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37519 discussions

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

2 11

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.

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sr-users