0/UDP 192.168.169.102:5060 ;received=46.214.187.67;branch=z9hG4bK80e3f7e5cc50ea11806b6eeeb899592c;rport=5060. From: "+37379844267" <sip:+37379844267@sbc.pride.md>;tag=1604785394. To: "+37360844269" <sip:+37360844269@sip.pstnhub.microsoft.com:5061 ;user=phone>. Call-ID: 80E3F7E5-CC50-EA11-8069-6EEEB899592C(a)192.168.169.102. CSeq: 227 INVITE. sip:+37379844267@192.168.169.102:5060;gr=008A94E3-CA50-EA11-805B-6EEEB899592C;alias=46.214.187.67~5060~1Content-Type: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK. Max-Forwards: 69. User-Agent: SIPPER for PhonerLite. Session-Expires: 1800. Supported: 100rel, replaces, from-change, gruu, timer. P-Preferred-Identity: <sip:+37379844267@sbc.pride.md>. Content-Length: 362. Contact: <sip:+37379844267@sbc.pride.md:5061;user=phone;transport=tls>. . v=0. o=- 2307737351 1 IN IP4 217.26.163.205. s=SIPPER for PhonerLite. c=IN IP4 217.26.163.205. t=0 0. m=audio 36864 RTP/SAVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:owGy5+mMZNyO5+9lFmUUOK3WqpEsxJH0+jtWz =============================================== Anybody has a good invite exaple for Teams? Or do you see any issue with my invite? I do use : record_route_preset("sbc.pride.md:5061;transport=tls","217.26.163.205:5060 ;nat=yes"); add_rr_param(";r2=on"); before sending this out. Please let me know if you can help. Thanks. Vitalie.

3 7

Trouble with 302 Redirect
by Jim Leahy 27 Feb '20

27 Feb '20

Hi All, I'm having some issues trying to properly send a '302 Moved Temporarily' reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server when UAC2 is in DND. Here are the pertinent parts of my config: route { # drop some things right away route(DROP); # perform sanity check route(SANITYCHECK); # handle registrations if(is_method("REGISTER")) { route(REGISTRATION); exit; } # handle invites if(is_method("INVITE")) { route("INVITE"); exit; } route(RELAY); } route[INVITE] { # add this proxy to the record-route so it stays in the loop record_route(); # if we can find this user in the location DB... if(lookup("location")) { # relay the invite to the new dURI t_on_failure("FOURDIGITFAIL"); t_on_reply("FOURDIGITREPLY"); route(RELAY); # go back to the main route return; } onreply_route[FOURDIGITREPLY] { if(t_check_status("4[0-9]{2}")) { t_reply("302", "Moved Temporarily"); } } I'm having issues with the t_reply in the onreply_route. It successfully sends the reply, but I can't seem to set the Contact header properly. I'm trying to set it to the location of the user's mailbox on the voicemail sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: remove_hf("Contact"); insert_hf("Contact: sip:1000@vmpbx.domain.com"); t_reply("302", "Moved Temporarily"); But as I understand it, that just changes the header in the reply from UAC2, not the reply I'm sending to UAC1 Then I tried: append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); t_reply("302", "Moved Temporarily"); But I can't use append_to_reply in an onreply block. How can I change the Contact header in the reply that's sent? Am I just going about this the wrong way? Any help would be appreciated. Thanks

5 4

Rtpproxy not work in 2 kamailio + rtpproxy setup
by Voip support 27 Feb '20

27 Feb '20

Dear Community, I have a problem with rtpproxy not changing the SDP IP in connection information in following scenario: SIP1 -> kamailio1 + rtpproxy1 -> kamailio2 + rtpproxy2 -> SIP2 The first kamailio put SDP IP correctly (from rtpproxy) then the second kamailio does not change SDP connection information. I think its related to SDP body which has: a=nortpproxy:yes line added by kamailio1. So i tried to remove the a=nortpproxy:yes line from SDP before calling rtpproxy_manage. But it looks like no difference in working - i think the SDP sent to rtpproxy is still having the line even when i removed it. The issue is not happening when i remove kamailio1 + rtpproxy1. In that case my IP is put correctly in SDP in INVITE and in 200 OK. In case of two kamailio with rtpproxy (kamailio1 and kamailio2) in signalling path i am getting: rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy for replies (SIP2 IP address in 200OK SDP is present instead of kamailio2 IP address). For request INVITE i am not getting the error but the IP in SDP doesn't change. kamailio2 config is now like below (but the attempt to remove a=nortpproxy:yes, did not work for me) # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } # if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; -- commented this out because sometimes rtpproxy was not used and i wanted to use it always # added below code to remove a=nortpproxy:yes from SDP before rtpproxy_manage but its still not working if( has_body("application/sdp") && (1 == sdp_get_line_startswith("$avp(nortpproxy)", "a=nortpproxy" )) ){ sdp_remove_line_by_prefix("a=nortpproxy:yes"); xlog("removed nortpproxy:yes from SDP"); } if(nat_uac_test("8")) { xlog("rtpproxy CO"); rtpproxy_manage("co","MY_PUBLIC_IP_HERE"); } else { xlog("rtpproxy COR"); rtpproxy_manage("cor"," MY_PUBLIC_IP_HERE "); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } Best regards, Tom PS. thanks for any hints to resolve the issue.

2 1

record_route_advertised_address causes null send socket error
by Daniel Greenwald 26 Feb '20

26 Feb '20

I am using record_route_advertised_address to insert a record route header with an SRV record. The header is added correctly and everything appears to be working fine however I get a CRITICAL log error when calling record_route_advertised_address('INTERNAL_SRV'); 2020-02-25T20:14:57.078235+00:00,ip-10-0-16-53: CRITICAL: cbbcfe06-d181-467c-bbf4-78772aa0562f 2 16777053 INVITE:<core> [core/msg_translator.c:514]: lump_check_opt(): null send socket 2020-02-25T20:14:57.078352+00:00,ip-10-0-16-53: CRITICAL: cbbcfe06-d181-467c-bbf4-78772aa0562f 2 16777053 INVITE:<core> [core/msg_translator.c:514]: lump_check_opt(): null send socket Any ideas? I'm considering using insert_hf() to manually add it and bypass whatever check is failing. Thoughts? Thanks

3 7

Access to CANCEL messages sent during parallel forking
by Marco Capetta 26 Feb '20

26 Feb '20

Hi All, I have a question regarding call forking and how to access failed branches: the ones for which kamailio sends out the CANCEL because on another one a 200OK was received. This is the scenario: - A calls B - 3 devices are registered on B, so a parallel forking is done to B1, B2 and B3 - B1, B2, and B3 all reply with 180 Ringing - B1 replies with 200OK - Kamailio CANCELs the branches with B2 and B3 What I need is access to each single CANCEL that Kamailio sends out to B2 and B3. In particular I need to know the branch-id and Via branch of all the cancelled branches. Additionally it would be perfect to have them inside the TM module, in order to have access also to internal AVPs. I tried with dedicated failure routes and also with per-branch failure routes, but I was never able to access to those values. I also didn't found a specific event-route that can help me solve this problem. Does anyone have an idea? Kamailio version 5.2.5 Thanks Regards Marco

3 5

Siremis v5.3.0 Released
by Daniel-Constantin Mierla 26 Feb '20

26 Feb '20

Siremis v5.3.0 is out – the open source web management interface for Kamailio SIP Server. This version comes with major changes and many new features, among them the removal of legacy flash charts (now uses pure JS), make the code compatible with PHP 7.x and web views to manage many new modules added up to Kamailio v5.3.x, including the replacement of MI control interface with JSONRPC. This is the first full release in the 5.x series of Siremis. Step by step installation tutorial, screenshots and changelog are available on the web at: * https://kb.asipto.com/siremis:install53x:main * https://kb.asipto.com/siremis:install32x:screenshots * https://github.com/asipto/siremis/blob/v5.3.0/ChangeLog The deployment of Siremis can be also done using Docker, see the guidelines at: * https://github.com/asipto/siremis/tree/master/misc/docker Siremis project is hosted on Github at: * https://github.com/asipto/siremis/ The web site for the project is at: * https://www.siremis.org Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Understanding set_rtp_engine with two sets
by Anthony Alba 25 Feb '20

25 Feb '20

There is a use case mentioned "This is useful if you have a set of RTP proxies that the caller must use, and another distinct set of RTP proxies that the callee must use. This is supported by all rtpengine commands except rtpengine_manage(). " How do you actually implement this - is it something like this? Say caller supposed to target Set 1and callee supposed to target Set 2: # offer/request route set_rtpengine_set(1, 2) rtpengine_offer() # cannot use rtpengine_manage() # callee sees set 2 in SDP, right? # reply_route set_rtpengine_set(2, 1) # reverse the order? Is this correct?? # OR set_rtpengine_set(1, 2) # keep order, module does autoreversal magic # caller sees Set 1 in SDP My question: in the reply route do you keep the same order (module is clever enough to autoreverse so caller sees Set 1) OR do you explicitly reverse the order and THAT means caller sees Set 1. Cheer Anthony

3 2

kamcli postgres support
by Daniel-Constantin Mierla 25 Feb '20

25 Feb '20

Hello, anyone in the community with Python knowledge using Postgres that can spend a bit of time to add support in kamcli? * https://github.com/kamailio/kamcli It should be mainly about the content of the code corresponding to db subcommand: * https://github.com/kamailio/kamcli/blob/master/kamcli/commands/cmd_db.py The target for it is to replace kamdbctl and has some command line variants for couple of its functions, now being implemented for MySQL. The rest of the code relies on sqlalchemy, which should have transparent support for both mysql and postgress (even more). Testing the other commands connecting to database with postgres would be good, though. With more and more diversity in the default shell by main Linux/Unix distros, we hope to be able to replace in the near future the kamctl and kamdbctl tools, which are more like bash scripting (a few issue being reported and fixed lately). Besides that, using python allows better input/parameter checking/validation as well as more options for formatted output. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

2 3

Database errors with usrloc db_mode=1 and dmq_usrloc
by Joel Serrano 25 Feb '20

25 Feb '20

Hello, I have a setup with 2 kamailio in active/idle, locations are shared via dmq_usrloc, but database is also used as a cache where other systems can access and query "real-time" data. The config we have is: # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_load", 0) modparam("usrloc", "db_insert_update", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "server_id_filter", 1) modparam("usrloc", "timer_procs", 2) modparam("usrloc", "timer_interval", 60) # ----- dmq_usrloc params ----- modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) modparam("dmq_usrloc", "batch_msg_contacts", 50) # 50 contacts / message modparam("dmq_usrloc", "batch_size", 10000) # 10000 contacts / batch modparam("dmq_usrloc", "batch_usleep", 500000) # one batch every 500ms With this setup, we assume: 1- almost real-time updated location info in the database. 2- as long as one of the two Kam is up, location information will not be lost (as they will sync with each other via DMQ) 3- if both servers go down, locations are lost (as they will not reload from database on startup). Our tests work well with the above config, with one exception, on the idle node, we see a lot of (only happens on the node receiving location via dmq): Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: <core> [db.c:481]: db_use_table(): invalid parameter value Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: usrloc [ucontact.c:1136]: db_update_ucontact_ruid(): sql use_table failed Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: usrloc [ucontact.c:1657]: update_contact_db(): failed to update database I added debug logs to see if I could understand the reason: Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:478]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@10.2.1.181:5050 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:423]: usrloc_dmq_execute_action(): Received DMQ_UPDATE. Update contact info... Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:75]: add_contact(): aor: 1063362 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:76]: add_contact(): ci->ruid: uloc-1-5e502cd3-5629-1 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:77]: add_contact(): aorhash: 928261667 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:115]: add_contact(): '1063362' Not found in usrloc Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:117]: add_contact(): Insert record Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: usrloc [ul_callback.h:83]: run_ul_callbacks(): contact=0x7fe41ff41f40, callback type 1/15, id 0 entered Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:800]: dmq_ul_cb_contact(): Callback from usrloc with type=1 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:826]: dmq_ul_cb_contact(): Contact received from DMQ... skip Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: <core> [db.c:481]: db_use_table(): invalid parameter value Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: usrloc [ucontact.c:669]: db_insert_ucontact(): sql use_table failed Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: usrloc [urecord.c:598]: insert_ucontact(): failed to insert in database Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:121]: add_contact(): Insert ucontact Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:124]: add_contact(): Release record Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:126]: add_contact(): Unlock udomain Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: sl [sl.c:305]: send_reply(): reply in stateless mode (sl) Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: <core> [core/msg_translator.c:161]: check_via_address(): (10.2.1.181, 10.2.1.181, 0) Can anyone help me understand what's the problem? I have the server_id setting, and the modparam server_id_filter, so shouldn't the receiving node just ignore any database tasks? Thanks, Joel.

3 12

Using the To header to route a call
by Dovid Bender 24 Feb '20

24 Feb '20

Hi, We are fighting with a vendor and trying to explain to them that call routing should be based on the RURI on the INVITE and *not* the To header. I went through RFC 3261 and while there are many "indicators" and it seems like its screaming that we should be using the RURI, is anyone aware of any specific text (that I may have missed) where it says specifically call routing should be done based on the RURI and *NOT* the to header? TIA. Dovid https://tools.ietf.org/html/rfc3261#page-78 1) 8.1.1.1 Request-URI *The initial Request-URI of the message SHOULD be set to the value of the URI in the To field.* - Note that it says SHOULD but is NOT mandatory. Also this is for the *INITIAL* request. *This may or may not be the ultimate recipient of the request.* - Again this is the To field. The RURI is what we are supposed to follow when we want to know where the call should be delivered to. 2) 8.1.2 Sending the Request -Usage of the URI from the *To and From fields in the original request within subsequent requests is done for backwards compatibility with RFC 2543*, which used the URI for dialog identification. *In this specification, only the tags are used for dialog identification*. It is expected that mandatory reflection of the original To and From URI in mid-dialog requests will be deprecated in a subsequent revision of this specification. - While this is for subsequent requests (and not an original one) you see that the only point of the To field is to identify the dialog. 3) 8.2.2.1 To and Request-URI The To header field identifies the *original recipient of the request* designated by the user identified in the From field. - Note that the TO field identifies the ORIGINAL recipient, this **CAN** change. This is basically telling you that you *can't* rely on the To header for routing. *However, the Request-URI identifies the UAS that is to process the request*. - As you can see the Request-RU and *NOT* the TO header tells you where the request should go. 5) 12.2.1.1 Generating the Request a. The UAC uses the *remote target* and route set to build the *Request-URI and Route header field of the request*. - This is self explanatory. it shows that the RURI is using for routing and *NOT* the to header. b. The procedures in Section 8.1.2 will normally result in the request being *sent to the address* indicated by the topmost Route header field value or the *Request-URI* if no Route header field is present. Subject to certain restrictions, they allow the request to be sent to an alternate address (such as a default outbound proxy not represented in the route set). Again self explanatory. 6) 16.5 Determining Request Targets Each target in the set is *represented as a URI*. - This clearly shows that how we route is based on the RURI and *NOT* the to header. 7) 26.5 Privacy Thus it MAY be desirable for privacy reasons to create a To header field that differs from the Request-URI. - Again this shows that the to may not match up the RURI. Think of the To header as the contact in someones phone and the RURI as where we are calling. The to header may be bob(a)1.1.1.1 but the URI is where we are actually calling which is 12346(a)1.1.1.1 or bob_smith(a)1.1.1.1.1

3 2

Kamailio presence when account not equal extension
by Zhan Bazarov 24 Feb '20

24 Feb '20

Hello, I'm using presence and pua modules, it works perfectly when username is equal to extension... But now I have case when username is not equal to the extension.. Like: extension: 1111 username: aythnbghhgt How I can implement this? Any ideas? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 3

CentOS 8 RPMs missing from OBS
by Anthony Alba 21 Feb '20

21 Feb '20

The 5.3.2-4 builds for CentOS 8 are not available here: https://download.opensuse.org/repositories/home:/kamailio:/v5.3.x-rpms/Cent… Is https://rpm.kamailio.org/centos/8/5.3/5.3.2/x86_64/ aka Pulp the official release location now? (With -0 builds) For CentOS 7 Pulp has 5.3.2-0 and OBS has 5.3.2-4. The version mismatch is confusing. Cheers Anthony

2 1

Kamailio presence when account not equal extension
by Zhan Bazarov 21 Feb '20

21 Feb '20

Hello, I'm using presence and pua modules, it works perfectly when username is equal to extension... But now I have case when username is not equal to the extension.. Like: extension: 1111 username: aythnbghhgt How I can implement this? Any ideas? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 1

acc_radius
by volga629 21 Feb '20

21 Feb '20

2 1

Understanding some usrloc modparams
by Joel Serrano 20 Feb '20

20 Feb '20

Hello, I have an installation using usrloc + db_mode=1. If I want on db_mode=1 or db_mode=2 the expired contacts to be cleaned up I have to enable the modparam db_timer_clean: https://www.kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.db… The docs say: "*Enable (1) or disable (0) cleaning of expired db records on timer basis for db_mode WRITE-BACK and WRITE-THROUGH. It uses the secondary timer process.*" I have the following doubts: 1- What does the reference "it uses *the* secondary timer process" mean? Is there a specific secondary timer just for this purpose? 2- Is #1 related to the modparam timer_procs? (so it uses one of those when enabled?) 3- Why/When would you need a timer_proces modparam with a value higher than "1"? Thanks, Joel.

4 6

Active-Active Registrar setup with users behind NAT persistence in event of server failure
by Jonathan Hunter 19 Feb '20

19 Feb '20

Hi Guys, I have a quick question wondered if anyone could provide some suggestions please?! We have a setup of two registrars in EC2 (kamailio 4.4 running on debian VMs) which act in an active/active fashion with a number of devices behind NAT where we use the DMQ dmq_t_replicate function to send the registration details between each server and append the PATH parameter so the registrar the device registered with is always associated with the client device. This all works fine, the only issue is persistency should we have a failure of one of the registrars, be it process or server and recovering the correct PATH detail so the end device can still be reached. I have tried this with separate databases with each registrar, which works when using cache and db but I ideally want the same database for the two servers and not change the architecture at this time unless it is a necessary. I have also tested with dmq_usrloc but this again provides the usrlocation details of the notification servers in the group(not path), again which I understand but I wondered if there is anything else I could try or its just a question of in the event of a failure half the registrations will be effected until it recovers and they re-register again if I dont want to have separate databases (location tables) or I add more network elements into the flow, such as HA pair or load balancer but just want to make sure I have exhausted all options or consider the risk before I look to do this. Many thanks! Jon

2 2

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