- sr-users - kamailio.org

limit call per duration
by António Silva 13 Sep '18

13 Sep '18

Hi, I'm new to kamailio, i want to be able to limit call per duration, how can i do it? i found module call_control, http://www.kamailio.org/docs/modules/5.1.x/modules/call_control.html, do you recommend it to use? anyone using it that would like to share the experience, like the values for cpu usage, is it realtime, support for multiple calls per account, other... Thanks!! -- Saludos / Regards / Cumprimentos António Silva

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How to use Geoip Module in Kemi
by Naveed Ur Rehman 13 Sep '18

13 Sep '18

Hi, I am trying to use geoip2 module in kemi using LUA, The issue is that when i try to access the $gip2(src=>cc) variable using lua, the kamailio instance killed and stops, this is my code snippet, kindly advise anyone how this can be sorted local sourceIP=KSR.pv.get("$si"); if KSR.geoip2.match("sourceIP","src") then local x= KSR.pv.get("$gip2(src=>cc)"); end thanks Engineer Naveed Ur Rehman 0923415652825 Computer Systems Engineering

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TOPOH module im REGISTER method.
by Mojtaba 13 Sep '18

13 Sep '18

Hello, I am used topoh module in kamailio, Why this module does not work (No change ) in Register request Method? I need to change UA' ip address in Via header in Register request before forwarded to my network? Does the topoh module used in this situation? I have not this issue in INVITE request, I just have this issue in Register request. Thanks. With Regards.Mojtaba -- --Mojtaba Esfandiari.S

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Crashed drouting
by Евгений Голей 12 Sep '18

12 Sep '18

Good evening. I have errors on the server (kamailio v 5.1.0) "kamailio [30529]: segfault at 0 ip 00007f9c00c29cfb sp 00007fffa9808310 error 4 in drouting.so [7f9c00c26000 + 39000]" which cause the service to stop. Although the drouting module has been used for a long time, this has never happened. This was done after adding the functions of the dispatcher module. What could be the reason ? Example error in system kamailio [20728]: segfault at 0 ip 00007f32c1c8fcfb sp 00007fff78430d00 error 4 in drouting.so [7f32c1c8c000 + 39000] in the logs of kamailio 018-09-08T22: 30: 29.0448 "ALERT: <core> [main.c: 746]: handle_sigs (): child process 10556 exited by a signal 11" 2018-09-08T22: 30: 29.0452 "ALERT: <core> [main.c: 749]: handle_sigs (): core was generated" 2018-09-08T22: 30: 29.0456 "INFO: <core> [main.c: 771]: handle_sigs (): terminating due to SIGCHLD" -- B.R. Evgeniy

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remove_hf("Bla") for all requests
by Sergiu Pojoga 12 Sep '18

12 Sep '18

Hi there, Say I need to remove_hf("User-Agent") for all requests, back and forth. So I add it at the top of *request_route *section. However, replies don't seem to be affected by it. Do I really need to arm a *t_on_reply route* for this simple purpose? "Problem" with that is that *route[RELAY]* already has some *onreply_route* block doing things like *NATMANAGE *for some methods and as far as I know - only one* onreply_route* can be armed for a request? route[RELAY] { ... if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } t_relay(); } onreply_route[MANAGE_REPLY] { ... route(NATMANAGE); } Any suggestions? Thanks.

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Re: [SR-Users] kamailio + rtpengine + custom filename
by Жан Базаров 12 Sep '18

12 Sep '18

Message-ID: <CABVi_EzZRF8nUqoBGcUtUeAV2pknn5S0urcmJFgHSSz6N1kJ5g(a)mail.gmail.com> Content-Type: text/plain; charset="UTF-8" Hello, You could use metadata to save your favourite filename on it. After recording is finished, replace it. The metadata file is created based on <CALL_ID> for each call. With Regards. Mojtaba On Mon, Sep 10, 2018 at 8:53 PM Жан Базаров <chiefkeeft(a)gmail.com> wrote: > > Hello! Filenames have the format <call_ID>-<random>-<SSRC>.<extension> > How i can change filename in start_recording() ??? Thx you. Can you help me with it task? Show me, please, an example. I want filename of the following format, for example: 'output-dir/yyyy/mm/dd/fromnumber/callid-tonumber-uniqueid.wav'.

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Missing ACC record in case of canceled call forking
by Marco Capetta 12 Sep '18

12 Sep '18

Hi All, I'm facing a strange problem of missing ACC record in case of parallel call forking. The scenario is the following: - A subscriber with 1 device registered - B subscriber with 2 device registered (B1 and B2) CASE 1: - A calls B - B1 and B2 start ringing - A hangups the call before B1 or B2 answers Kamailio generates an ACC record. CASE 2: - A calls B - B1 and B2 start ringing - B1 rejects the call sending back a 486 - B2 is still ringing - A hangups the call before B2 answers Kamailio DOESN'T generate an ACC record. We have Kamailio v5.1.4 with TM module enabled. ACC configuration is the following: modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 3) I increased debug level of TM and ACC modules and I added some debug lines as well. The difference between the two calls is that, after the CANCEL sent by A is processed by Kamailio, in CASE 1 I have the following lines: Aug 28 14:21:11 spce proxy[13188]: DEBUG: tm [t_hooks.c:258]: run_trans_callbacks_internal(): DBG: trans=0x7f5a88049198, callback type 512, id 0 entered Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_logic.c:670]: tmcb_func(): acc callback called for t(0x7f5a88049198) event type 512, reply code 487 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 0 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 1 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 2 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 3 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 4 Those debug lines are not printed for CASE 2. Is there any configuration I'm missing or is it a bug? Thank you very much for your support. Marco

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Call termination via dialog RPC dlg.end_dlg for the SIP-UA behind NAT
by Denys Pozniak 11 Sep '18

11 Sep '18

Hello! When calling RPC command dlg.end_dlg, BYE for leg behind NAT sends according to the Contact header not to the real ip:port. RPC ul.dump shows correct address in Received: sip:2.2.2.2:7488, but looks dialog does not know real address. How can I force sending to the real address? [root@proxy1]# kamcmd dlg.list { h_entry: 3753 h_id: 6741 call-id: 0_618427980(a)10.10.10.10 from_uri: sip:1234567890@1.1.1.1:5060 to_uri: sip:3333333333@1.1.1.1:5060 state: 2 start_ts: 0 init_ts: 1536671492 timeout: 0 lifetime: 7200 dflags: 512 sflags: 4096 iflags: 32 caller: { tag: 2750096738 contact: sip:1234567890@10.10.10.10:5060 cseq: 2 route_set: socket: udp:1.1.1.1:5060 } callee: { tag: contact: cseq: route_set: socket: } profiles: { } variables: { } } [root@proxy1]# kamcmd ul.dump Info: { AoR: 1234567890 HashID: 262653578 Contacts: { Contact: { Address: sip:1234567890@10.10.10.10:5060 Expires: 1269 Q: -1.000000 Call-ID: 0_213504837(a)10.10.10.10 CSeq: 8 User-Agent: Yealink SIP-T21P Received: sip: 2.2.2.2:7488 Path: [not set] State: CS_NEW Flags: 0 CFlags: 65600 Socket: udp: 1.1.1.1:5060 Methods: 16383 Ruid: uloc-5b97c358-6052-1 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1536674159 Last-Modified: 1536674159 } } } -- BR, Denys Pozniak

1 0

Remove branch from branch list
by Yuriy Gorlichenko 11 Sep '18

11 Sep '18

Hi I'm trying to manipulate with branches at the $branch variable. 8 actually need to remove unneeded branch if not passed some checks. For now I Just rewriting uri and dst_uri with some face address but it looks little bit dirty way. Is here any an other more clear way to do this?

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Record Routes
by Wilkins, Steve 11 Sep '18

11 Sep '18

Hello All, Can a Route: be removed?, if so, how? Thank you?

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Logging of messages generated by Kamailio itself
by Konstantin Polyakov 11 Sep '18

11 Sep '18

Hello, Could somebody help with one issue? In some cases Kamailio can send messages itself. For example, when fr_inv_timer is expired Kamailio sends 408 to the INVITE sender and CANCEL to the INVITE receiver. These two messages (408 and CANCEL) are not printed into the log and I cannot find a way to do that. There are also other scenarios when some Kamailio modules send SIP messages by its own. So is there way to print out into log all such messages? My goal is to get full message flow in the log per certain call-id. Thank you in advance. Best regards. Konstantin

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kamailio + rtpengine + custom filename
by Жан Базаров 11 Sep '18

11 Sep '18

Hello! Filenames have the format <call_ID>-<random>-<SSRC>.<extension> How i can change filename in start_recording() ???

2 1

tpops module
by volga629＠networklab.ca 11 Sep '18

11 Sep '18

Hello Everyone, I am using tcpops module for keepalive and on BYE when tcp keeplaive close is called produce this error in log. Sep 8 11:21:26 canlvprx01 /usr/sbin/kamailio[18933]: ERROR: <core> [core/sr_module.c:1781]: get_int_fparam(): Could not convert PV to int Sep 8 11:21:26 canlvprx01 /usr/sbin/kamailio[18933]: ERROR: tcpops [tcpops_mod.c:293]: w_tcp_keepalive_disable1(): invalid parameter 'con' (must be a number) kamailio-5.1.3-1.gitf0dce0c99.fc27.x86_64 if (is_method("BYE|CANCEL")) { $avp(bye_conid) = $conid; rtpengine_delete(); t_on_reply("4"); } onreply_route[4] { if(is_method("BYE") && t_check_status("200")) { tcp_keepalive_disable("$avp(bye_conid)"); } } volga629

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Simultaneous usage of t_on_reply() and t_on_failure()
by Володимир Іванець 10 Sep '18

10 Sep '18

Hello everyone, I wanted to ask those who know if Kamailio's behavior I'm facing is expected or I should make some improvements to the configuration. Kamailio version is 5.1.0. I have a route where RTPEngine parameters are being collected and *rtpengine_offer()* is called. After that *t_on_reply("REPLY_SIP_TO_SIP");* followed by the *t_on_failure("FAILURE_SIP_TO_SIP");* are used. The idea is to process all responces except 415 or 488 from UAC as usual in *onreply_route[REPLY_SIP_TO_SIP]* and use *failure_route[FAILURE_SIP_TO_SIP]* to update SDP with *rtpengine_offer()* if necessary. *onreply_route[REPLY_SIP_TO_SIP]* just goes to *exit;* if *$rs* equals 415 or 488. This works fine with Htek phone which sends 100, 180 and then 488. But I can not see *failure_route[FAILURE_SIP_TO_SIP]* execution for calls to Zoiper which replies with 100 and immediately 415. t_on_failure(failure_route) documentation says: "Sets failure routing block, to which control is passed after a transaction completed with a negative result but before sending a final reply." and to be honest I don't really understand how lacking of responce prevents failure_route from executing. Thanks a lot!

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Crashed kamailio drouting
by Евгений Голей 08 Sep '18

08 Sep '18

Good evening. I have errors on the server (kamailio v 5.1.0) "kamailio [30529]: segfault at 0 ip 00007f9c00c29cfb sp 00007fffa9808310 error 4 in drouting.so [7f9c00c26000 + 39000]" which cause the service to stop. Although the drouting module has been used for a long time, this has never happened. This was done after adding the functions of the dispatcher module. What could be the reason ? Example error in system kamailio [20728]: segfault at 0 ip 00007f32c1c8fcfb sp 00007fff78430d00 error 4 in drouting.so [7f32c1c8c000 + 39000] in the logs of kamailio 018-09-08T22: 30: 29.0448 "ALERT: <core> [main.c: 746]: handle_sigs (): child process 10556 exited by a signal 11" 2018-09-08T22: 30: 29.0452 "ALERT: <core> [main.c: 749]: handle_sigs (): core was generated" 2018-09-08T22: 30: 29.0456 "INFO: <core> [main.c: 771]: handle_sigs (): terminating due to SIGCHLD" -- BR Evgeniy

1 0

kamailio.cfg errors with this example from documentation
by Wilkins, Steve 07 Sep '18

07 Sep '18

$var(x) = "abcd"; $(var(x){s.substr,1,0}); Hello all, I took the example from the documentation, but Kamailio.cfg has errors with this example. Errors=> 0(14444) CRITICAL: <core> [core/cfg.y:3489]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 1-23: pvar with transformations in assignment left side 0(14444) CRITICAL: <core> [core/cfg.y:3492]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 24: syntax error 0(14444) CRITICAL: <core> [core/cfg.y:3492]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 827, column 24: bad command Thank you,

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Via header line missing
by Karlheinz Knapp 07 Sep '18

07 Sep '18

Hi, hm.... the only reason why I'm playing with kamailio is that it should be able to translate IPv6 from customer phones behind Fritz!Box (NAT) to IPv4 used by the provider (which is not able to handle IPv6). But it looks that he accept in general the packet with 2 Via lines (also with IPv6 as address) and send a normal reply. The second Via is not needed, because he have to send to the first Via. And if it really comes to trouble, I can remove it with kamailio :) If it is not possible to make a complete translation with kamailio, then I have to search again for an other solution. But still: why is this one Via line not transmitted on the IPv6 side? Only when I do some modifications it appears magically. That's not a normal behaviour in my opinion. If you or Henning can not reproduce this behaviour, then it's possible that I start TeamViewer and I can show this on our installation (only at normal working hours). Btw. I use sngrep to display all the messages, but it works stable and reliable. Best regards, i.A. Bernd

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Event: Kamailio Developers Meeting, Sep 27-28, 2018, in Dusseldorf, Germany
by Daniel-Constantin Mierla 07 Sep '18

07 Sep '18

Kamailio SIP Server project is organizing a meeting of its developers during September 27-28, 2018, hosted by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join. Please note we have a limited capacity of 20 seats in the meeting room. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: - https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- 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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kamailio with evapi crashing on high volume
by Jayesh Nambiar 07 Sep '18

07 Sep '18

Hello, I'm using kamailio 5.1.5 with evapi. I have a node.js connecting with kamailio evapi to which I send events and also consume events based on which I do the routing. I have 8 evapi workers defined in the config. The problem is that kamailio randomly crashes on high load. I'm assuming that it is related to the evapi module as rest of the config is pretty straight forward. I could get a core file and here's the core dump: [New LWP 14042] [Thread debugging using libthread_db enabled] Using host libthread_db library “/lib/x86_64-linux-gnu/libthread_db.so.1”. Core was generated by `/usr/local/kamailio/sbin/kamailio -P /var/run/siptrunk.pid -f /usr/local/carrie’. Program terminated with signal SIGABRT, Aborted. #0 0x00007f9995283428 in __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:54 54 ../sysdeps/unix/sysv/linux/raise.c: No such file or directory. (gdb) bt #0 0x00007f9995283428 in __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:54 #1 0x00007f999528502a in __GI_abort () at abort.c:89 #2 0x000000000041a029 in sig_alarm_abort (signo=14) at main.c:646 #3 <signal handler called> #4 0x00007f999534f497 in __libc_cleanup_routine (f=<optimized out>) at ../sysdeps/nptl/libc-lockP.h:291 #5 closelog () at ../misc/syslog.c:415 #6 0x0000000000000000 in ?? () Any help in this regards is would allow me to identify the reason of the crash. Thanks for the support. - Jayesh

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Kamailio with CGNAT
by Amar Tinawi 06 Sep '18

06 Sep '18

Hello Although my question it's not related to Kamailio directly, but i think someone could help in this strange behavior in our tests, we have some clients under private networks, where the provider of those networks implements CGNAT (NAT444) for it's customers to reach internet (Client Private IP) --->(Home router: Private to Private NAT)--->(FW with CGNAT)---------------------------->(Kamailio Proxy Public IP) Registration is performing fine, but making calls from this client had strange problem as follow: Client -------------INVITE--------------> Proxy Proxy -------------Trying---------------->Client Proxy--------------INVITE---------> Called Party The called party should send Ringing msg, but the massage get lost somewhere and didn't reach to the Proxy. Although all the IP's in sip massages are public i tried with STUN/TURN and NAThelper module solutions, but didn't help is there any way to slove CGNAT with SIP in Kamailio? Thanks in Advance

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dlg.end_dlg on RTP timeout
by Igor Olhovskiy 06 Sep '18

06 Sep '18

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

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call_stats
by David Villasmil 06 Sep '18

06 Sep '18

Hello all, Anyone got the table definition for call_stats? I'm getting db_mysql_submit_query(): driver error on query: Table 'kamailio.call_stats' doesn't exist (1146) And i can't find the table anywhere. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Via header line missing
by Karlheinz Knapp 06 Sep '18

06 Sep '18

Hi, of course kamailio is listening on IPv4 and IPv6. Register coming from IPv6 and goes out on IPv4: REGISTER sip:sip.ggew-net.de SIP/2.0 Via: SIP/2.0/UDP 217.113.176.155;branch=z9hG4bK98fc.314c428d232717f129932376c83e2c02.0 Via: SIP/2.0/UDP [2A05:E2C2:0:401:F40D:E8C5:5D85:2A95]:5060;received=2A05:E2C2:0:401:F40D:E8C5:5D85:2A95;branch=z9hG4bK001755af58b0e811a048bb9f393b8bfe;r rt=5060 From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de> Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER Contact: <sip:4962519448281@ [2A05:E2C2:0:401:F40D:E8C5:5D85:2A95]:5060>;+sip.instance="<urn:uuid:00E0F96C-5BAF-E811-A877-86B14B4F5249>" Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK Max-Forwards: 69 Allow-Events: org.3gpp.nwinitdereg User-Agent: SIPPER for PhonerLite Supported: replaces, from-change, gruu Expires: 900 Content-Length: 0 P-hint: outbound Answer on IPv4: SIP/2.0 401 Unauthorized Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07683-7956ba80-5065f1a44 Via: SIP/2.0/UDP 217.113.176.155;received=217.113.176.155;rport=5060;branch=z9hG4bK98fc.314c428d232717f129932376c83e2c02.0 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="7956b03f7acf80ba100f00e723284f58",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 The same answer on th IPv6 side after kamailio without something special: SIP/2.0 401 Unauthorized Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07683-7956ba80-5065f1a44 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="7956b03f7acf80ba100f00e723284f58",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 It contains no Via header!!! If I add my append_hf() append_hf("Via: SIP/2.0/UDP [2a05:e2c1:2::111];branch=$sel(via[1].branch);rport=5060\r\n", "To"); msg_apply_changes(); I get: SIP/2.0 401 Unauthorized Call-ID: 8080BCAE-58B0-E811-A046-BB9F393B8BFE@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=4236924539 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07683-7956ba80-5065f1a44 Via: SIP/2.0/UDP 217.113.176.155;received=217.113.176.155;rport=5060;branch=z9hG4bK98fc.314c428d232717f129932376c83e2c02.0 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="7956b03f7acf80ba100f00e723284f58",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 With the original Via header !!! If I add a second Via with append_hf() append_hf("Via: SIP/2.0/UDP [2a05:e2c2:0:401:3002:AA:754A:F475]:5060;rport\r\n", "To"); append_hf("Via: SIP/2.0/UDP [2a05:e2c1:2::111];branch=$sel(via[1].branch);rport=5060\r\n", "To"); msg_apply_changes(); I can see this additional line in the IPv6 packet too. SIP/2.0 401 Unauthorized Call-ID: 0038C332-5AB0-E811-ABC9-80F5AA6CB72C@2A05 :E2C2:0:401:F40D:E8C5:5D85:2A95 CSeq: 1 REGISTER From: "PhonerLite" <sip:4962519448281@sip.ggew-net.de>;tag=2618219059 Server: Cirpack/v4.76 (gw_sip) To: "PhonerLite" <sip:4962519448281@sip.ggew-net.de >;tag=00-07871-795b4386-0dc31d606 Via: SIP/2.0/UDP [2a05:e2c1:2::111];branch=z9hG4bK0ce8.b94e5d3389049004487a8551ac2c8c41.0;rport=5060 Via: SIP/2.0/UDP 217.113.176.155;received=217.113.176.155;rport=5060;branch=z9hG4bK0ce8.b94e5d3389049004487a8551ac2c8c41.0 WWW-Authenticate: Digest realm="sip.ggew-net.de ",nonce="795b3345266c3dd57a52e3237f4276dd",opaque="7950c9065c0c955",stale=false,algorithm=MD5 Content-Length: 0 But since I don't have the right branch, PhonerLite starts a new registration instead of reregister with username password. Best regards

2 1

null $au
by David Villasmil 06 Sep '18

06 Sep '18

Hello, in the block: if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } I'm trying to print out $au, but it's coming up NULL many times, i don't get why this is. If i remember correctly, the script goes through this block twice, once when we get the REGISTER with no credentials, and once when the REGISTER comes with credentials after the challenge... is this correct? i need to know how to catch $au when the authentication is successful... help? Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 2

Via header line missing
by Karlheinz Knapp 05 Sep '18

05 Sep '18

Hi, we want to use kamailio as a IPv6 to IPv4 proxy. IPv4 to IPv4 bridging with NAT works. If we come from an IPv6 address, the registration packet is send via IPv4 to the provider. He answers with 401 Unauthorized and one Via line in the header. On the IPv6 side was no Via line in the packet. So I tried to add one line by hand using an onreply_route and append_hf(), msg_apply_changes() To my suprise, this line does never appear, but the original Via line is now in the packet !!! If I add now additional Via header lines, they appear in the packet too. This looks realy like a bug to me. A solution is needed, because the phone does not respond to the 401 with user and password. It starts a new registration instead. Maybe I can fix it with my additional Via linem but I need the original branch ID (I think) How can I get this? Btw. we have this behaviour with 4.4.2 and 5.1.5 If I get no answer during the next 12 hours I'll try 3.3.7 Best regards

2 1

dispatcher local socket
by Евгений Голей 05 Sep '18

05 Sep '18

Hello! Kamilio version 5.1.0 Please tell me why the module dispatcher does not correctly determine the availability of destination gateways. The scheme is as follows: Kamailio (udp:XX.XX.XX.XX:5060) ----- Provider_1 -----> (sip:aa.bb.cc.dd:5060) Dest GW Kamailio (udp:YY.YY.YY.YY:5060) ------- Provider_2 -----/ In table dispatcher: setid | destination | flags | priority | attrs 17 | sip:aa.bb.cc.dd:5060 | 8 | 50 | socket=udp:XX.XX.XX.XX:5060 17 | sip:aa.bb.cc.dd:5060 | 8 | 40 | socket=udp:YY.YY.YY.YY:5060 That is, our server controls the availability of the remote gateway through two local sockets, in order to switch to the backup if it is not available through the primary Internet provider. But as I understand at the moment when determining the availability of a remote gateway, the outgoing socket does not play any role, that is, if the remote gateway is not accessible through the secondary operator (priority is less than the primary one), only the one that has priority above is always inactive. Are there any solutions to this problem? BR. Evgeniy

1 0

From URI attribute for dispatcher target
by Giacomo Vacca 05 Sep '18

05 Sep '18

Hi all, I was looking for a solution to set a dedicated From URI in OPTIONS requests to a specific dispatcher target. ds_ping_from ( https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatc…) applies globally, why I needed the granularity of a single target. With these changes to the dispatcher module I get what I need: https://github.com/kamailio/kamailio/commit/3f9c38be11416f7abe5c4448088ba66… I can just add an attribute to the others, e.g. instead of *1 sip:127.0.0.1:7070 <http://127.0.0.1:7070> 0 1* something like: *1 sip:127.0.0.1:7070 <http://127.0.0.1:7070> 0 1 ping_from=sip.custom.com <http://sip.custom.com>* When the ping_from attribute is not specified, the general ds_ping_from applies and is used. This would save me from a solution like intercepting the outgoing OPTIONS and manipulating the From header. Does this approach make sense? Are there other ways already available to achieve this? Thanks, Giacomo

1 0

SCA: sca_call_info_bye_handler: no in-use callee appearance for BYE
by Asgaroth 05 Sep '18

05 Sep '18

Hi All, I've just started playing with the SCA module and am encountering an issue whereby SCA cannot seem to find an in-use appearance for the BYE message. From my limited understanding of SCA, the module, in this case, appears to be trying to track the state of the call using callid as there are no call-info headers in the signalling for it to extract and use. I call sca_call_info_update() for initial invite, 100 response, 180 response, 200 response and again for the terminating bye. When I answer the call, at the 200 ok for the initial invite, I can see sca sending a notify to the subscribed devices indicating appearance state being active. I can verify this with kamcmd sca.all_appearances. However, when I hang up the call, when sca_call_info_update() is fired for the bye message it fails with the following error message: sca [sca_call_info.c:1587]: sca_call_info_bye_handler(): sca_call_info_bye_handler: no in-use callee appearance for BYE <aor> from <from uri>, call-ID 7989c223-2ba4-1237-1888-20474786734e I cannot seem to track down where this is failing for me, or where I may be missing something in my configuration. sca.all_appearances does appear to have the entry mentioned in the failure message, there must be something I'm missing somewhere. Is there anyone out there who is using the sca module and may have a moment to assist with this issue. I have a pcap trace that I can send privately. Any assistance would be greatly appreciated. Thanks

1 0

Kamailio v5.1.5 Released
by Daniel-Constantin Mierla 04 Sep '18

04 Sep '18

Hello, Kamailio SIP Server v5.1.5 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.5. For more details about version 5.1.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/08/kamailio-v5-1-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 1

Problem with reconnection to database server
by Minh Phan 04 Sep '18

04 Sep '18

Hi List, My Kamailio instance is disconnected from database server (Postgres) quite often (after running for some days) and it won't manage to reconnect. Any idea why this happens? Normally it try to reconnect when connection with the DB is lost, desn't it? Bellow are the related log messages. Best regards, Minh Phan Sep 4 11:31:20 : ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:559]: db_postgres_store_result(): invalid query, execution aborted Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:561]: db_postgres_store_result(): driver error: PGRES_FATAL_ERROR, Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:609]: db_postgres_insert(): unexpected result returned Sep 4 11:31:20 : ERROR: topos [tps_storage.c:599]: tps_db_insert_dialog(): failed to store message Sep 4 11:31:20 : ERROR: topos [tps_storage.c:412]: tps_storage_record(): failed to store Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: select rectime,a_callid,a_uuid,b_uuid,direction,x_via,x_vbranch,x_rr,y_rr,s_rr,x_uri,x_tag,s_method,s_cseq,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag from topos_t where x_vbranch='z9hG4bK66fa.3f007792000000000000000000000000.0' Sep 4 11:31:20 : ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Sep 4 11:31:20 : ERROR: topos [tps_storage.c:961]: tps_db_load_branch(): failed to query database Sep 4 11:31:20 : WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 kamailio[26608]: WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 kamailio[26608]: WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: insert into topos_d (rectime,a_callid,a_uuid,b_uuid,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag,a_rr,b_rr,s_rr,iflags,a_uri,b_uri,r_uri,a_srcaddr,b_srcaddr,s_method,s_cseq ) values ('2018-09-04 11:31:20','1ff8d1d7606f7416-26611(a)192.168.70.106 ','','','','','','','ae86e22a93159b5c6ac75234cdf26c1b-44e5','','','','',0,'','','','','','OPTIONS','10') Sep 4 11:31:20 kamailio[26608]: WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Sep 4 11:31:20 kamailio[26608]: ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: select rectime,a_callid,a_uuid,b_uuid,direction,x_via,x_vbranch,x_rr,y_rr,s_rr,x_uri,x_tag,s_method,s_cseq,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag from topos_t where x_vbranch='z9hG4bK86fa.424ded00000000000000000000000000.0' Sep 4 11:31:20 kamailio[26608]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:559]: db_postgres_store_result(): invalid query, execution aborted Sep 4 11:31:20 kamailio[26608]: ERROR: topos [tps_storage.c:961]: tps_db_load_branch(): failed to query database Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:561]: db_postgres_store_result(): driver error: PGRES_FATAL_ERROR, Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:609]: db_postgres_insert(): unexpected result returned Sep 4 11:31:20 : ERROR: topos [tps_storage.c:599]: tps_db_insert_dialog(): failed to store message Sep 4 11:31:20 : ERROR: topos [tps_storage.c:412]: tps_storage_record(): failed to store Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: select rectime,a_callid,a_uuid,b_uuid,direction,x_via,x_vbranch,x_rr,y_rr,s_rr,x_uri,x_tag,s_method,s_cseq,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag from topos_t where x_vbranch='z9hG4bK76fa.d33cc801000000000000000000000000.0' Sep 4 11:31:20 : ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Sep 4 11:31:20 : ERROR: topos [tps_storage.c:961]: tps_db_load_branch(): failed to query database Sep 4 11:31:20 : WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : WARNING: db_postgres [km_dbase.c:259]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [no connection to the server#012] Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:267]: db_postgres_submit_query(): 0x2b11c92b8b18 PQsendQuery Error: Query: insert into topos_d (rectime,a_callid,a_uuid,b_uuid,a_contact,b_contact,as_contact,bs_contact,a_tag,b_tag,a_rr,b_rr,s_rr,iflags,a_uri,b_uri,r_uri,a_srcaddr,b_srcaddr,s_method,s_cseq ) values ('2018-09-04 11:31:20','1ff8d1d7606f7417-26611(a)192.168.70.106 ','','','','','','','ae86e22a93159b5c6ac75234cdf26c1b-917f','','','','',0,'','','','','','OPTIONS','10') Sep 4 11:31:20 : ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:559]: db_postgres_store_result(): invalid query, execution aborted Sep 4 11:31:20 : ERROR: db_postgres [km_dbase.c:561]: db_postgres_store_result(): driver error: PGRES_FATAL_ERROR,

2 1

DMQ mem leak issues
by Rogelio Perez 04 Sep '18

04 Sep '18

Hello, We're running three instances of Kamailo v5.14 as registrars handling registrations from ~2000 SIP clients, with one instance being primary and the other two as backups. The three of them are using the dmq and dmq_usrloc modules to synchronize user locations, however after a couple of days of operation the two failover instances show memory leak behaviors, with mem usage assigned to the core taking all available resources. When this happens we've noticed that: - The shared memory used by the function "sip_msg_shm_clone" spikes (from 1kb to 1.5GB). - The shared memory used by the function "dmq:worker.c:job_queue_push" also increases, but not as much (from 1kb to 1MB) - DMQ request are not being answered (with a 200 OK) by the affected instance during this memory leak, which make us think that DMQ module becomes unresponsive. A few more notes: - The failover instances are doing nothing except receiving replicated contacts. - The shared memory grows at the same rate on both instances, but the critical behavior never happens at the same time. - We are allocating 1GB memory on startup to each instance. - We store the location DB in a psql DB and we load it at startup. - We didn't find any errors in syslog, even at debug level. Has anyone experienced a similar issue who can suggest a possible solution? Thanks, Rogelio Perez Telnyx

6 23

Ipv6_to_ipv4
by Karlheinz Knapp 04 Sep '18

04 Sep '18

i have tested the sample config from: https://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6 i found that i can register with v4 to kamailio and to the provider if i user ipv6 to kamailio and ipv4 to provider i can´t register. the packet from the provider contain a via entry which is missing from kamailio to my phone. Someone have a solution ?

1 0

event_route [tcp:closed] not executed
by Juha Heinanen 04 Sep '18

04 Sep '18

Based on comment: https://github.com/kamailio/kamailio/issues/171#issuecomment-418175840 I gave event_route [tcp:closed] a try: event_route [tcp:closed] { # Handle TCP connection close xlog("L_INFO", "$proto connection closed ($conid)\n"); } Nothing comes to syslog when I quit my baresip UA that according to wireshark closes its TCP connection after un-registration (fin - fin - ack). I have not set closed_event param, because according to README, it defaults to 1.

2 10

dispatcher probe
by volga629＠networklab.ca 04 Sep '18

04 Sep '18

Hello Everyone, Is possible access OPTIONS packet generated by dispatcher probe ? I am trying insert something like this. if(is_method("OPTIONS") && $hdr(User-Agent)=~"kamailio") { append_hf("X-Node-ID: $avp(node_id)\r\n"); } volga629

3 4

Can't connect to local MySQL server (13 "Permission denied")
by Ginhoux, Patrick 04 Sep '18

04 Sep '18

Hi, In a previous post ([SR-Users] Questions about changes between 5.0.1 and 5.0.7 <https://lists.kamailio.org/pipermail/sr-users/2018-August/102442.html> ), I reported problem to upgrade my kamailio from 5.0.1 to 5.0.7-7 level. With help from some people, it was finally fixed on my test server in my lab. Now I was expecting to do the same upgrade on a real server using these packages: kamailio-5.0.7-7.el7.x86_64.rpm kamailio-ims-5.0.7-7.el7.x86_64.rpm kamailio-mysql-5.0.7-7.el7.x86_64.rpm kamailio-perl-5.0.7-7.el7.x86_64.rpm kamailio-snmpstats-5.0.7-7.el7.x86_64.rpm But kamailio failed to connect the local MySQL server : Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: DEBUG: <core> [db.c:209]: db_bind_mod(): using db bind api for db_mysql Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: DEBUG: <core> [db.c:314]: db_do_init2(): connection 0x7fef779958f8 not found in pool Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: DEBUG: db_mysql [km_my_con.c:98]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@localhost/siprouter Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: db_mysql [km_my_con.c:129]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (13 "Permission denied") Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: htable [ht_db.c:92]: ht_db_open_con(): failed to connect to the database Aug 31 08:16:20 in52inmvt2router1-mvt-tpm-sipr /usr/sbin/kamailio[6365]: ERROR: <core> [core/sr_module.c:974]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) My "kamailio.cfg" contains the following command: modparam("htable", "db_url", "mysql://siprouter:password@localhost/siprouter") The service unit : "/usr/lib/systemd/system/kamailio.service" content is : [Unit] Description=Kamailio (OpenSER) - the Open Source SIP Server After=network.target mysql.service [Service] Type=simple User=kamailio Group=kamailio Environment='CFGFILE=/etc/kamailio/kamailio.cfg' Environment='SHM_MEMORY=1024' Environment='PKG_MEMORY=8' EnvironmentFile=-/etc/sysconfig/kamailio ExecStart=/usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY Restart=on-failure [Install] WantedBy=multi-user.target The "/etc/sysconfig/kamailio" content is : # # Kamailio startup options # # # Kamailio startup options # # Set to yes to enable kamailio, once configured properly. RUN_KAMAILIO=yes # User to run as USER=kamailio # Group to run as GROUP=kamailio # Amount of shared memory to allocate for the running Kamailio server (in Mb) SHM_MEMORY=1024 # Amount of per-process (package) memory to allocate for Kamailio (in Mb) PKG_MEMORY=8 Below the existing user in mysql : mysql:root:(none)> use mysql; Database changed mysql:root:mysql> select user,host from user; +-----------+-----------+ | user | host | +-----------+-----------+ | myadmin | % | | siprouter | % | | kamailio | localhost | | root | localhost | +-----------+-----------+ I have a symlink to locate the my.cnf : ln -s /myqdata/mysiprdb/data/my.cnf /etc/my.cnf And also this link to fix a htable reload problem : ln -s /myqdata/mysiprdb/data/mysiprdb.sock /var/lib/mysql/mysql.sock I don't find what the problem is this time. So if there is people that could help on the issue, thanks in advance. Cordialement. Patrick GINHOUX

3 6

load-balance to registered trunks?
by David Villasmil 03 Sep '18

03 Sep '18

Hello guys, I've got this scenario where some gateways will register on my kamailio and i need to send them calls on a load-balance fashion. is this possible? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

4 6

Dispatcher question
by Igor Olhovskiy 03 Sep '18

03 Sep '18

Hi! I’m having some issues with dispatcher and cannot get why it’s not working Idea is to use ds_next_dst right after ds_select_dst I’m having it like this modparam("dispatcher", "list_file", "/tmp/dispatcher.list") modparam("dispatcher", "flags", 2) # Enable dispather failover modparam("dispatcher", "xavp_dst", "_dispacher_dst_list_") … ds_select_dst(«1», «0»); … # Some extra check on $du here … ds_next_dst(); But get in logs WARNING: dispatcher [dispatch.c:2285]: ds_update_dst(): no xavp uri field in next destination record But contents of /tmp/dispatcher.list is 1 sip:172.28.0.12:5060 1 sip:172.28.0.19:5060 Can I somehow dump xavp_dst var to check if ds_select_dst is correct? kamailio 5.2.0-dev6 Regards, Igor

5 17

handle_ruri_alias() question or issue?
by Wilkins, Steve 02 Sep '18

02 Sep '18

Good Morning All, The following Incoming request came in => SIP Incoming Request: [[BYE sip:2406506175@20.20.20.20:5060;alias=10.10.10.5~55157~2 SIP/2.0 I called handle_ruri_alias(), and expected the destination and port to be set to 10.10.10.5:55157, but it was not. It was left 20.20.20.20:5060. Is my thinking correct or is there another way to set the destination and port to the alias? Thanks All!, -Steve

3 9

Having trouble associating response to message request
by Byran Carlock 31 Aug '18

31 Aug '18

Hello, I'll be the first to admit I'm a complete novice to Kamailio, but I'm learning due to necessity. I'm working on setting up Kamailio as a sip proxy (SBC?). Any help would be massively appreciated. The desired setup would be for Kamailio to as an external connection point to authenticate the connection and reverse proxy/NAT the rest of the traffic to the asterisk server, hiding internal topology as we go. Right now I'm using dispatcher with NAT and ALIAS to push connections to proxy@{PROXY IP} to proxy@{ASTERISK EXTERNAL IP}. I'm running into an issue where a "Message" is being sent, forwarded to the Asterisk server and responded to. Then Asterisk spawns a new call ID "To" the proxy rather than the original requestor which of course gets a 484 response. How can I somehow associate these two messages and complete the route? MESSAGE sip:proxy@{ASTERISK<sip:proxy@%7bASTERISK> EXTERNAL IP} SIP/2.0 Via: SIP/2.0/UDP {PROXY IP}:5060;branch=z9hG4bK0542.4d3d95c66b82f5825d9b90ba7ae88191.0 Via: SIP/2.0/UDP {CLIENT IP}:37916;received={CLIENT EXTERNAL IP};rport=37916;branch=z9hG4bKPj9b82293b-8ac4-4314-9cb8-d76716746e Max-Forwards: 69 From: <sip:{CLIENT IP}<sip:%7bCLIENT%20IP%7d>>;tag=8aabdd54-6312-4ba4-bbe4-e713d2cf55a7 To: <sip:proxy@{PROXY IP}<sip:proxy@%7bPROXY%20IP%7d>> Call-ID: 9d7bc193-55d9-479b-8c34-3663c3a8bf3b CSeq: 50215 MESSAGE Accept: text/plain, application/im-iscomposing+xml User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-AppServer-RequestType: Preconfig X-Digium-AppServer-Preconfig-Id: ca0175e891557c4e9d6f7314cdbf4044 X-Digium-Software-Version: 4.1.1 (2fccfc8f) X-Digium-AppServer-ID: mobileui0000000004 Content-Type: text/plain Content-Length: 182 P-hint: outbound Message Contents MESSAGE sip:{PROXY<sip:%7bPROXY> IP}:5060 SIP/2.0 Via: SIP/2.0/UDP {ASTERISK EXTERNAL IP}:5060;rport;branch=z9hG4bKPj6c1bdda3-2b78-4ae2-ac19-5ec2ca8b8928 From: <sip:proxy@{ASTERISK INTERNAL IP}<sip:proxy@%7bASTERISK%20INTERNAL%20IP%7d>>;tag=79cb0374-5b41-4550-9158-304a18bb1ee6 To: <sip:{PROXY IP}<sip:%7bPROXY%20IP%7d>> Contact: <sip:asterisk@{ASTERISK EXTERNAL IP}:5060<sip:asterisk@%7bASTERISK%20EXTERNAL%20IP%7d:5060>> Call-ID: 94c778a4-d3b3-42c1-ac26-25664d9d37bc CSeq: 4076 MESSAGE X-Digium-AppServer-ResponseType: PreconfigResponse X-Digium-AppServer-Password: ofPnRi4Pt/tGqdSYNn97tiO8ju/maPvAsG7ZeESGyUUWUgvF0xlMuhAUDVzJE866dv0+QeOowFTawa6UQJKQsQ== X-Digium-AppServer-Session: 471800327353850017 X-Digium-AppServer-Message-Salt: 85DA16A6C8F17E46 X-Digium-AppServer-Message-Sequence: 0 X-Digium-AppServer-ID: mobileui0000000004 Max-Forwards: 70 User-Agent: Asterisk PBX 13.1.0-6.7.rl.1516160751.a9f0643 Content-Type: text/plain Content-Length: 600 MESSAGE CONTENT

1 0

Having trouble associating response to message request
by Byran Carlock 31 Aug '18

31 Aug '18

Hello, I'll be the first to admit I'm a complete novice to Kamailio, but I'm learning due to necessity. I'm working on setting up Kamailio as a sip proxy (SBC?). The desired setup would be for Kamailio to as an external connection point to authenticate the connection and reverse proxy/NAT the rest of the traffic to the asterisk server, hiding internal topology as we go. Right now I'm using dispatcher with NAT and ALIAS to push connections to proxy@{PROXY IP} to proxy@{ASTERISK EXTERNAL IP}. I'm running into an issue where a "Message" is being sent, forwarded to the Asterisk server and responded to. Then Asterisk spawns a new call ID "To" the proxy rather than the original requestor which of course gets a 484 response. How can I somehow associate these two messages and complete the route? MESSAGE sip:proxy@{ASTERISK EXTERNAL IP} SIP/2.0 Via: SIP/2.0/UDP {PROXY IP}:5060;branch=z9hG4bK0542.4d3d95c66b82f5825d9b90ba7ae88191.0 Via: SIP/2.0/UDP {CLIENT IP}:37916;received={CLIENT EXTERNAL IP};rport=37916;branch=z9hG4bKPj9b82293b-8ac4-4314-9cb8-d76716746e Max-Forwards: 69 From: <sip:{CLIENT IP}>;tag=8aabdd54-6312-4ba4-bbe4-e713d2cf55a7 To: <sip:proxy@{PROXY IP}> Call-ID: 9d7bc193-55d9-479b-8c34-3663c3a8bf3b CSeq: 50215 MESSAGE Accept: text/plain, application/im-iscomposing+xml User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-User-Agent: Digium Android Softphone 4.1.1 (2fccfc8f) X-Digium-AppServer-RequestType: Preconfig X-Digium-AppServer-Preconfig-Id: ca0175e891557c4e9d6f7314cdbf4044 X-Digium-Software-Version: 4.1.1 (2fccfc8f) X-Digium-AppServer-ID: mobileui0000000004 Content-Type: text/plain Content-Length: 182 P-hint: outbound Message Contents MESSAGE sip:{PROXY IP}:5060 SIP/2.0 Via: SIP/2.0/UDP {ASTERISK EXTERNAL IP}:5060;rport;branch=z9hG4bKPj6c1bdda3-2b78-4ae2-ac19-5ec2ca8b8928 From: <sip:proxy@{ASTERISK INTERNAL IP}>;tag=79cb0374-5b41-4550-9158-304a18bb1ee6 To: <sip:{PROXY IP}> Contact: <sip:asterisk@{ASTERISK EXTERNAL IP}:5060> Call-ID: 94c778a4-d3b3-42c1-ac26-25664d9d37bc CSeq: 4076 MESSAGE X-Digium-AppServer-ResponseType: PreconfigResponse X-Digium-AppServer-Password: ofPnRi4Pt/tGqdSYNn97tiO8ju/maPvAsG7ZeESGyUUWUgvF0xlMuhAUDVzJE866dv0+QeOowFTawa6UQJKQsQ== X-Digium-AppServer-Session: 471800327353850017 X-Digium-AppServer-Message-Salt: 85DA16A6C8F17E46 X-Digium-AppServer-Message-Sequence: 0 X-Digium-AppServer-ID: mobileui0000000004 Max-Forwards: 70 User-Agent: Asterisk PBX 13.1.0-6.7.rl.1516160751.a9f0643 Content-Type: text/plain Content-Length: 600 MESSAGE CONTENT

1 0

CentOS_7/kamailio_v5.1.x-rpms/ [Errno -1] Package does not match intended download.
by Denys Pozniak 31 Aug '18

31 Aug '18

Hello! Please check repository for Kamailio v5.1.x for CentOS7: http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… I got error: kamailio-debuginfo-5.1.5-1.el7 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not SatisfiableB --:--:-- ETA Trying other mirror. kamailio-outbound-5.1.5-1.el7. FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.1.x-rpms clean metadata Trying other mirror. kamailio-5.1.5-1.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.1.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. kamailio-debuginfo-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. kamailio-outbound-5.1.5-1.el7.centos.x86_64: [Errno 256] No more mirrors to try. -- BR, Denys Pozniak

2 1

Re: [SR-Users] Final reply code is wrong in case of 302 and 486 busy
by Marco Capetta 31 Aug '18

31 Aug '18

Hi Daniel, did you have some time to have a look at the issue? Thank you very much Regards Marco

2 1

How to detect and Hangup a call on network lost at Caller or Callee?
by Hamid Hashmi 31 Aug '18

31 Aug '18

Dear Users Please advise how to detect a network lost at caller or callee side. I have tried timeout_socket<https://kamailio.org/docs/modules/5.1.x/modules/rtpproxy.html#rtpproxy.p.ti…> and keepalives <https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dialog.p.ka_int…> method but nothing worked for me. RTPproxy did not send timeout notification to Kamailio (Issue #1597<https://github.com/kamailio/kamailio/issues/1597>) while there is no options packet within a dialog. Is there any other method to cater such scenario. Regards Hamid R. Hashmi

5 11

Quick way to enable/disable all uac outbound registrations
by Joel Serrano 31 Aug '18

31 Aug '18

Hi, On an active/passive kamailio cluster, we use keepalived to move a virtual IP between the nodes and all signaling goes to that VIP. One of our DID providers requires us to register with every single DID we have from them in order to receive calls for such DID (NOTE: we only want to have the "active" kamailio sending outbound registrations to the carrier). My idea was to have keepalived enable all the outbound registrations when the current node changes to MASTER and disable them when it changes to BACKUP. Is there a quick "enable all" or "disable all" for uac module outbound registrations or do I have to "know in advanced" all of them and enable/disable one by one? In the docs we have: *8.4. uac.reg_disable* *Disable a remote registration record based on a filter. The command has two parameter: attribute and value. The attribute can be: l_uuid, l_username, r_username or auth_username. The value is what should be matcheg against the value of the attribute in the remote registration record.* I don't know if there is an "all" or maybe if a wildcard would work, but from the description it doesn't seem to. Any ideas? Do you guys think this is taking a correct approach? All suggestions are more than welcome!! Thanks, Joel.

3 7

Request Line alases
by Wilkins, Steve 30 Aug '18

30 Aug '18

Hello All, If the below Requests from 40.40.40.40(Asterisk) is received by 10.10.10.10(Kamailio) Request Line: INFO sip:3145553313@10.10.10.10:5060;alias=30.30.30.30~43508~2 SIP/2.0 Shouldn't Kamailio forward this request to 30.30.30.30? so that the '200 OK' can sent back to Asterisk for the response. (In this scenario) Would it be ok to just force a '200 OK' back to Asterisk from Kamailio? Thank you,

2 1

Delay
by Daniel W. Graham 30 Aug '18

30 Aug '18

Recently encountered a delay issue, packets arrive on system but Kamailio doesn't process right away. Here are a few details, any direction on what I can do to narrow down is appreciated. The receive queue normally sits at 0 or has random spike, when this issue occurs receive queue remains high like the below output. Version 5.1.5 ss -4 -n -l | grep 5060 udp UNCONN 321536 0 207.241.130.2:5060 *:* tcp LISTEN 0 128 207.241.130.2:5060 *:* -dan

2 2

IPv6 2 Ipv4
by Karlheinz Knapp 30 Aug '18

30 Aug '18

Hi i serach for a config to convert IPv6 to ipv4. My Provider give me only IPv4 and i have 4000 Customer which have ipv6. Anybody have expierience and a running config ?

3 2

Interesting issue with ACK back from 200 OK INVITE
by Wilkins, Steve 30 Aug '18

30 Aug '18

Hello All, I have had an issue for quite some time where I needed to open two UDP ports (via LISTEN) for some softphone UAC's to stay connected. Funny enough there is one Provider that works with the Single UDP. I think I finally see why I was needing two listening UDP ports. When the ACK is returning from Asterisk to Kamailio, the ACK is coming back as protocol TCP, not SIP, causing the call to drop. I finally realized this but I am not sure what would cause this and why one provider phone works but not the others. Once this is resolved, I may actually say...It works! Thank you, -Steve

1 0

Notify: stateless error reply used: I'm terribly sorry, server error occurred
by Soltanici Ilie 29 Aug '18

29 Aug '18

For some reason, Voicemail Notify messages coming from Asterisk Boxes - are not forwarded to the SIP Extensions by Kamailio. This is the SIP Packet coming to the Kamailio: 2018/08/28 10:59:19.165644 1.1.1.1:5060 -> 2.2.2.2:5060 NOTIFY sip:11@3.3.3.3:21755;rinstance=e6369e13becab3ef;transport=tcp;alias=3.3.3.3~21755~2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK5f628579;rport Max-Forwards: 70 Route: <sip:2.2.2.2:5060;transport=tcp;ftag=0472e823;lr=on;nat=yes> From: "asterisk" <sip:asterisk@1.1.1.1>;tag=as1833c329 To: <sip:11@3.3.3.3:21755;rinstance=e6369e13becab3ef;transport=tcp;alias=3.3.3.3~21755~2>;tag=0472e823 Contact: <sip:asterisk@1.1.1.1:5060> Call-ID: 93183ZjE3YmI0OGYxMDNiMzhlNzVjZjRjNTc2NTMwZGNmYzk CSeq: 103 NOTIFY User-Agent: NUACOM Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@1.1.1.1 Voice-Message: 0/0 (0/0) And this is the message which I'm receiving in the Kamailio logs: [RELAY] Error when forwarding Request ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) This is the Relay route: # Forwarding Statefully if ( ! t_relay()) { xlogl( "L_ERR" , "[RELAY] Error when forwarding Request \n " ); sl_reply_error(); } exit ;

2 2

Info: Cluecon Weekly about Kamailio, today at 17:00GMT
by Daniel-Constantin Mierla 29 Aug '18

29 Aug '18

Hello, if you haven't seen the news article on kamailio.org, today at 17:00GMT I will be part of the Cluecon Weekly online video conference to present about what's new lately in Kamailio, plans for the future as well as address questions about using Kamailio together with FreeSwitch. Anyone can join via audio or video calls, more details about at: * https://www.kamailio.org/w/2018/08/kamailio-on-cluecon-weekly-aug-29-2018/ * https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere… Cheers, Daniel -- 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

1 1

Kamailio consuming UDP packets very slowly leading to high RECV - Q
by sagar malam 29 Aug '18

29 Aug '18

Hello, I am using Kamailio as a SIP proxy.So it receives SIP packet from internet and forwards it to FS servers in local network.When i execute "ss" command i see very high value in RECV-Q column.I THINK IT IS NOT NORMAL.PLEASE CORRECT ME IF I AM WRONG. ==================================================== [root@fep-1 proc]# ss -u -a -n -e | grep 5060 UNCONN 0 0 10.50.8.1:5060 *:* ino:6831630 sk:fd <-> UNCONN 0 0 10.50.7.18:5060 *:* ino:6831629 sk:fe <-> UNCONN *1183104* 0 10.50.7.254:5060 *:* ino:6831627 sk:ff <-> UNCONN *84864* 0 2607:f900:1:3::254:5060 :::* ino:6831628 sk:100 v6only:1 <-> ====================================================== Initially i thought that there is something in script which must be causing kamailio to process UDP request slower but i faced same issue with a very simple script where i simply reply with stateless 200 OK for each sip request : request_route { sl_send_reply("200","OK");exit; $avp(uuid) = $rm + "-" + $ci; ........ ......... ......... ....... Server configuration : OS : CENTOS 7 Kernel : 4.16 CPU : 5 X Intel(R) Xeon(R) CPU E5-2680 v2 @ 2.80GHz RAM : 32 GB Please help me debugging this issue.Thanks in advance

4 9

Using SEMS in IMS-ZTE Network
by Mojtaba 28 Aug '18

28 Aug '18

Hello, I am working with IMS (ZTE vendor), I want to use SBC in front of IMS network for some security policy (Hide Topology), I used SEMS as SBC in this regards, (like what we have in IMS/SEMS in kamailio by Mr Carsten Bock). When the Sebscriber want to register, The Register request received by sems and the forward to IMS(ZTE), But the "503 Service Unavailable" is received from IMS. I enable "enable_reg_caching=yes" in register.sbcprofile''s configuration file in sems. If i use kamailio, instead of SEMS, The Registration request is done successfully.In Kamailio i just forward the Register request to IMS.like this: route { if (is_method("REGISTER")) { $du = "sip:172.23.1.1:5080"; route(RELAY); } } Let me know do the issue is from myself or from IMS(ZTE)? In the fillowing, I paste all SIP Register signalling. First in kamailio, second with SEMS. ================Kamailio================== 172.23.1.252.4080 > 172.23.1.1.5080: [bad udp cksum 0x5e58 -> 0x4193!] UDP, length 787 E../....@.]...............^XREGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bK018b.e091260b1b7395975a2ff1be0b9bec60.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-7c208b9d587de818-1---d8754z- Max-Forwards: 69 Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=e664e178fc12691e;transport=UDP> From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 0 08:20:53.114198 IP (tos 0x0, ttl 255, id 25839, offset 0, flags [none], proto UDP (17), length 685) 172.23.1.1.5080 > 172.23.1.252.4080: [udp sum ok] UDP, length 657 E...d......$..............l.SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bK018b.e091260b1b7395975a2ff1be0b9bec60.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-7c208b9d587de818-1---d8754z- To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipTR9m3KdZ*3-5-16648*daei.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 1 REGISTER Content-Length: 0 WWW-Authenticate: Digest realm="r-kh.com",domain="sip:scscf1.r-kh.com",nonce="f42b36e39bdbb33b51599924b304445a",opaque="aW1zLmNvbS5jbg==",stale=TRUE,algorithm=MD5,qop="auth" 08:20:53.120142 IP (tos 0x10, ttl 64, id 48874, offset 0, flags [none], proto UDP (17), length 1137) 172.23.1.252.4080 > 172.23.1.1.5080: [bad udp cksum 0x5f9a -> 0xcc62!] UDP, length 1109 E..q....@.\V.............]_.REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bKd08b.4d2858f41d496b471c01a28e76190390.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-a4d19146375d56f6-1---d8754z- Max-Forwards: 69 Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=e664e178fc12691e;transport=UDP> To: <sip:+985137054518@r-kh.com;transport=UDP> From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Authorization: Digest username="+985137054518(a)r-kh.com",realm="r-kh.com",nonce="f42b36e39bdbb33b51599924b304445a",uri="sip:r-kh.com;transport=UDP",response="9469852a06c01efeabf3a7a5dd1a9f3b",cnonce="79d36600575648a1cc7221d8e1b5d53d",nc=00000001,qop=auth,algorithm=MD5,opaque="aW1zLmNvbS5jbg==" Allow-Events: presence, kpml Content-Length: 0 08:20:53.203871 IP (tos 0x0, ttl 255, id 25842, offset 0, flags [none], proto UDP (17), length 1118) 172.23.1.1.5080 > 172.23.1.252.4080: [udp sum ok] UDP, length 1090 E..^d......p.............J.5SIP/2.0 200 OK Via: SIP/2.0/UDP 172.23.1.252:4080;branch=z9hG4bKd08b.4d2858f41d496b471c01a28e76190390.0 Via: SIP/2.0/UDP 192.168.122.1:64407;branch=z9hG4bK-d8754z-a4d19146375d56f6-1---d8754z- To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipjRydzgMa*3-5-16648*daej.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5aef0541 Call-ID: ZWYxMzUzNTQzN2I2MTRiMjE2NWZjMzUzMDFmMDhkOWY. CSeq: 2 REGISTER Contact: <sip:+985137054518@192.168.122.1:64407;transport=UDP;rinstance=e664e178fc12691e>;expires=1800 Contact: <sip:ZTEAGCF000f425704be720000@agcf2.r-kh.com:5079;appid=2-7;UserRegInfo=002-02-00-0b80-04-172.30.48.143;BACaddr=172.30.48.143;BACport=2944>;expires=1800;+sip.instance="<urn:uuid:172.30.35.41-000>";reg-id=1 P-Associated-URI: <tel:+985137054518> P-Associated-URI: <sip:+985137054518@r-kh.com> Date: Sat, 25 Aug 2018 12:25:17 GMT Authentication-Info: nextnonce="00cfbd64b136c12a4e27b67fd28d19fb",qop=auth,rspauth="91ca956c5eb6ee9f1934e2e770750077",cnonce="79d36600575648a1cc7221d8e1b5d53d",nc=00000001 Content-Length: 0 ============================SEMS======================== 172.23.1.252.5080 > 172.23.1.1.5080: [bad udp cksum 0x5e16 -> 0xe11e!] UDP, length 721 E...C.@.@.................^. REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:5080;branch=z9hG4bK~0uk0ahN;rport From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=34384541-5B814A44000AE296-4B5F5700 To: <sip:+985137054518@r-kh.com;transport=UDP> CSeq: 10 REGISTER Call-ID: 37B52BC0-5B814A44000AE592-4B5F5700 Route: <sip:172.23.1.1:5080;lr> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=868560f8328d39e9;transport=UDP>;expires=3600 Content-Length: 0 08:23:32.790414 IP (tos 0x0, ttl 255, id 35790, offset 0, flags [none], proto UDP (17), length 593) 172.23.1.1.5080 > 172.23.1.252.5080: [udp sum ok] UDP, length 565 E..Q.....................=..SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.23.1.252:5080;rport=5080;branch=z9hG4bK~0uk0ahN To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipDEmH2AWT*3-5-16648*egei.3 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=34384541-5B814A44000AE296-4B5F5700 Call-ID: 37B52BC0-5B814A44000AE592-4B5F5700 CSeq: 10 REGISTER Content-Length: 0 WWW-Authenticate: Digest realm="r-kh.com",domain="sip:scscf1.r-kh.com",nonce="179797f920a09d65b5641e5e87df0d8c",opaque="aW1zLmNvbS5jbg==",stale=TRUE,algorithm=MD5,qop="auth" 08:23:32.855067 IP (tos 0x0, ttl 64, id 17364, offset 0, flags [DF], proto UDP (17), length 1071) 172.23.1.252.5080 > 172.23.1.1.5080: [bad udp cksum 0x5f58 -> 0xf553!] UDP, length 1043 E../C.@.@................._X REGISTER sip:r-kh.com SIP/2.0 Via: SIP/2.0/UDP 172.23.1.252:5080;branch=z9hG4bKeAfKkaZa;rport From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5A64C27D-5B814A44000CD034-4B6F6700 To: <sip:+985137054518@r-kh.com;transport=UDP> CSeq: 10 REGISTER Call-ID: 5380C97E-5B814A44000CD2A5-4B6F6700 Route: <sip:172.23.1.1:5080;lr> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Authorization: Digest username="+985137054518(a)r-kh.com",realm="r-kh.com",nonce="179797f920a09d65b5641e5e87df0d8c",uri="sip:r-kh.com;transport=UDP",response="017efb986e17506a82bb01e8eb136b90",cnonce="3764fd5eb2ea9539d61ec14196a2bf00",nc=00000001,qop=auth,algorithm=MD5,opaque="aW1zLmNvbS5jbg==" Allow-Events: presence, kpml Contact: <sip:+985137054518@192.168.122.1:64407;rinstance=868560f8328d39e9;transport=UDP>;expires=3600 Content-Length: 0 08:23:32.860417 IP (tos 0x0, ttl 255, id 35804, offset 0, flags [none], proto UDP (17), length 508) 172.23.1.1.5080 > 172.23.1.252.5080: [udp sum ok] UDP, length 480 E...........................SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 172.23.1.252:5080;rport=5080;branch=z9hG4bKeAfKkaZa To: <sip:+985137054518@r-kh.com;transport=UDP>;tag=ztesipxxsD3FR2VG8e9zxR_1dx*1-1-16648*ihad.1 From: <sip:+985137054518@r-kh.com;transport=UDP>;tag=5A64C27D-5B814A44000CD034-4B6F6700 Call-ID: 5380C97E-5B814A44000CD2A5-4B6F6700 CSeq: 10 REGISTER X-ZTE-Cause: "CSCF-1.1.BC004016-BC004288-0-0.pcscf1.r-kh.com" Retry-After: 30 Content-Length: 0 -- --Mojtaba Esfandiari.S

2 3

Parmas in lcr_gw
by Ashutosh chaubey 28 Aug '18

28 Aug '18

Hi, How to give username and password for the gateway in params column in lcr_gw? Regards, Ashutosh

2 1

regular expression in kamailio
by Wilkins, Steve 25 Aug '18

25 Aug '18

Hi All, I am trying get the 10digit number called in on using the following $var(caller) = $(fu{re.subst,[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]}); I thought I could use a regular expression for the expression in re.subst,expression Thank you, -Steve

2 8

Listner problems
by Wilkins, Steve 25 Aug '18

25 Aug '18

Hi Joel, Actually, I have a strange issue. If I have a single UDP IP:Port listening, most Providers/Phones have two-way Audio/Video, however, with some Providers/Phones, I need to have two UDP IP:Port listners in order to get two-way Audio/Video. Here is the strange thing. If I enable two, the Providers/Phones that worked when I had a single listener only have one-way Audi/Video. I use Kamailio 5.1 and Asterisk 15.3 (pjsip). This behavior is so strange, Thank you, -Steve

2 2

add listen dynamically
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hi All, Is it possible to add or remove "listen" dynamically? Thank you, -Steve

2 2

Struggling with RTPEngine
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hi Florian, Thank you for your response. I checked and direct-media is off. Just as a recap, here is where I am I just can't get rtpengine to work. I have tried multiple configurations, but to no avail. Note that calls work good if rtpengine is disabled. Here is my setup => Public IP: 20.20.20.20 Private IP 10.10.10.10 flow => webrtc client <-> kamailio+rtpengine <-> asterisk <-> kamailio <-> legacy sip phone rtpenngine startup (I have tried a few different startups) => rtpengine --interface=int/10.10.10.10 --interface=ext/10.10.10.10\!20.20.20.20 --listen-ng=127.0.0.1:12221 --pidfile=/var/run/rtpengine --dtls-passive -f -m 10000 -M 20000 -E kamailio => Invites: rtpengine_manage("trust-address replace-origin replace-session-connection direction=ext direction=int ICE=remove RTP/AVP"); Reply's: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); I have tried direction ext ext; and many other combinations, each producing its own incorrect behavior. Thanks again, Steve

1 0

Struggling with RTPProxy and RTPEngine
by Wilkins, Steve 24 Aug '18

24 Aug '18

Hello all, I can't seem to get either RTPProxy or RTPEngine to work correctly. I have decided to concentrate on RTPEngine because I have read that it works the best. I am using Asterisk with Kamailio in front. When I make calls, I see RTPEngine being hit but I continually get the error "Call-ID not found", calls work but media traffic doesn't go through RTPEngine. I have tried many combination of the flags but I seem to always get the same error. When I look at the tcpdump log, I see that the media offer is not on the RTPEngine port. Is there a common mis-configuration error that can cause this? I know I can send all my logs and configurations but I really want to try and resolve this as a learning experience. Thanks all, -Steve

8 17

Send a BYE for the call
by Nicolas Breuer 23 Aug '18

23 Aug '18

Hello, Any ideas how to cut a call in progress ? How to send a forced "BYE" to all parties ? Thanks

2 2

Kamailio 5.1.4 crashes
by Abdulaziz Alghosh 23 Aug '18

23 Aug '18

Hi Everyone, I am trying to start kamailio 5.1.4 with two own modules. After /etc/init.d/kamailio start , I am getting one error of the dispatcher module and bunch of warrnings because of missed reply and failure routes. The first error before crashing is: ERROR: dispatcher [dispatcher.c:814]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Later on, when I am trying to register a user (from a soft phone) i am getting the following: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) Aug 22 10:02:52 sgw1a /sbin/kamailio[6960]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 20 Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ALERT: <core> [main.c:746]: handle_sigs(): child process 6942 exited by a signal 11 Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ALERT: <core> [main.c:749]: handle_sigs(): core was not generated Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: INFO: <core> [main.c:771]: handle_sigs(): terminating due to SIGCHLD Aug 22 10:02:52 sgw1a /sbin/kamailio[6959]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6958]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6957]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6956]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6955]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6954]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6953]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6950]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6941]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6944]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6948]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6951]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6946]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6952]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6943]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6949]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6947]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6938]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6937]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6940]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6939]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a kernel: [296301.271457] kamailio[6942]: segfault at 0 ip (null) sp 00007fffb3ddf368 error 14 in kamailio[400000+442000] Aug 22 10:02:52 sgw1a /sbin/kamailio[6945]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6960]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6936]: INFO: <core> [main.c:826]: sig_usr(): signal 15 received Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ERROR: jsonrpcs [jsonrpcs_fifo.c:594]: jsonrpc_fifo_destroy(): cannot delete the fifo (/tmp/kamailio_jsonrpc.fifo): Operation not permitted Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: ERROR: jsonrpcs [jsonrpcs_sock.c:511]: jsonrpc_dgram_destroy(): cannot delete the socket (/tmp/kamailio_rpc.sock): Operation not permitted Aug 22 10:02:52 sgw1a /sbin/kamailio[6935]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Regarding to the first error of the dispatcher module i could not find out which calibration i can follow with its parameters to get rid of this error. Regarding the jasonrpcs and the last two erros, here are my configuration in kamailio.cfg: #/* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_jsonrpc.fifo") modparam("jsonrpcs", "fifo_mode", 0755) modparam("jsonrpcs", "fifo_group", "kamailio") modparam("jsonrpcs", "fifo_user", "kamailio") #/* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") modparam("jsonrpcs", "dgram_mode", 0755) modparam("jsonrpcs", "dgram_user", "kamailio") modparam("jsonrpcs", "dgram_group", "kamailio") I think the root cause is even beyond the dispatcher and jsonrpcs modules setting. Please check the attached where you can find my complete log file after starting Kamailio. Best regards Abdulaziz

2 4

Dispatcher - Force one end point active
by Marrold 23 Aug '18

23 Aug '18

Hi, We are currently using the dispatcher module and probing end points with OPTIONS, and making them active / inactive depending on the response: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;code=404") modparam("dispatcher", "ds_ping_from", "[REDACTED]") modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_probing_mode", 1) One of our suppliers endpoints doesn't respond to SIP OPTIONS, is it possible to force a single endpoint as active, but continue probing the rest? I've tried various settings - both globally and flags for the individual end point, but I couldn't seem to find the correct combination. Thanks Matthew

2 2

Re: [SR-Users] Fail to reload htable
by Henning Westerholt 23 Aug '18

23 Aug '18

Am Montag, 20. August 2018, 20:58:43 CEST schrieb Soltanici Ilie: > In the logs, I see these errors: > > ERROR: <core> [db.c:463]: db_use_table(): invalid parameter value > Aug 20 19:56:41 kamailio[12716]: ERROR: <core> [db.c:463]: db_use_table(): > invalid parameter value Aug 20 19:56:41 kamailio[12716]: ERROR: htable > [ht_db.c:211]: ht_db_load_table(): failed to use_table Hi, please keep the list in CC. This error is related to a database issue. Please double check if there are any issues during startup in the log. Do you see any other error there? A similar (not 100% identical) issue was discussed recently here: https://lists.kamailio.org/pipermail/sr-users/2018-August/102393.html Maybe this helps you as well. Best regards, Henning -- Henning Westerholt https://skalatan.de/blog/

4 7

Re: [SR-Users] Struggling with RTPProxy and RTPEngine
by Wilkins, Steve 23 Aug '18

23 Aug '18

Good morning Daniel, If RTPEngine is on the same server as Kamailio (Asterisk being on another server), and RTP traffic is sent to and from RTPEngine, then the provider only needs to whitelist one IP-Address. I thought with RTPEngine that all RTP traffic would go through it and then it would pass it on to the correct destination. Is this correct? Thank you -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Daniel Tryba Sent: Thursday, August 23, 2018 4:36 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine MITRE WARNING: Do not open unexpected password-protected attachments. >>>Email originates from a non-MITRE system. Use caution.<<< On Wed, Aug 22, 2018 at 05:05:02PM +0000, Wilkins, Steve wrote: > The SIP traffic is working this way for me but I still see RTP traffic going directly from Asterisk to the UAC, which means they need to whitelist asterisk IP. Am I missing something? In what sense do they need whitelisting? In a common NATed solution where is no white/blacklist needed. UA gets RTP endpoints from SDP, starts sending packets to ip/port and the destination will send back packets to the source ip/port, the router/firewall will just send this to the actual UA. I have yet to find an UA that cares about where the RTP stream is coming from with regards to the SIP traffic.

2 1

SIP and RTP Proxy for multiple Servers
by sewlist＠gmail.com 23 Aug '18

23 Aug '18

Im very new to kamailio, we have some ptime issues which causes one way audio due to varies PBX vendors connecting to our class 5 softswitches I would like to use Kamailio as the SIP/RTP proxy in front of our class 5 soft switches I followed the example of freeswitch and kamailio. However the config is still hard for me below setup works, however RTP not being proxied, Looks like its its not, because im not using NAT, so RTP goes direct https://freeswitch.org/confluence/display/FREESWITCH/Kamailio+basic+setup+a… I would like to to ensure RTP proxy is always used(possible re-framing ptime) and we have multiple soft switches Is it possible to setup kamailio.cfg so my customers can point the endpoint to kamailio and domain names contains the server we want to use to register with? eg, softswitch 1 = 1.1.1.2 softswitch 2 = 1.1.1.3 softswitch 3 = 1.1.1.4 etc etc my current config #!KAMAILIO #!define IPADDRESS "1.1.1.1" #!define SWITCH_IPADDRESS "1.1.1.2" #!define FLAG_FROM_SWITCH 1 #!define FLAG_FROM_USER 2 # ------------------ module loading ---------------------------------- loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "path.so" loadmodule "rtpproxy.so" # ----------------- setting module-specific parameters --------------- modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") # ------------------------- request routing logic ------------------- #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif # main routing logic route { # per request initial checks route(SANITY_CHECK); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } exit; } route(CHECK_SOURCE_IP); ################################## ### HANDLE SEQUENTIAL REQUESTS ### route(WITHINDLG); ############################### ### HANDLE INITIAL REQUESTS ### t_check_trans(); if (is_method("INVITE|REFER")) { record_route(); } if (is_method("REGISTER")) { add_path(); } if (isflagset(FLAG_FROM_SWITCH)) { # don't send INVITE from SWITCH back to SWITCH, set reply route to handle NAT and forward them along t_on_reply("EXTERNAL_REPLY"); } else { # set destination to your SWITCH $du = "sip:1.1.1.2:5060"; } route(RELAY); } route[SANITY_CHECK] { if (!sanity_check()) { #xlog("L_WARN", "$ci|end|message is insane"); exit; } if (!mf_process_maxfwd_header("10")) { #xlog("L_WARN", "$ci|end|too much hops, not enough barley"); send_reply("483", "Too Many Hops"); exit; } if ($ua == "friendly-scanner" || $ua == "sundayddr" || $ua =~ "sipcli" ) { #xlog("L_WARN", "$ci|end|dropping message with user-agent $ua"); exit; } if ($si == IPADDRESS) { #xlog("L_WARN", "$ci|end|dropping message"); exit; } } route[CHECK_SOURCE_IP] { if ($si == SWITCH_IPADDRESS) { setflag(FLAG_FROM_SWITCH); } else { setflag(FLAG_FROM_USER); } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(RELAY); } else { if (is_method("NOTIFY")) { route(RELAY); } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard #xlog("ACK without matching transaction ... ignore and discard"); exit; } } sl_send_reply("404","Not here"); } exit; } } onreply_route[EXTERNAL_REPLY] { route(NAT_TEST_AND_CORRECT); } route[NAT_TEST_AND_CORRECT] { if (nat_uac_test("3")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } force_rport(); } if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); } } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { #setbflag(FLB_NATB); route(RTPPROXY); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif route(RTPPROXY); /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; # if (!t_relay()) { # sl_reply_error(); # } # exit; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { #unforce_rtp_proxy(); force_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; }

2 1

RTPEngine wow's
by Wilkins, Steve 22 Aug '18

22 Aug '18

Hello all, I am still trying to get RTPEngine to work. At this point, when I make a call, I do see the offer in the rtpengine log, I also see the update of the SDP (c) changing it to 20.20.20.20. However, after that, there are no logs other than a bunch of "timer run time = 0.nnnnnnn sec", and the delete when I hang up the call. There is no RTP traffic what so ever in the rtpengine log (however, the call is completely successful, other than the RTP traffic not going through RTPEngine). Here is a breakdown of my configs (I borrowed this format from Maxim Fedorov because I thought it was very concise) Starting rtpengine => rtpengine --interface=internal/21.21.21.21--interface=external/21.21.21.21\!20.20.20.20 --listen-ng=127.0.0.1:12221 --dtls-passive -f -m 10000 -M 20000 -E -L 7 --log-facility=local1 RTPEngine Public IP: 20.20.20.20 RTPEngine Private IP: 21.21.21.21 kamailio.cfg modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:12221") listen=tcp: 21.21.21.21:5060 advertise 20.20.20.20:5060 listen=udp: 21.21.21.21:5060 advertise 20.20.20.20:5060 rtpengine_manage call rtpengine_manage("trust-address replace-origin replace-session-connection external external media-address=20.20.20.20"); (I have placed this rtpengine_manage call at the beginning of all routes). I have tried many variations of rtpengine_manage parameters, and placement of the call itself, but I never see RTP traffic in the rtpengine log. Thank you to all who have looked at this, -Steve

1 0

Error at startup
by Nicolas Breuer 22 Aug '18

22 Aug '18

Hello, How can i clear this error ? Aug 22 13:05:56 proxy1 kamailio: ERROR: <core> [core/pvapi.c:228]: pv_table_add(): pvar [rtpstat] already exists Aug 22 13:05:56 proxy1 kamailio: ERROR: <core> [core/pvapi.c:513]: register_pvars_mod(): failed to register pseudo-variable <rtpstat> for module rtpproxy

2 2

Releasing Kamailio v5.1.5
by Daniel-Constantin Mierla 22 Aug '18

22 Aug '18

Hello, with the summer holidays in the northern hemisphere approaching the end, I am considering to release v5.1.5 out of Kamailio git branch 5.1 -- there were several fixes done since 5.1.4 and it is time to package another minor release. The propose date is next week on Wednesday, August 22, or slightly after in case there are unexpected events delays it. Should there be any issues you are aware of and not yet reported on the bug tracker, do it asap in order to have a chance to be fixed: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

1 2

Encoding mechanism in UAC Module
by Duarte Rocha 21 Aug '18

21 Aug '18

Greetings, Is there a way to choose the encoding algorithm that builds the vsf and vst parameters in the Record-Route? What is the default method that Kamailio uses for this ? Thanks in advance, Duarte Rocha

2 1

invalid version 8 for table aliases found, expected 9
by Abdulaziz Alghosh 21 Aug '18

21 Aug '18

Hi everyone, i am trying to start kamailio 5.1.4 after the migration from Kamailio 3.0.3 but getting newly the following error: ERROR: <core> [db.c:450]: db_check_table_version(): invalid version 8 for table aliases found, expected 9 (check table structure and table "version") Allegedlly, this error is because of a wrong version of "aliases" table. I followed the Upgrade process starting from 3.0.3 till 5.1.x and I altered the structure of my aliase table accordinglly. The last new version of aliases is 8 as it is described at the upgrade from 4.2.x to 4.3.0 Even though, the error here says that the newest version is 9: May someone help me to find out if the aliases table's structure was renewed in between ?? Is there any modification in the data types? Thanks in advance Abdulaziz

3 4

ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed
by Abdulaziz Alghosh 21 Aug '18

21 Aug '18

Greetings, i am at the final stages of emigrating from kamailio 3.0.3 to 5.1.0 including our own module. Unluckily, after writing the command " kamctl start", the following error is popping up: * PID file /var/run/kamailio.pid does not exist -- Kamailio start failed* Thus, I used the command " kamailio -M 8 -E -e -dd " to get more debugging Info and the output is detailing the following: 0(10334) ERROR: db_mysql [km_my_con.c:129]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on '192.168.1.207' (4) 0(10334) ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool 0(10334) ERROR: ABDUL_functions [ABDUL_chk.c:1069]: init_db(): failed to connect database URL=mysql://ABDUL:ABDUL@192.168.1.207/ABDUL 0(10334) ERROR: ABDUL_functions [ABDUL_functions_mod.c:296]: mod_init(): Module ABDUL_functions not correktly initialized, pid[10334] 0(10334) ERROR: <core> [core/sr_module.c:991]: init_mod(): Error while initializing module ABDUL_functions (//lib64/kamailio/modules/ABDUL_functions.so) Seemingly the error relates to old databases url ( "URL=mysql:// ABDUL:ABDUL@192.168.1.207/ABDUL" ) inside an old Kamailio.cfg file from my 3.0.3 package which I put under another different directory. This old Kamailio.cfg file #includes my own module, so he is exporting this url. Consequently, my module is trying to connect again to this databases. I do not know why both kamailio.cfg (new: /etc/kamailio/kamailio.cfg and old one out of /etc/kamailio) are both seen for kamctl or (/sbin/kamailio). I know that my new url inside the new /etc/kamailio/kamailio.cfg is read and connected but again not sure why my hidden old kamailio.cfg is still being read as well. the attached shows my init file for "SUSE Linux Enterprise Server 11 (x86_64)" and my used kamailio *default *file under /etc/default. Thanks in advance and best regards Abdulaziz

2 2

help
by Pravin . 21 Aug '18

21 Aug '18

Hello Team, I want to install Kamailio SIP server on centOS server...do we need to download kamailio ISO file and install on server..pls provide the installation procedure/guidelines. Regards Pravin

6 14

Dialplan priority
by igor.potjevlesch＠gmail.com 21 Aug '18

21 Aug '18

Hello, I'm not sure to understand in which situation "pr" value is used. For example: if I have two identical rules except pr value: 1 with 1 and the second with 2, then the rule with pr 2 is never used. Someone can help me to understand? Thank you. The finality I'd like to accomplish is having some kind of load balacing between two values into "repl_exp" with the same inputs conditions. Thank you. Regards, Igor.

2 2

Call Transfer Scenario Using Freeswitch
by Sunil More 21 Aug '18

21 Aug '18

Hi All, I am using kamailio and freeswitch in my setup. While all users are registered on Kamailio, calls are routed to Freeswitch back to kamailio and then upstream providers. I can make user to user calls but in cases of transfer I am stuck. I am using a Polycom IP phone to test this scenario. While I initiate a attended transfer and click on transfer all the legs hangup. When I register all users on Freeswitch this works well. I am missing something on Kamailio side. Please point me in the right direction. Thanking You, Sunil More Ph : 9503338275

2 1

Enabling tcp_write_buf
by Gholamreza Sabery 21 Aug '18

21 Aug '18

Hi, In the documentation of "tcp_wq_max" option it is written that : > It has no effect if *tcp_write_buf *is not enabled. > > The interesting fact is that, there is not description of what it is and how one can enable it. There is even a question about it in the mailing lists: https://lists.kamailio.org//pipermail/sr-dev/2011-June/011521.html It seems that it is an inaccuracy in the documentation. If there is such an option how to enable it? Can anyone tell? Regards

3 2

Best practice/routing block for TCP connection errors to UAC?
by Anthony Alba 21 Aug '18

21 Aug '18

HI SR-Users Scenario: clients registered by TCP/TLS dropping off the network, kamailio stil thinks the client is registered and relays the INTIVE. The error occurs at the TCP layer so no failure route is called. ERROR: <core> [core/tcp_main.c:2690]: tcpconn_1st_send(): connect 127.0.0.1:48338 failed (RST) Connection refused ERROR: <core> [core/tcp_main.c:2700]: tcpconn_1st_send(): 127.0.0.1:48338: connect & send for 0x7f5494e906c0 failed: Connection refused (111) ERROR: tm [../../core/forward.h:251]: msg_send_buffer(): tcp_send failed WARNING: tm [t_fwd.c:1567]: t_send_branch(): sending request on branch 0 failed ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) I want to send the caller instead a graceful failure route type message as if the callee is not registered (404/486/603/408 type) but don't understand how to hook a routing block for TCP errors. Anthony

1 0

Fail to reload htable
by Soltanici Ilie 20 Aug '18

20 Aug '18

I'm trying to reload a hash table but receiving an error: error: 500 - Mtree reload failed [root@kamailio]# /usr/local/sbin/kamcmd htable.listTables { name: auth dbtable: dbmode: 0 expire: 0 updateexpire: 1 size: 256 dmqreplicate: 0 } [root@kamailio]# /usr/local/sbin/kamcmd htable.dump auth { entry: 69 size: 1 slot: { { name: username::8.8.8.8 value: 0 type: int } } } [root@kamailio]# /usr/local/sbin/kamcmd htable.reload auth error: 500 - Mtree reload failed [root@kamailio]# /usr/local/sbin/kamailio -v version: kamailio 5.1.4 (x86_64/linux) e8b2bf

2 1

Kamailio 5.1 Nathelper sipping on different transport protocols.
by Володимир Іванець 20 Aug '18

20 Aug '18

Hello everyone! I have system with Kamailio v5.1 which needs to send OPTIONS packets to all active clients. Here is full Nathelper module configuration: *modparam("nathelper", "received_avp", "$avp(s:received)")* *modparam("nathelper", "natping_interval", 30)* *modparam("nathelper", "ping_nated_only", 0)* *modparam("nathelper", "sipping_bflag", FLB_SIPPING)* *modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org <sip%3Apinger(a)kamailio.org>")* It works fine if client is connected over UDP. But as soon as I will switch to TCP or register with client that uses WS, Kamailio stops generating OPTIONS. Does anyone know if it is correct behavior or something is wrong? Thanks a lot!

3 3

remove_hf function in INVITE message
by Mojtaba 20 Aug '18

20 Aug '18

Today i encounter with strange issue in SIP signalling in Kamailio, In INVITE message, I just change Contact Header and relay message to destination, But the message is like this: 192.168.122.245.4080 > 192.168.122.174.5060: [bad udp cksum 0x7b5e -> 0x3a4d!] SIP, length: 1104 INVITE sip:8000@r-kh.com;transport=UDP SIP/2.0 Record-Route: <sip:192.168.122.245:4080;lr;ftag=8e47e95e;nat=yes> Via: SIP/2.0/UDP 192.168.122.245:4080;branch=z9hG4bK14d7.dc974309aba269b24213204d3ea085e2.0 Via: SIP/2.0/UDP 192.168.122.1:36853;rport=36853;branch=z9hG4bK-d8754z-423b431a3ebc8a22-1---d8754z- Max-Forwards: 69 sip:7000@192.168.122.1:36853;transport=UDP;alias=192.168.122.1~36853~1To: <sip:8000@r-kh.com;transport=UDP> From: <sip:7000@r-kh.com;transport=UDP>;tag=8e47e95e Call-ID: YTZmYTJmYzg0ZDNiMjM2OGUxZGYwNzg4OWY1ZmIxYzc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 241 <SDP Body> The code in Kamailio is: #!ifdef WITH_SBC_NASIM #if the Invite request is MO, forward it to Asterisk . if (is_method("INVITE") && $si!="192.168.122.245") { xlog("L_INFO", "The Invite message from $si:$sp, Forwart to Asterisk. <192.168.122.174:5060>.\n"); remove_hf("Contact"); append_hf("Contact:sip:$fU@sbc.r-kh.com\r\n"); $du = "sip:192.168.122.174:5060"; } #!endif Be notice that, i have the same procedure in REGISTER message, But it is work correct, if(is_method("REGISTER")) { xlog("L_INFO", "forward register to Asterisk\n"); remove_hf("Contact"); append_hf("Contact:sip:$fU@sbc.r-kh.com\r\n"); rewriteuri("sip:r-kh.com"); $du = "sip:192.168.122.174:5060"; route(RELAY); } Let me know what is the problem? -- --Mojtaba Esfandiari.S

2 2

Upgrade from Kamailio 4.4>5
by Nicolas Breuer 20 Aug '18

20 Aug '18

Hello, After upgrading, i cannot start anymore the service. Aug 19 10:19:57 proxy1 kamailio: ERROR: <core> [core/pvapi.c:828]: pv_parse_spec2(): error searching pvar "NULL" Aug 19 10:19:57 proxy1 kamailio: ERROR: <core> [core/pvapi.c:1032]: pv_parse_spec2(): wrong char [L/76] in [$NULL] at [4 (0)] Aug 19 10:19:57 proxy1 kamailio: ERROR: <core> [core/rvalue.c:2551]: mk_rval_expr_v(): memory allocation failure Any ideas ?

2 2

JSONRPC-S and async RPC commands
by Juha Heinanen 20 Aug '18

20 Aug '18

JSONRPC-S module README has: 2. Limitations This module does not implement asynchronous RPC commands. It is unlikely that asynchronous RPC commands will be executed from an JSON-RPC over HTTP client. Does this mean that, for example, tm.t_uac_wait command cannot be executed using JSON-RPC over HTTP? If so, why "it is unlikely that asynchronous RPC commands will be executed from an JSON-RPC over HTTP client", since those commands are currently commonly executed from XMLRPC client using xmlrpc module? It would be nice if JSONRPC-S module could be used as replacement of xmlrpc module. -- Juha

2 8

domain-based routing
by David Villasmil 19 Aug '18

19 Aug '18

Hello all, I need to set up a multi-domain kamailio, i think in terms of accepting invites to different domains is pretty simple by using the "alias" parameter, that much i have clear. My doubt comes from the fact that I need to route outgoing calls also based on the domain. As of today, i have single-domain kamailios using dispatcher to load-balance and failover said calls. I think i should be able to use dispatcher here as well, separating the dispatcher list by GID, so that Domain1->gid:1 Domain2->gid: 2 Domain3->gid: 3 and so on... I don't have registrations on this server and none are expected. So i'm thinking i will received the invites like: sip:12345@domain1.com sip:12345@domain1.com sip:12345@domain1.com and so on... How would you guys go about associating the domain with the gid list in dispatcher? Thanks a lot for your help! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

Re: [SR-Users] Mysql 5.7 vs 8.0
by Henning Westerholt 17 Aug '18

17 Aug '18

Am Freitag, 17. August 2018, 16:37:23 CEST schrieb Wilkins, Steve: > Yes, I did read that post, thank you for that; my problem is that installing > Kamailio is one of the last things I do in a script so I was hoping to find > a way to tell Kamailio to point to a different version. I guess I will > just need to go another route. Hi Steve, (add sr-users) kamailio by itself will not force the installation of a mysql version. I think this is done from the distribution package management and from the respective kamailio package. There are ways to adapt the installed dependencies during this installation, but you need to investigate it, as its depend on your linux distribution. Best regards, Henning -- Henning Westerholt https://skalatan.de/blog/

1 0

Mysql 5.7 vs 8.0
by Wilkins, Steve 17 Aug '18

17 Aug '18

Hi All, I would like Kamailio to use MySQL 5.7, however when Kamailio installs and I say I want to use MySQL, it installs 8.0. Can I someway direct Kamailio to install version 5.7? Thank you, -Steve

2 1

Registered User limit
by Philip Fleck 17 Aug '18

17 Aug '18

Is there a limit to the number of users which can be registered on Kamailio at any one time, or a configuration file which sets this. I have just installed Kamailio in a lab environment for testing purposes but I seem to only be able to register 1024 users, all remaining Register requests are ignored. I am not using a database to authenticate users or anything, just want to be able to register upto 10,000 users. regards Philip

6 6

Bogus event 8 in state 1 for dlg
by Phillman25 Kyriacou 17 Aug '18

17 Aug '18

Hi! I keep receiving the below logs on Kamailio: CRITICAL: dialog [dlg_hash.c:955]: log_next_state_dlg(): bogus event 8 in state 1 for dlg 0x7f59862cd440 [3685:10496] I understand that state 1 indicates an unconfirmed dialog, can anyone tell me what bogus event 8 is? The following is the call flow: User A ===> Kamailio ==> Provider >From the PCAP trace an INVITE is sent to the provider, the provider answers with 100 Trying, 180 Ringing and immediately an OPTIONS message with the same call id Kamailio Provider INVITE =================> 100 Trying <================= 180 Ringing <================ OPTIONS <================ Im assuming bogus event 8 is the OPTIONS message? Is an OPTIONS message normal in this case? Thanking you very much in advance for your help! Regards Phillip

3 4

register with Kamailio
by Pravin . 16 Aug '18

16 Aug '18

Hello, I have been registered with Kamalio with pravin(a)bolindia.com id ..its not working with login credential registered with kamailio. Pls suggest Regards, Pravin bolindia Networks Pvt Ltd.

2 1

Kamailio as a P-CSCF user
by serkan celik 16 Aug '18

16 Aug '18

Hi all, I want to use Kamailio as a P-CSCF on ubuntu 18. Kamailio will be p-cscf and will communicate scscf in another open-ims box. How can I implement this feature. I have already looked following link. However it did not help me since I could not see any example pcscf configuration files after installation https://www.kamailio.org/wiki/tutorials/ims/installation-howto Regards, Serkan

3 11

how to enable tcp_write_buf?
by Juha Heinanen 16 Aug '18

16 Aug '18

I tried to issue mtree.list command using xmlrpc query that would result in large response, and got error Aug 15 18:12:07 char /usr/bin/sip-proxy[10348]: ERROR: <core> [core/tcp_main.c:618]: _wbufq_add(): (221375 bytes): write queue full or timeout (0, total 0, last write 55534052 s ago) Aug 15 18:12:07 char /usr/bin/sip-proxy[10348]: ERROR: xmlrpc [xmlrpc.c:811]: send_reply(): Error while sending reply Aug 15 18:12:07 char /usr/bin/sip-proxy[10367]: ERROR: <core> [core/tcp_main.c:3551]: handle_ser_child(): received CON_ERROR for 0x7f112ad22be0 (id 5), refcnt 3, flags 0x4018 I though that perhaps I could get rid of the error by increasing value of core var tcp_conn_wq_max. Description of the var tells: Maximum bytes queued for write allowed per connection. Attempting to queue more bytes would result in an error and in the connection being closed (too slow). If tcp_write_buf is not enabled, it has no effect. What is tcp_write_buf and how to enable it? I could not find anything about it in core cookbook. -- Juha

3 7

Issue with RTPEngine(6.5.0.0+0~mr6.5.0.0)
by Mojtaba 16 Aug '18

16 Aug '18

Hello, Before i used RTPEngine (6.4.0.0+0~mr6.4.0.0) without any problem, Nowday i install RTPEngine from git. I notice that it's varsion is changed to 6.5.0.0+0~mr6.5.0.0. When i want install ngcp-rtpengine-recording-daemon, I have this issue: Failed to start ngcp-rtpengine-recording-daemon.service: Unit rpcbind.socket failed to load: No such file or directory. invoke-rc.d: initscript ngcp-rtpengine-recording-daemon, action "start" failed. dpkg: error processing package ngcp-rtpengine-recording-daemon (--install): subprocess installed post-installation script returned error exit status 6 Processing triggers for systemd (215-17+deb8u7) ... Errors were encountered while processing: ngcp-rtpengine-recording-daemon Please let me what is this issue? -- --Mojtaba Esfandiari.S

3 4

kamailio rtp proxy set not working
by ANOOP V M 16 Aug '18

16 Aug '18

I have installed kamailio 5.1.4 version in ubuntu server. I need to route RTP packets through kamailio with SIP packets. I will explain the way i have set up everything. I have installed rtpproxy by apt-get install rtpproxy. i have edited "/etc/default/rtpproxy" file with the line 'EXTRA_OPTS="-F -s udp:127.0.0.1:7722 -l 34.201.122.249 -d DBUG:LOG_LOCAL0"' i have edited "/etc/init.d/rtpproxy" file as follows, DAEMON=/usr/bin/$NAME PIDFILE="/run/$NAME/$NAME.pid I have provided "#!define WITH_NAT" and "modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")" in my cfg file. started rtpproxy "service rtpproxy start" when i try "netstat -tulpn | grep rtp" it shows udp 0 0 127.0.0.1:7722 0.0.0.0:* 1437/rtpproxy So i think the rtpproxy is currently running fine. Now we can go to kamailio part. I have gone through the document " https://www.kamailio.org/docs/modules/5.0.x/modules/rtpproxy.html" to set up rtpproxy in the cfg file. checking the rtp ports "kamcmd rtpproxy.list" showing { setid: 0 url: udp:127.0.0.1:7722 index: 0 disabled: 0 weight: 1 recheck: 0 } I have enabled rtp proxy using "kamcmd rtpproxy.enable udp:127.0.0.1:7722 1". So i think ports are ready for rtp proxy. Now i will explain the call flow. I have enabled log using L_BUG. when i try a call from hangout number through my twilio trunk to my asterisk freepbx extension, set_rtp_proxy_set("1"); giving true value. But rtpproxy_offer(); giving -1 as result. syslog shows some errors like below, Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: DBUG:handle_command: received command "1934_4 Uc0,8,3,111,9,101 52001b5b71435b1c5392b39f2f6c975b@asterisk_ip:5060 asterisk_ip 14540 as4083c85f;1" Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: INFO:handle_command: new session 52001b5b71435b1c5392b39f2f6c975b@asterisk_ip:5060, tag as4083c85f;1 requested, type strong Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: ERR:create_twinlistener: can't bind to the IPv4 port 47926: Cannot assign requested address Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: ERR:handle_command: can't create listener Aug 13 10:28:57 ip-172-30-0-10 rtpproxy[1437]: DBUG:doreply: sending reply "1934_4 E10#012" And also my header is not changed. As a result actuall rtp is going end to end between asterisk and twilio not through kamailio. Please help me with the details.

3 4

Very strange reinvite behaviour
by Alex Balashov 15 Aug '18

15 Aug '18

Hi, We have a scenario like this: ITSP -----> Kamailio -----> Endpoint (UDP) (TLS) So, TLS is only being used on the last hop, and the upstream interaction with the ITSP is plain-old UDP. Kamailio has the following listeners: listen=udp:1.1.1.1:5060 listen=udp:1.1.1.2:5060 listen=tcp:10.0.0.1:5060 listen=tls:1.1.1.1:5061 At some point, 'Endpoint' sends a reinvite which has the following RURI: INVITE sip:1.1.1.1:5061 SIP/2.0 This is clearly improper, and caused a loop that led to the rtpengine SDP loop issue I previously reported in another thread. So, in an effort to stop this, I added the following: if(has_totag()) { if(loose_route()) { ... if(is_method("INVITE")) { xlog("L_INFO", "[R-MAIN:$ci] Re-INVITE received from $si:$sp to RURI $ru\n"); xlog("L_INFO", "[R-MAIN:$ci] Reinvite body: $mb\n"); if(!is_method("ACK") && uri == myself) { sl_send_reply("400", "Bad Request"); exit; } } } But it doesn't work. It appears that the '400 Bad Request' rejection never happens, presumably because the this domain doesn't match 'myself'. Another perplexing mystery: the log message containing the reinvite's '$ru' does not show a RURI of 'sip:1.1.1.1:5061', but rather the remote target in the initial inbound INVITE, which we also logged: Contact: "Anonymous" <sip:Restricted@3.3.3.3:5060> The log message says: [R-MAIN:...] Re-INVITE received from 4.4.4.4:5060 to RURI sip:Restricted@3.3.3.3:5060;lr. Note a subtle detail here: the ';lr' parameter is present, which is an attribute of the Record-Route inserted by the sending ITSP (3.3.3.3). It's at the bottom of the Route set, of course, below Kamailio's two RRs (inserted for the ingress UDP interface and the egress TLS interface): Route: <sip:Restricted@3.3.3.3:5060;lr> This leads to two questions whose causes seem to be related: 1. Why does Kamailio think the request URI of this re-invite is something other than what $mb reveals it to be? 2. Is that, presumably, why it does not match 'myself'? 3. Why would Kamailio think it is actually set to the far-end Record-Route URI? This is version: kamailio 4.4.5 (x86_64/linux) d48094. Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 9

rtpproxy or rtpengine with Kamailio and Asterisk
by Wilkins, Steve 14 Aug '18

14 Aug '18

Good Morning everyone, Has anyone ever gotten rtpproxy/rtpengine to work with Kamailio and Asterisk? I have tried both, and even though I see no errors with either, and when I make calls I see rtpproxy/rtpengine traffic, I am still seeing media traffic go directly to and from Asterisk. I just can't get the media packets to not go directly from Asterisk to the Callee and visa-versa. Any ideas on where I am messing up in my kamalio.cfg file? Thank you!, -Steve

3 2

Migrate from RtpProxy to Rtpengine
by Nicolas Breuer 14 Aug '18

14 Aug '18

Hello, I migrated from Rtp_proxy module to Rtp_Engine. Everythings works fine apart the rewriting of the SDP description in Kamailio subst_body('/(^s=.*)/s=Test of a Sip call.\r/') Any ideas if this is supposed to be supported?

4 11

Can Kamailio interface with an asterisk server based on the fqdn of a call?
by Wilkins, Steve 14 Aug '18

14 Aug '18

Hi All, Is it possible for Kamailio to interface with a particular Asterisk Server based on the FQDN of a caller? I would like to pass calls received by Kamailio through to different Asterisk Servers based on the FQDN of the caller. I have used Load Balancing before, but I want to select which Asterisk Server the call is directed to. Thank you,

3 2

Rtpengine?
by Wilkins, Steve 14 Aug '18

14 Aug '18

HI All, I am not sure if I understand it correctly but I thought that I could use rtpengine to redirect media packets. My current SIP flow is => Softphone=>Kamailio=>Asterisk=>Kamailio=>Softphone, and Media flows from is Asterisk<=>softphone. But I don't want Media to flow this way. That is, I do not want Asterisk and the Softphone to be aware of each other. I would like the Media to go through Kamailio just like SIP packets do. I installed rtpengine and it starts and runs with no errors. I even see traffic with the call is made but media traffic still flows from softphone<=>asterisk. Any ideas? Thank you,

2 3

Upgrade from 4.4 to 5.1
by Nicolas Breuer 13 Aug '18

13 Aug '18

Hello, I want to upgrade my both Kamailio from 4.4 to 5.1 I have to upgrade the SQL structure before running the 5.1 release. But my SQL servers are synchronized - Kamailio 4.4 can works with the 5.1 SQL structure ?

2 4

Re: [SR-Users] problem with compiling kamailio version 5.1.4 including a private Module
by Abdulaziz Alghosh 13 Aug '18

13 Aug '18

Hi Henning, thanks again. I found meanwhile the root cause. It caused as you said by the old logic of the make file. I used instead the new Formula of the tm module’s make file (as i remember). Consequently, it did not bring the compilation through those unwanted libraries. I will spot the details later when i am back in office next Monday. BR Abdulaziz On Thu 9. Aug 2018 at 21:19 Henning Westerholt <hw(a)kamailio.org> wrote: > Am Donnerstag, 9. August 2018, 14:27:44 CEST schrieb Abdulaziz Alghosh: > > thanks a lot for your response and explanation. strutils.[c,h] and > > strcommon.[c,h] are almost the same. So I modified my included header > from > > stcommon[] to strutils as you mentioned. > > Unfortunately, I surfed sequentially Kamailio version upgrades. Even > > though, i did not found such hints about this modification. So I have > some > > doubts the called headers and codes from our own module. Hence, I am > asking > > for yor advice. > > Hello Abdul, > > lets move this discussion to the sr-dev list, as its a development related > topic. > > > What I am trying to do now is compiliing our module separately under it > is > > own directory so I gain the Shared Object file (.so). Somehow our Module > is > > requring cds, presence and xcap libraries. But the strange thing is that > > headers from these libraries are #including other headers assuming these > > included header files are under the same library's directory (but in fact > > most of the header files are under ../src/core/). For example : > > Do you placed your own module in the src/modules directory as well? If > not, I > would suggest to place it there. Its hard to find the issue just with the > logs, there are also many errors here. > > My suggestion would be to go from your own module and try to compile a > single > file with gcc (for the actual command you can call "make quiet=0" and copy- > paste it). I would try to investigate one error a time, in the end they > should > be all mostly related. > > There were also some changes in the Makefile logic, maybe you should > compare > your Makefile to one of the official modules. > > > Making in cds > > CC (gcc) [L libser_cds.so.0.1] cds.o > > In file included from cds.c:2: > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:125:21: > > warning: mem/mem.h: No such file or directory > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:126:25: > > warning: mem/shm_mem.h: No such file or directory > > In file included from cds.c:3: > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/sync.h:30:22: > > warning: locking.h: No such file or directory > > In file included from cds.c:4: > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/logger.h:42:20: > > error: dprint.h: No such file or directory > > make[2]: *** [cds.o] Error 1 > > make[1]: [cds] Error 2 (ignored) > > > > > > Those libraries are delivered with version 5.1.4 itself so i do not think > > it is a good idea to modify the paths of the included header files to > > compile my module. because most probably these libraries are needed by > > standard default modules of kamailio. > > > > I hope this is not too much for you to inspect. > > Note: The attached has the log files ( resulting from make all and > install > > ) . > > > > Your kind response is highly appreciated > > BR > > Abdulaziz > > > > On Thu, Aug 9, 2018 at 11:31 AM, Abdulaziz Alghosh <aziz647(a)gmail.com> > > > > wrote: > > > Hello Henning, > > > > > > thanks a lot for your response and explanation. strutils.[c,h] and > > > strcommon.[c,h] are almost the same. So I modified my included header > from > > > stcommon[] to strutils as you mentioned. > > > Unfortunately, I surfed sequentially Kamailio version upgrades. Even > > > though, i did not found such hints about this modification. So I have > some > > > doubts the called headers and codes from our own module. Hence, I am > > > asking > > > for yor advice. > > > > > > What I am trying to do now is compiliing our module separately under > it is > > > own directory so I gain the Shared Object file (.so). Somehow our > Module > > > is > > > requring cds, presence and xcap libraries. But the strange thing is > that > > > headers from these libraries are #including other headers assuming > these > > > included header files are under the same library's directory (but in > fact > > > most of the header files are under ../src/core/). For example : > > > > > > Making in cds > > > CC (gcc) [L libser_cds.so.0.1] cds.o > > > In file included from cds.c:2: > > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:125:2 > > > 1: warning: mem/mem.h: No such file or directory > > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:126:2 > > > 5: warning: mem/shm_mem.h: No such file or directory > > > In file included from cds.c:3: > > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/sync.h:30:22: > > > warning: locking.h: No such file or directory > > > In file included from cds.c:4: > > > > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/logger.h:42:20 > > > : > > > error: dprint.h: No such file or directory > > > make[2]: *** [cds.o] Error 1 > > > make[1]: [cds] Error 2 (ignored) > > > > > > > > > Those libraries are delivered with version 5.1.4 itself so i do not > think > > > it is a good idea to modify the paths of the included header files to > > > compile my module. because most probably these libraries are needed by > > > standard default modules of kamailio. > > > > > > I hope this is not too much for you to inspect. > > > Note: The attached has the log files ( resulting from make all and > install > > > ) . > > > > > > Your kind response is highly appreciated > > > Abdulaziz > > > > > > > > > On Thu, Aug 9, 2018 at 8:11 AM, Henning Westerholt <hw(a)kamailio.org> > > > > > > wrote: > > >> Am Mittwoch, 8. August 2018, 17:47:07 CEST schrieb Abdulaziz Alghosh: > > >> > i am somehow newbie with Kamailio and trying to migrate from > kamailio > > >> > version 3.0.3 to 5.1.4. Former fellows had developed own modules > which > > >> > needed "strcommon.h" and "strcommon.c". > > >> > > > >> > These last two files are not delievered with version 5.1.4. but I > > >> > copied > > >> > them under ../src/core/ thinking it would be beneficial. > Unfortunately, > > >> > after make prefix / (and including our own module) all, it seem that > > >> > several dependencies exist. Especially from header files under > > >> > ..src/lib/xcap/ and ../src/lib/presence > > >> > > > >> > Can somebody tell me how "strcommon.h" and "strcommon.c" where > > >> > > >> replaced in > > >> > > >> > version 5.1.4 ? > > >> > > >> Hello Abdul, > > >> > > >> the functions in strcommon.[c,h] were moved into > src/core/strutils.[c,h] > > >> in > > >> releases starting from 5.0. I'd suggest to have a look to the file to > > >> verify > > >> if there are no other changes. > > Best regards, > > -- > Henning Westerholt > https://skalatan.de/blog/ >

1 1

Kamailio IMS with TLS
by Amar Tinawi 13 Aug '18

13 Aug '18

Hello i'm running tls conf over PCSCF, register and audio calls works fine, but when it comes to video calls it's dropped when it comes to video second-invite it worked when the call is initiated from the beginning as video call should i enable tls also on icscf and scscf ? or the video is not supported over tls ?

1 0

app_lua and lua_run parameter count
by Asgaroth 13 Aug '18

13 Aug '18

Hi All, Quick question with regards the lua_run function in app_lua module. The documentation for v5.0.x states: "Execute the Lua function 'func' giving params as parameters. There can be up to 3 string parameters. The function must exist in the script loaded at startup via parameter 'load'. Parameters can be strings with pseudo-variables that are evaluated at runtime." Does this mean that lua_run can only have 3 parameters, or, does it mean that the function within the lua script can have up to 3 parameters? I have a function within lua that currently needs 3 parameters, so I'm hoping I can do the following: lua_run("some_function", "param_1", "param_2", "param_3") However, I'm not sure, from the documentation, if I can do this. On a side note, but related, can I pass defined variables/constants as parameters to lua_run too, for example, something like: #!define VAR_1 1234 #!define VAR_2 hello lua_run("some_function", "$rd", VAR_1, VAR_2) Some clarification would be greatly appreciated. Thanks for reading.

2 2

Lua KEMI crash
by Ivaylo Markov 13 Aug '18

13 Aug '18

Hello, I have Kamailio 5.1.4 (from deb.kamailio.org) running on a Debian 9 AMD64 VM with 1 CPU and 1G of RAM on ESXi. For KEMI I am using the liblua5.1-0:amd64 package, version 5.1.5-8.1+b2. The configuration is rather simple - dispatcher module reading the gateways from a file and forwarding everything to them based on a custom variable. There are 2 gateways in the file being pinged with OPTIONS packets every 10 seconds. Only a single SIP user is connecting to the machine (besides the occasional scanner). I can post the config if necessary, but I don't think that is the issue (see below). The crash has happened twice already - once on 2018-08-07 19:45 and 2018-08-08 14:33 EEST. You can find an excerpt from the Kamailio debug log at https://paste.fedoraproject.org/paste/bvyUMj3s5g2CcBN3JSBt5A/raw. As you can see, it crashes when trying to run the Lua route on an incoming packet. I am assuming the problem is this: Aug 8 14:33:14 lb-dev1 kamailio[13209]: *** Error in `/usr/sbin/kamailio': realloc(): invalid next size: 0x00005573dbff1ef0 *** However, I could not find any (direct) `realloc()` calls in the source for app_lua, so maybe this is an issue somehwere in liblua... Thoughts? Ivaylo

1 1

USRLOC db_mode=4 and server_id
by Darren Williams 13 Aug '18

13 Aug '18

I am currently using multiple instances of Kamailio with a common database and using the usrloc module with db_mode=4 and server_id to filter the contacts that get loaded at startup on each server. The problem I seem to have is that using the RPC call ul.add there I cannot see any parameter to send to specify the server_id. The server_id that the contact is given is 0. If I manually go into the db and modify the server_id everything works as expected. I think I must be missing something somehwere as I'mn sure you shouldn't need to do this. Can anyone help please. Thanks Daz

1 0

Kamailio TLS + letsencrypt
by Amar Tinawi 13 Aug '18

13 Aug '18

Hello i'm wondering if i can generate tls certificate for kamailio using letsencrypt with test domain name, in testing enviroment following this post : https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ got an error by letsecrypt because of checking of the domain name authority and no certificate was generated. Regards

3 2

mysql version
by Wilkins, Steve 12 Aug '18

12 Aug '18

Hello All, When Kamailio installs, it installs mysql 8.0. I would like it to install mysql 5.7. Is there a way to force this? Thank you

2 1

MYSQL and various
by Laura 12 Aug '18

12 Aug '18

Dear list, on my kamailio 4.4.7 i use interactions with MYSQL. For example .. i use it for some black list control and other like this.. sql_query("cb", "select id from blacklist where cli='$var(g2)' and (ippeer = '$si' or ippeer = 'ALL')", "rabl"); if($dbr(rabl=>rows)>0){ sql_result_free("rabl"); sl_send_reply("403","Forbidden blacklist"); exit; } sql_result_free("rabl"); I was wonder what was the best method to prevent the query going on if mysql don't respond for high load or other situation.. If Mysql is not able to reply to the query tight now the call continue over the configuration... what is the best way to prevent this and for examples drop it with a standard SIP 503 error. I have search on docs.. but i haven't found any valid way. Thanks Laura

2 1

problem with rewriteuri
by Eric Hosch 12 Aug '18

12 Aug '18

If I manually pass a string value to rewriteuri it works, but if I pass the same string from a database value it doesn't: if ($dbr(ra=>[0,4]) == "callfwd_noanswer") { $ does not work --> rewriteuri("$dbr(ra=>[0,6])"); works --> rewriteuri("sip:someone@somewhere"); append_branch(); route(0); Eric Hosch Chief Engineer KLBK-TV | KAMC-TV 7403 S. University Ave. Lubbock, TX 79423 O * 806.748.2208<tel:806.748.2208> F * 806.748.2269<tel:806.748.2269> ehosch(a)klbk13.tv<mailto:ehosch@klbk13.tv> www.everythinglubbock.com<http://www.everythinglubbock.com/> [Nexstar Media Group, Inc. | everythinglubbock.com | KLBK-TV | KAMC-TV] Nexstar Media Group, Inc. provides services for KAMC-TV and Mission Broadcasting, Inc. through an Outsourcing Agreement. KLBK and KAMC do not discriminate in advertising contracts on the basis of race, ethnicity or gender and further requires that in the performance of all KLBK/KAMC advertising agreements. KLBK/KAMC requires that each party not discriminate on the basis of race or ethnicity. This e-mail and any files transmitted with it are the property of NexstarMedia Group, Inc., are confidential, and are intended solely for the use of the individual or entity to whom this e-mail is addressed and/or as indicated in the applicable file. If you are not one of the named recipient(s) or otherwise have reason to believe that you have received this message in error, please notify the sender and delete this message immediately from your computer. Any other use, retention, dissemination, forwarding, printing or copying of this e-mail is strictly prohibited.

2 1

Nat issue
by eyas barhouk 11 Aug '18

11 Aug '18

hello dears when i'm using kamailio 5.2 as IMS i'm facing the following NAT issue : * when p-cscf handle a call from user1 to user2 that pre-registered on private address "192.X.X.X" (on another LAN) the user can't receive the invite message . in another word: proxy ip : 10.192.X.X user1 ip : 10.X.X.X user2 ip : 192.X.X.X the call scenario : user1 ---invite----> user2 (not arrived) * when p-cscf handle a call from user2 that pre-registered on private IP "192.X.X.X" (on another LAN) to user1 the user receive the invite message . the call scenario : user2 ---invite----> user1 (arrived) * when both of the user on different LANs the message 200 OK doesn't arrive to proxy :\ so based on that does any one has an experiment with NAT handling & nathelper module and how to configure it on P-cscf . thanks in advance

2 2

Ratio between cores ram and kamailio limits
by Renato Gallo 10 Aug '18

10 Aug '18

Hello, We're here to ask you what's the ratio between cpu cores/ghz, ram and how many contemporary calls kamailio can manage before it maxes out its capabilities and crash. Kind Regards Renato Gallo System Engineer sede legale e operativa: Via San brunone, 13 - 20156 - Milano (MI) Tel. +39 02 - 87049490 Fax +39 02 - 48677349 Mobile. +39 342 - 6350524 Wi | FreeNumbers: https://freenumbers.way-interactive.com Wi | SMS: https://sms.way-interactive.com Wi | Voip: https://voip.way-interactive.com Asterweb: http://www.asterweb.org Le informazioni contenute in questo messaggio e negli eventuali allegati sono riservate e per uso esclusivo del destinatario. Persone diverse dallo stesso non possono copiare o distribuire il messaggio a terzi. Chiunque riceva questo messaggio per errore è pregato di distruggerlo e di informare immediatamente [ mailto:info@sigmaware.it | info@ ] asterweb.org

2 1

UDP workers block when one or more rtpengine instances go offline
by Sebastian Damm 10 Aug '18

10 Aug '18

Hi, we run multiple rtpengine servers to share the load. Whenever we need to take an rtpengine server offline, we used to just block the control port via iptables, then no new calls ended up on this instance of rtpengine. This worked pretty well in Kamailio 4.4.5. However, since Kamailio 5.0, and the problem persists with 5.1.4, Kamailio hangs almost immediately after we block the control port traffic. In the log file there are almost no packets processed except every few seconds, which looks like a timeout thing. Did we configure anything wrong there? Or is the "dead rtpengine detection" just broken? Our configuration: loadmodule "rtpengine.so" modparam("rtpengine", "rtpengine_disable_tout", 120) modparam("rtpengine", "setid_avp", "$avp(rtpsetid)") modparam("rtpengine", "rtpengine_sock", "0 == udp:1.2.3.4:9001=2 udp:1.2.3.5:9001=2 udp:1.2.3.6:9001=2") modparam("rtpengine", "rtpengine_sock", "1 == udp:2.3.4.5:9001=2") Any help appreciated. Regards, Sebastian

4 11

kamcmd htable.reload propertiesHash problem
by Ginhoux, Patrick 10 Aug '18

10 Aug '18

Hi, I run Kamailio 5.0.1 with MariaDB database. There is no issue at the initialization, meaning the db is opened by kamailio and the hash tables loaded from the db tables. Example: # kamcmd htable.dump propertiesHash { entry: 1 size: 1 slot: { item: { name: cfg.env But the htable.reload command fails: error: 500 - Failed to open htable db connection The Kamailio log reports that it can't connect the db : DEBUG: ctl [../../core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0x7f68c74ab300, 10, 3, 0xdc9410), fd_no=1 DEBUG: ctl [io_listener.c:442]: handle_new_connect(): handle_stream read: new connection (1) on /var/run/kamailio//kamailio_ctl DEBUG: <core> [core/sr_module.c:694]: find_mod_export_record(): find_export_record: found <db_bind_api> in module db_mysql [/usr/lib64/kamailio/modules/db_mysql.so] DEBUG: <core> [db.c:209]: db_bind_mod(): using db bind api for db_mysql DEBUG: <core> [db.c:314]: db_do_init2(): connection 0x7f68cc354c10 not found in pool DEBUG: db_mysql [km_my_con.c:98]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@/siprouter ERROR: db_mysql [km_my_con.c:124]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2 "No such file or directory") ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool ERROR: htable [ht_db.c:92]: ht_db_open_con(): failed to connect to the database My local db is located at /myqdata/mysiprdb/data where is the mysiprdb.sock file. But the log shows that Kamailio is looking for /var/lib/mysql/mysql.sock. Does this error make sense to you ? Is it possible to fix this problem by configuration in Kamailio ? My kamailio.cfg script contains : # ----- htable modparam("htable", "db_url", "mysql://[kamailio]/siprouter") modparam("htable", "fetch_rows", 1000) modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") modparam("htable", "htable", "abpqHash=>size=12;dbtable=abpq;") modparam("htable", "htable", "zabpqHash=>size=12;dbtable=zabpq;") modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") In the /myqdata/mysiprdb/data/my.cnf, I added the following entries for Kamailio: [kamailio] socket = /myqdata/mysiprdb/data/mysiprdb.sock user = siprouter password = ########## default-character-set = utf8 Thanks all for help. Cordialement. Patrick GINHOUX

2 12

Kamailio troubleshooting
by Sergey Safarov 09 Aug '18

09 Aug '18

One my server with perl external script time to time is stopping processing SIP messages. Perl script make calls to kamailio database on mysql. I have found this in logs DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0x56427eb40c40, 495, 2, 0x7f1a9f9ba408), fd_no=33 DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0x56427eb40c40, 495, -1, 0x0) fd_no=34 called DEBUG: <core> [core/tcp_main.c:4196]: handle_tcpconn_ev(): sending to child, events 1 DEBUG: <core> [core/tcp_main.c:3871]: send2child(): WARNING: no free tcp receiver, connection passed to the least busy one (56) DEBUG: <core> [core/tcp_main.c:3875]: send2child(): selected tcp worker 4 20(24) for activity on [tcp:0.0.0.0:81], 0x7f1a9f9ba408 And no next messages. How get info what is load on "tcp receiver". This host in dev mode and handles not more 5 devices. Sergey

2 2

Re: [SR-Users] problem with compiling kamailio version 5.1.4 including a private Module
by Abdulaziz Alghosh 09 Aug '18

09 Aug '18

Hello Henning, thanks a lot for your response and explanation. strutils.[c,h] and strcommon.[c,h] are almost the same. So I modified my included header from stcommon[] to strutils as you mentioned. Unfortunately, I surfed sequentially Kamailio version upgrades. Even though, i did not found such hints about this modification. So I have some doubts the called headers and codes from our own module. Hence, I am asking for yor advice. What I am trying to do now is compiliing our module separately under it is own directory so I gain the Shared Object file (.so). Somehow our Module is requring cds, presence and xcap libraries. But the strange thing is that headers from these libraries are #including other headers assuming these included header files are under the same library's directory (but in fact most of the header files are under ../src/core/). For example : Making in cds CC (gcc) [L libser_cds.so.0.1] cds.o In file included from cds.c:2: /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:125:21: warning: mem/mem.h: No such file or directory /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:126:25: warning: mem/shm_mem.h: No such file or directory In file included from cds.c:3: /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/sync.h:30:22: warning: locking.h: No such file or directory In file included from cds.c:4: /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/logger.h:42:20: error: dprint.h: No such file or directory make[2]: *** [cds.o] Error 1 make[1]: [cds] Error 2 (ignored) Those libraries are delivered with version 5.1.4 itself so i do not think it is a good idea to modify the paths of the included header files to compile my module. because most probably these libraries are needed by standard default modules of kamailio. I hope this is not too much for you to inspect. Note: The attached has the log files ( resulting from make all and install ) . Your kind response is highly appreciated BR Abdulaziz On Thu, Aug 9, 2018 at 11:31 AM, Abdulaziz Alghosh <aziz647(a)gmail.com> wrote: > Hello Henning, > > thanks a lot for your response and explanation. strutils.[c,h] and > strcommon.[c,h] are almost the same. So I modified my included header from > stcommon[] to strutils as you mentioned. > Unfortunately, I surfed sequentially Kamailio version upgrades. Even > though, i did not found such hints about this modification. So I have some > doubts the called headers and codes from our own module. Hence, I am asking > for yor advice. > > What I am trying to do now is compiliing our module separately under it is > own directory so I gain the Shared Object file (.so). Somehow our Module is > requring cds, presence and xcap libraries. But the strange thing is that > headers from these libraries are #including other headers assuming these > included header files are under the same library's directory (but in fact > most of the header files are under ../src/core/). For example : > > Making in cds > CC (gcc) [L libser_cds.so.0.1] cds.o > In file included from cds.c:2: > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:125:21: > warning: mem/mem.h: No such file or directory > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/memory.h:126:25: > warning: mem/shm_mem.h: No such file or directory > In file included from cds.c:3: > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/sync.h:30:22: > warning: locking.h: No such file or directory > In file included from cds.c:4: > /Abdul/install/clean_test/kamailio-5.1.4/src/lib/cds/../cds/logger.h:42:20: > error: dprint.h: No such file or directory > make[2]: *** [cds.o] Error 1 > make[1]: [cds] Error 2 (ignored) > > > Those libraries are delivered with version 5.1.4 itself so i do not think > it is a good idea to modify the paths of the included header files to > compile my module. because most probably these libraries are needed by > standard default modules of kamailio. > > I hope this is not too much for you to inspect. > Note: The attached has the log files ( resulting from make all and install > ) . > > Your kind response is highly appreciated > Abdulaziz > > > On Thu, Aug 9, 2018 at 8:11 AM, Henning Westerholt <hw(a)kamailio.org> > wrote: > >> Am Mittwoch, 8. August 2018, 17:47:07 CEST schrieb Abdulaziz Alghosh: >> > i am somehow newbie with Kamailio and trying to migrate from kamailio >> > version 3.0.3 to 5.1.4. Former fellows had developed own modules which >> > needed "strcommon.h" and "strcommon.c". >> > >> > These last two files are not delievered with version 5.1.4. but I copied >> > them under ../src/core/ thinking it would be beneficial. Unfortunately, >> > after make prefix / (and including our own module) all, it seem that >> > several dependencies exist. Especially from header files under >> > ..src/lib/xcap/ and ../src/lib/presence >> > >> > Can somebody tell me how "strcommon.h" and "strcommon.c" where >> replaced in >> > version 5.1.4 ? >> >> Hello Abdul, >> >> the functions in strcommon.[c,h] were moved into src/core/strutils.[c,h] >> in >> releases starting from 5.0. I'd suggest to have a look to the file to >> verify >> if there are no other changes. >> >> Best regards, >> >> Henning >> -- >> Henning Westerholt >> https://skalatan.de/blog/ >> > >

1 0

problem with compiling kamailio version 5.1.4 including a private Module
by Abdulaziz Alghosh 09 Aug '18

09 Aug '18

Greetings, i am somehow newbie with Kamailio and trying to migrate from kamailio version 3.0.3 to 5.1.4. Former fellows had developed own modules which needed "strcommon.h" and "strcommon.c". These last two files are not delievered with version 5.1.4. but I copied them under ../src/core/ thinking it would be beneficial. Unfortunately, after make prefix / (and including our own module) all, it seem that several dependencies exist. Especially from header files under ..src/lib/xcap/ and ../src/lib/presence Can somebody tell me how "strcommon.h" and "strcommon.c" where replaced in version 5.1.4 ? Best regards Abdul Aziz

2 1

Questions about changes between 5.0.1 and 5.0.7
by Ginhoux, Patrick 08 Aug '18

08 Aug '18

Hi, Following a problem with the kamcmd htable.reload with Kamailio 5.0.1 (see related thread for this), I have upgraded to the 5.0.7 level (with some difficulties due to my specific environment). My server runs the RHEL 7.4 distribution. Now the upgrade is complete, I notice that that are changes not really documented: - The kamailio-perl-5.0.7-1.el7.x86_64.rpm requires to install the mod_perl module that doesn't exist anymore in RHEL 7. It needs to be installed from the EPEL repository - The init script for kamailio has been changed from the old method to the fully compliant method for system unit for RHEL 7 for example: With the 5.0.1 level we have the "/etc/rc.d/init.d/kamailio" With the 5.0.7 level we have now the "/usr/lib/systemd/system/kamailio.service" So my questions/comments are : - Why the mod_perl module is required between 5.0.1 and 5.0.7 (question posted in the attached message) ? - In the previous kamailo init script, I added a condition to start MySQL before kamailio by adding this entry in the script : # Required-Start: $syslog $network $mysqld The new script contains the After command : [Unit] Description=Kamailio (OpenSER) - the Open Source SIP Server After=network.target Should I change the After line with this : After=network.target mysql.target to have the same condition? In all the cases, some documentation must be changed to provide more details. Thanks in advance for the answers. Cordialement. Patrick GINHOUX

3 16

RTPEngine UDP receive queue issue
by Alex Balashov 08 Aug '18

08 Aug '18

Hi, We have an RTPEngine that occasionally finds itself in this state of CPU usage: PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 25476 root 20 0 2929600 80820 1900 S 156.2 0.1 259:05.54 rtpengine The journal is a dizzying stream of: Aug 06 20:58:47 gw1 rtpengine[25476]: ERR: [1074537188_44545952 port 61666]: Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible It eventually leads to increasing CPU usage and a degeneration of audio quality until RTPEngine is restarted. It would be appreciated if someone could shed some light on the nature of this queue and what we can do about this problem. Is this referring to a netfilter packet forwarding queue, or is it a receive queue of the RTPengine user space control process? Can its size be increased or tweaked with kernel parameters? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 10

sipwise/sems for video
by Amar Tinawi 08 Aug '18

08 Aug '18

Hello i don't know if the question is related to this list, but as far as i'm using Kamailio i though i should ask here i'm using Kamailio and sems as media server, the audio is passing through fine but when it comes to Video, SEMS block it, in the outgoing SDP m=video is removed with it's codec my question is does sems support Video ? or is there any tuning for sems to allow Video media streams ? Cheers Amar

2 3

Regarding OpenSource Meetup in India
by surendra tiwari 08 Aug '18

08 Aug '18

Hello Guys, We are having a Meetup for Telecommunication Geeks. We are talking around Opensource Technology for building telecommunication solution. Here's the agenda for our first meetup. We do have one slot open. Please contact us or drop a comment if you want to present. Hoping that everyone would have fun geeking out on telecom and voip tech. 10:00 - 10:25 - How VOIP is changing the Communication landscape - Rajat Dwivedi 10:30 - 10:55 - Generic VoIP Solutions on the cloud - Chandra Prakash Singh 10:55 - 11:10 - Break 11:10 - 11:35 - Developing VoIP solutions using Open Source Tech - Surendra Tiwari 11:40 - 12:05 - WebRTC: Telephony in your browser - Altanai Bisht 12:05 - 12:30 - Open Slot (looking for speakers) 12:30 - 1:30 - Lunch + Networking 13:30 - 14:15 - Birds of a Feather Session (BoF) Please RSVP. With Regards, Surendra Tiwari VoIP Application Developer, Plivo Comm +91-9967609476

3 2

MSILO messages to users behind NAT
by Michalis Michael 07 Aug '18

07 Aug '18

I am using Kamailio (5.0.1) and MSILO module for offline messages but I have a problem with users behind NAT. According to module overview, MSILO is forwarding the offline messages to contact address provided in REGISTER request. Is there any way to overcome this and send the messages to the external client IP? Thanks, Michalis Michael

1 0

IRC Channel
by Enrico Bandiera 07 Aug '18

07 Aug '18

Hi, currenty (the last few days) there has been a spam attack on freenode, #Kamailio is currently full of spam messages, I was wondering if who has operator rights could set the channel to +r (only registered users can send messages) as suggested by freeenode admins The channel usually has only few messages and with all this spam it becomes impossible at all to use it. Greetings, Enrico.

4 4

Kamctl Stats Update
by Noah Mehl 07 Aug '18

07 Aug '18

Hey all, I was wondering if the kamctl stats output could be updated to provide a bit more parseable JSON? For instance: # kamctl stats shmem | jq . { "jsonrpc": "2.0", "result": [ "shmem:fragments = 18", "shmem:free_size = 467187808", "shmem:max_used_size = 69694104", "shmem:real_used_size = 69683104", "shmem:total_size = 536870912", "shmem:used_size = 41048984" ], "id": 4300 } I would hope would be more like this: { "jsonrpc":"2.0", "result":[ { "shmem":[ { "fragments":18, "free_size":467187808, "max_used_size":69694104, "real_used_size":69683104, "total_size":536870912, "used_size":41048984 } ] } ], "id":4300 } I apologize in advanced because I don’t have the skillset to contribute such a feature. Thanks! ~Noah

5 8

"kamdbctl create" failing with MySQL 8
by Wilkins, Steve 05 Aug '18

05 Aug '18

Hello All, There appears to be issues with running "kamdbctl create" when using MySQL 8. When this is ran there are syntax SQL errors. One such error is when doing grants using "IDENTIFIED BY 'password'"; This throws a sequel error for version 8 of MySQL. Thank you,

2 1

DMQ problems
by Aleksandar Sosic 05 Aug '18

05 Aug '18

Hi, I've set up a dmq server for other kamailio nodes with this parameters: ``` modparam("dmq", "server_address","sip:10.0.0.101:5060") modparam("dmq", "notification_address", "sip:127.0.0.1:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 15) ``` On the kamailio nodes the config is the following: ``` modparam("dmq", "server_address", "sip:10.0.0.x:5060") modparam("dmq", "notification_address", "sip:10.0.0.101:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 15) ``` On the kamailio dmq server I've got this in the logs: ``` 25(41) ERROR: dmq [notification_peer.c:596]: notification_resp_callback_f(): deleting server sip:127.0.0.1:5060 because of failed request 25(41) ERROR: dmq [notification_peer.c:599]: notification_resp_callback_f(): not deleting notification_peer ``` On the kamailio nodes the same thing with the ip of the dmq server: ``` 25(112) ERROR: dmq [notification_peer.c:596]: notification_resp_callback_f(): deleting server sip:10.0.0.101:5060 because of failed request 25(112) ERROR: dmq [notification_peer.c:599]: notification_resp_callback_f(): not deleting notification_peer ``` The request route on the dmq server is simple as this: ``` request_route { if(is_method("KDMQ")){ dmq_handle_message(); } }#end request_route ``` I've tried with some xlog-s inside the if(is_method("KDMQ")) but they don't show up so the dmq_handle_message() is not even executed. I've also tried to bind the dmq server to a different port but with no success on making it work. The dmq.list_nodes shows only the dmq server IP and the current node IP, the IPs of other nodes are not shown: ``` root@proxy-94b6ccf46-t8fx2:/# kamcmd dmq.list_nodes { host: 10.0.0.101 port: 5060 resolved_ip: 10.0.0.101 status: active last_notification: 0 local: 0 } { host: 10.0.0.45 port: 5060 resolved_ip: 10.0.0.45 status: active last_notification: 0 local: 1 } ``` And it's the same on all other kamailio nodes: dmq server ip and local ip. The kamailio version on all the nodes is: 5.2.0~dev6+0~20180616010152.1138+xenial Thanks, -- Aleksandar Sosic mail: alex.sosic(a)evosip.cloud

4 20

Security announcement related to Kamailio
by Henning Westerholt 04 Aug '18

04 Aug '18

Hello, I want to highlight that the last stable versions (for the two maintained series: 5.0 and 5.1) include fixes for an security issues that can crash a running instance of Kamailio, therefore it is strongly recommended to upgrade. Details: There exists a security vulnerability in the Kamailio SIP server related to "To" header processing. A specially crafted SIP message with double "To" header and an empty "To" tag causes a segmentation fault and crashes Kamailio. The reason is missing input validation in the "build_res_buf_from_sip_req" core function. If an attacker sends many of this messages this would lead to a Denial of Service of the attacked infrastructure. This is especially critical as no authentication for the remote source is needed. You find all the details including a proof of concept code in the published security announcement in my blog: https://skalatan.de/blog/advisory-hw-2018-05 A CVE report for this issue is going to be created as well in the near future. The issues were found some months ago and were fixed quickly. The code related to the reported issues is rather old and there are no known incidents of exploiting these issues so far. However, once the CVE report becomes public, there could be a higher risk of exploitation. Please address any detailed technical questions related to this to the developer list at sr-dev(a)lists.kamailio.org . In case of confidential remarks related to this or other security issues, please address them to the Kamailio Management. Best regards, Henning Westerholt -- Henning Westerholt https://skalatan.de/blog/

1 1

Nathelper not obeying Advertised IP
by John Martirosian 03 Aug '18

03 Aug '18

Hello, The nathelper module does not obey the advertise IP. Is there any work around for this? When kamailio is running behind a NAT with advertised IP set the nathelper module doesn't use that IP when sending SIP OPTIONS keepalives. Here is a picture of the packet coming from Kamailio. https://i.imgur.com/9vcpOoo.png Thanks,

1 0

Final reply code is wrong in case of 302 and 486 busy
by Marco Capetta 03 Aug '18

03 Aug '18

Hello, we have a problem of final reply code in case of 302 and then 486 busy. This is our scenario: A, B and C are subscribers. A ----------> Kamailio ----------> B Kamailio <---302---- B Kamailio -----------------------> C Kamailio <---486----------------- C A <---302---- Kamailio Expected behavior: Kamailio should reply to A with 486, as C was busy, and not 302. We have kamailio v5.1.4 In the call to B we set both failure and branch_failure: t_on_failure("failure_route"); t_on_branch_failure("failure_redirect"); in order to handle the 302 redirect per branch with: event_route[tm:branch-failure:failure_redirect] { if($T_reply_code == 301 || $T_reply_code == 302) { do redirect ... } } and the final failure with: failure_route[FAILURE_ROUTE_LOCAL] { do something ... } Since we have 'modparam("tm", "failure_reply_mode", 3)' my expectation is to get the winning reply from the branch of last step. Is there any configuration I'm missing? Since we manage the redirect with a branch-failure, is the winning reply calculated in a different way? Thanks Marco

2 1

Other issues encountered with MySQL 8 and Kamailio
by Wilkins, Steve 03 Aug '18

03 Aug '18

Good Morning all, When I do get Kamailio to compile with MySQL 8, I encounter the following issue when running "kamdbctl create" ERROR 1449 (HY000) at line 1: The user specified as a definer ('mysql.infoschema'@'localhost') does not exist WARNING: Your current default mysql characters set cannot be used to create DB. Please choice another one from the following list: Any ideas? Thank you.

1 1

5.1 compilation error on Centos 7
by Wilkins, Steve 03 Aug '18

03 Aug '18

Hello all, I started getting the following error while trying to compile Kamailio 5.1+ on Centos 7 CC (gcc) [M db_mysql.so] my_fld.o In file included from my_fld.c:22:0: my_fld.h:37:2: error: unknown type name 'my_bool' my_bool is_null; Has anyone seen this before. Thank you

2 2

is_method not matching message type KDMQ breaking DMQ bus. (master branch)
by Asgaroth 02 Aug '18

02 Aug '18

Hi All, I've come across and oddity whereby my is_method("KDMQ") checks are not matching. I'm testing with kamailio master: version: kamailio 5.2.0-dev6 (x86_64/linux) a16272 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: a16272 compiled on 15:27:21 Jul 31 2018 with gcc 7.3.1 Sample routing logic: request_route { route("InitialMessageChecks"); exit; } route[InitialMessageChecks] { xlog("L_INFO", "Received message type $rm from $si:$sp on $Ri:$Rp"); if ( is_method("KDMQ") ) { xlog("L_INFO", "KDMQ check passed."); } if ( is_method("OPTIONS") ) { xlog("L_INFO", "OPTIONS check passed."); } if ( is_method("SUBSCRIBE") ) { xlog("L_INFO", "SUBSCRIBE check passed."); } } In the logs I am only receiving the 1st log line in the initial message checks route for message types of KDMQ, eg: INFO: <script>: Received message KDMQ from 1.1.1.1:5060 on 1.1.1.2:5060 INFO: <script>: Received message KDMQ from 1.1.1.3:5060 on 1.1.1.2:5060 I never seem to hit the is_method check log line when the method type is KDMQ. I do see the options/subscribe logs though: INFO: <script>: OPTIONS check passed. INFO: <script>: SUBSCRIBE check passed. nothing for KDMQ. Reason for asking is: this check never matches and breaks the dmq bus: if ( is_method("KDMQ") ) { if ( dmq_is_from_node() ) { dmq_handle_message(); exit; } else { exit; } } Any thoughts, or am I going crazy :) Thanks

3 4

Re: [SR-Users] Method Destination Variable
by Duarte Rocha 02 Aug '18

02 Aug '18

Greetings, I thought about that but i wasn't sure it would be safe. But, in this case, the Request-URI domain will be the only place where Kamailio will look to find out the destination right within the dialog right? I'm going the statelless way, so that might be the correct answer. Thanks for your help

2 2

get_out_socket(): no socket found
by Ilie Soltanici 02 Aug '18

02 Aug '18

Daniel, Not, those IP are in a different subnet. There is a firewall - running on a different device. Ping and SIP Messages are going very well from 2.2.2.2 to 8.8.8.8. I was able to solve this issue by disabling mhomed and set "listen" external interface with IP address 2.2.2.2 on top in the configuration file. Now, Kamailio is starting without any errors and is registering to the upstream provider, but I just I am wondering if it's possible to specify an outgoing socket for uac module in the configuration file? Thank you.

1 0

get_out_socket(): no socket found
by Ilie Soltanici 02 Aug '18

02 Aug '18

Daniel, Yes, IP Address has been edited, 2.2.2.2 and 8.8.8.8 are not real IP Address-es, this is just to show that those IP are external IP's. Regards, Ilie

2 1

Method Destination Variable
by Duarte Rocha 01 Aug '18

01 Aug '18

Greetings, I have a Kamailio server set up without Registers. I'm using Dispatcher module for balancing and selecting the destinations. I need to do some changes in the message according to the chosen destination. If it's the first INVITE, after Dispatcher operations, i can do it using $dd or $du variables. However, in later messages, like a BYE, i can't seem to access the method's destination IP. Is there any variable that i'm missing? I've tried $route_uri but it only returns the kamailio's own IP, since it's the only IP in the "Route" header. Thanks in advance, Cheers

2 1

get_out_socket(): no socket found
by Ilie Soltanici 01 Aug '18

01 Aug '18

Hi All, I'm unable to register an extension to the Upstream Provider by using uac module. Getting the following error: *Jul 31 15:49:39 k1-core1 k1[13146]: ERROR: <core> [core/forward.c:181]: get_out_socket(): no socket found* *Jul 31 15:49:39 k1-core1 k1[13146]: ERROR: <core> [core/forward.c:183]: get_out_socket(): no corresponding socket found for(udp:8.8.8.8:5060 <http://8.8.8.8:5060>)* *Jul 31 15:49:39 k1-core1 k1[13146]: ERROR: tm [ut.h:317]: uri2dst2(): no corresponding socket found for "svr04.nuacom.net <http://svr04.nuacom.net>" af 2 (udp:8.8.8.8:5060 <http://8.8.8.8:5060>)* *Jul 31 15:49:39 k1-core1 k1[13146]: ERROR: tm [uac.c:449]: t_uac_prepare(): no socket found* *Jul 31 15:49:39 k1-core1 k1[13146]: ERROR: uac [uac_reg.c:1161]: uac_reg_update(): failed to send request for [1050]* Kamailio is listening on 2 interfaces: [root@kamailio user]# /usr/local/sbin/kamcmd core.sockets_list { socket: { proto: udp address: 10.1.1.11 ipaddress: 10.1.1.11 port: 5060 mcast: no mhomed: no } socket: { proto: udp address: 2.2.2.2 ipaddress: 2.2.2.2 port: 5060 mcast: no mhomed: no } mhomed is enabled in the configuration file: *mhomed=1* I'm not using ' force_send_socket' anywhere in the configuration file. This is the uac module configuration: modparam("uac", "reg_keep_callid", 1) modparam("uac", "reg_contact_addr", "2.2.2.2") modparam("uac", "reg_db_url","cluster://kamailio") If i'm disabling mhomed from configuration file - i don't receive any errors - but then kamailio is trying to send the message to the external server (8.8.8.8) through internal interface (10.1.1.1). Thank You.

2 1

kamalio in virtual environment
by AL S 31 Jul '18

31 Jul '18

Hi, Is recent kamalio release interoperable in virtual environment? Thanks, Alex

5 5

How to get Formatted Current TIME (NOT CACHED)
by Alvaro Araos 31 Jul '18

31 Jul '18

Hi, If i call to $timef multiple times, always returns the same time string (CACHED), how i get formatted current time?? Example: $dlg_var(start)=$timef(%Y-%m-%d %H:%M:%S) // RETURNS "2018-07-28 18:44:35" .. some seconds later $dlg_var(start)=$timef(%Y-%m-%d %H:%M:%S) // RETURNS THE SAME DATE STRING "2018-07-28 18:44:35" Saludos, ALVARO ARAOS MÉNDEZ Director de Tecnología en FONOIP +56 2 2947 9000 alvaro(a)fonoip.com | www.fonoip.com [1] Moisés Chacón 46, El Monte, RM Links: ------ [1] http://www.fonoip.com

1 1

RTP Proxy
by Alex Tatham 31 Jul '18

31 Jul '18

Hello everyone, I'm in the process of moving approx. 100 Kamailio/Media Proxy (AG Projects) servers to Kamailio/RTPEngine. I have all the functionality in place and I have a handful of servers in production. I'm trying to monitor the RTP engine (using Munin) and I'm graphing the values returned from the RPC command rtpengine.get_hash_total. This value is slowly increasing over time and I'm trying to find out if this is something I should worry about. I want to make sure that sessions are not being left open and therefore the software will eventually run out of ports. To give some values, if I make 100 calls the call will be around 25 after those calls. >From all other tests I have done including a check through the verbose logs I can see the sessions are being closed at the end of the calls. Can anyone help with this at all or does anyone possibly have any better ideas? Thank you, Alex Alex Tatham Technical Director T:01233 220 943 E:alex.tatham@dmcplc.co.uk W:www.dmctechnologies.co.uk DMC Technologies formerly HP Technology Unit 5, Invicta Business Centre | Orbital Park | Ashford | Kent | TN24 0HB DMC Technologies is a trading name of Hobbs Parker 2000 Limited Registered Office: Romney House, Orbital Park, Monument Way, Ashford TN24 0HB Registered in England No.04027755

2 1

Is anyone having issues with dialog metrics when DMQ is enabled?
by Joel Serrano 30 Jul '18

30 Jul '18

Hi guys, Is anyone having issues gathering active/early dialog metrics from kamailio when DMQ is enabled? I have an open ticket in GH ( https://github.com/kamailio/kamailio/issues/1591) and I'm wondering if this could be happening to anyone else? So far the theory is that the counter goes below 0 and that creates the problem. I'd love to hear from you if you are experiencing this too. If any of you are using Kamailio 5.1 + DMQ and gather active/early dialog metrics using the stats.fetch or stats.get_statistics rpc commands, do you ever get values close to 64BIT ULONG_MAX? (18446744073709551615)?? Thanks! Joel.

2 3

Kamailio 5.0.X RPM packages/repo broken
by Patrick Wakano 27 Jul '18

27 Jul '18

Hello list, Hope you all doing fine! I've been trying to install Kamailio packages from the opensuse repo (as per https://www.kamailio.org/wiki/packages/rpms) but the installation is failing with this message: Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-ims-5.0.7-1.el7.cento FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.0.x-rpms/C…: [Errno 14] HTTP Error 416 - Requested Range Not Satisfiable --:--:-- ETA Trying other mirror. kamailio-5.0.7-1.el7.centos.x8 FAILED 77% [============================================================- ] 1.0 MB/s | 4.9 MB 00:00:01 ETA http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.0.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.0.x-rpms clean metadata Trying other mirror. I've successfully installed the 5.0.7 version on Friday (06/07) but I've noticed the packages have changed last Saturday (07/07/2018).... Given this error message looks like the repository is broken somehow.... Is anyone aware of such issue? I haven't tried the 5.1.X repo because my script was not migrated to 5.1.X yet.... Thanks, Kind regards, Patrick Wakano

3 13

NGN and NNI functionalities
by Duarte Rocha 27 Jul '18

27 Jul '18

Greetings, Does Kamailio has any module that deals with NGN and NNI ? Also, is there any variable or transformation that gives me access to a parameter in the User Part of an URI ? Best Regards, Duarte Rocha

2 1

No cdr after Kamailio 5.0.4 and 5.2 dev restart
by Kordován Szabolcs 27 Jul '18

27 Jul '18

Hi All, In March there was a discus about this issue. We tested the new 5.2 dev and 5.0.4 version of Kamailio and we experience that If we restart Kamailio during the call the cdr won't be stored in table acc_cdrs but the data will be inserted to table acc and data will be deleted from table dialog and dialog_vars. Did I omit the fix for it? Thanks for your help. Best regards, Szabolcs

2 3

Package hashes for Kamailio 5.1.4 rpms are wrong.
by Skip Morse 26 Jul '18

26 Jul '18

Hi, The package hashes for Kamailio 5.1.4 rpms are wrong. I actually saw this happen with 5.1.3 and then somehow I think it got corrected. Repo details: http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C… Filename: kamailio-5.1.4-2.el7.centos.x86_64.rpm Path: /repositories/home:/kamailio:/v5.1.x-rpms/CentOS_7/x86_64/kamailio-5.1.4-2.el7.centos.x86_64.rpm Size: 5.4M (5645948 bytes) Last modified: Thu, 19 Jul 2018 02:53:15 GMT (Unix time: 1531968795) SHA-256 Hash: aed812fa3ccf502634fe6de848eca2ce167ee70f6ace31aa1541a5787502667d SHA-1 Hash: ae0977ca32e6b5ba2a627e58948098f386639582 MD5 Hash: 68784949f33d6259420b50de5e70a4ec BitTorrent Information Hash: f743a7c45d50065084d40a314bd80e3ec143c384 What I get locally: $ wget http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… $ sha256sum kamailio-5.1.4-2.el7.centos.x86_64.rpm a21d2b6c98da041af85ebc0f8b6fe8525e8e247e84317c32db021e90ec43b209 kamailio-5.1.4-2.el7.centos.x86_64.rpm $ sha1sum kamailio-5.1.4-2.el7.centos.x86_64.rpm 9cbf6de153fb8ffc4913f0582d6e85535b44975f kamailio-5.1.4-2.el7.centos.x86_64.rpm $ md5sum kamailio-5.1.4-2.el7.centos.x86_64.rpm d36149f4bd91d0879c239b8e81264fb3 kamailio-5.1.4-2.el7.centos.x86_64.rpm Let me know if I should report this elsewhere. Thanks! -Skip

1 0

DMQ with dialog: KDMQ messages being looped back to the source
by Patrick Murphy 26 Jul '18

26 Jul '18

Hi, I am using Kamailio 5.1.1 with DMQ with the following config (relevant parts) on 2 nodes. Node A: modparam("dialog", "enable_dmq", 1) modparam("dmq", "server_address", "sip:no.de.a.ip:5060") modparam("dmq", "notification_address", "sip:myvoip.local") modparam("dmq", "multi_notify", 1) Node B: modparam("dialog", "enable_dmq", 1) modparam("dmq", "server_address", "sip:no.de.b.ip:5060") modparam("dmq", "notification_address", "sip:myvoip.local") modparam("dmq", "multi_notify", 1) "myvoip.local" is a DNS A record that resolves to both the IPs of A and B. I have the dmq_handle_message(); called if method is of type "KDMQ" in the request_route. Now, in order to test it, I only sent a few calls only on node A. KDMQ messages flowed to node B and dialog state replicated. But I noticed, node B was randomly sending one of the KDMQ dialog messages back to A and receiving a 500 "Server Internal Error" back from node A... (for a subset of those calls, not all). Why would this be the case when the nodes correctly recognize their local status by the "local: 1" set as seen from dmq.list_nodes Anyone else experienced this? What could I be doing wrong? Thanks, Pat.

2 5

FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service, ClueCon Presentation Slides
by Giovanni Maruzzelli 25 Jul '18

25 Jul '18

Hello fellow RTCers, just finished presenting about scaling videoconferencing, chat and moderation/direction controls in SIP, with web clients, deskphones, smartphone apps, server push (google fcm) via cordova and SIP.js at www.cluecon.com. Here's the slides of my presentation: http://178.79.181.125/Maruzzelli_FreeSWITCH_OpenSIPS_WebRTC_Pure_SIP_Video_… or, shorter, http://178.79.181.125/Maruzzelli_ClueCon_2018.pdf >From ClueCon blurb: FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service On top of traditional SIP deskphones and softphones, WebRTC let us extend the reach of SIP to all browsers, and to smartphone apps that wake up with a server push. OpenSIPS has superb SIP and WebRTC support and can protect, balance and scale FreeSWITCH insuperable conferencing, video MCU, media switching/mixing, SIP SIMPLE message processing capabilities. We will see how to build and implement such a complete platform, touching both on servers and on clients. Hint: SIMPLE, the SIP Instant Messaging Protocol, can be leveraged to transport statuses, commands and chats between participants and administrators of SIP audio/video calls/conferences. Giovanni MaruzzelliPrincipal at OpenTelecom.IT ------------------------------ Giovanni is a consultant for the Telco sector, developing software and training courses for FreeSWITCH, SIP, WebRTC and Kamailio. He is heavily engaged with FreeSWITCH, of which he wrote the interface with Skype and with cellular phones. An Internet tech pioneer, in 1996 Giovanni was cofounder of Italia Online, the most popular Italian portal and consumer ISP, and architect of its Internet technologies – www.italiaonline.it Then supervisor of Internet operations and architect of the first engine for paid access to www.ilsole24ore.com , the most read financial newspaper in Italy and to its databases (migrated from mainframe). After that, he was CEO of venture capital funded Matrice, developing Telemail unified messaging and multi language phone access to email (Text To Speech), and CTO of incubator funded Open4, an open source managed applications provider. Then he was for two years in Serbia as Internet and Telecommunication Investment Expert for World Bank – IFC. Since 2005 he’s based in Italy, and serves ICT and Telco companies worldwide. -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

3 4

sipdump excluding dmq exchanges
by Patrick Murphy 24 Jul '18

24 Jul '18

Hi Can we have sipdump only log SIP messages and not KDMQ? Thanks, Pat.

2 1

Error when using http_async_client $http_query_id var
by jenus＠cyberchaos.nl 24 Jul '18

24 Jul '18

Hello, I'm trying to build a async http call to a external server and was trying to use the $http_query_id var to identity the http call in the response route. But when i try the following: $http_req(suspend) = 0; $http_req(hdr) = "Content-Type: application/json"; $http_req(body) = "{'r_uri':'" + $rU + "', 'f_uri':'" + $fU + "'}"; http_async_query("http://10.10.10.1/", "ACC_HTTP_REPLY"); xlog("L_WARN", "ACC Query id: $http_query_id"); route[ACC_HTTP_REPLY] { xlog("L_WARN", "route[ACC_HTTP_REPLY]: received reply for query $http_query_id\n"); } I get errors in the kamailio log file : Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: <core> [core/pvapi.c:903]: pv_parse_spec2(): error searching pvar "http_query_id" Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [d/100] in [$http_query_id] at [13 (0)] Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: xlog [xlog.c:513]: xdbg_fixup_helper(): wrong format[route[ACC_HTTP_REPLY]: received reply for query $http_query_id#012] Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:5308 Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: <core> [core/pvapi.c:903]: pv_parse_spec2(): error searching pvar "http_query_id" Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [)/41] in [$(http_query_id)] at [15 (1)] Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: xlog [xlog.c:513]: xdbg_fixup_helper(): wrong format[route[ACC_HTTP_REPLY]: received reply for query $(http_query_id)#012] Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:5308 Should i escape the var $http_query_id in some way? I'm running kamailio 5.1.3. Thanks, Jan

4 4

Kamailio Calls Concurrency
by Abtin Mir 24 Jul '18

24 Jul '18

Hello Support, As a proxy server , How many concurrent calls can be processed by a Kamailio which is installed on a VM with 4 cores and 16GB of RAM? Your answer is much appreciated. -- *Regards,* *Abtin Mir*

3 4

Re: [SR-Users] NATHELPER issue
by Ali Taher 23 Jul '18

23 Jul '18

Thank you Konstantin, What I can’t understand is why media packets are sent correctly to the public IP , while early media (RBT) are sent on private. Regards, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru> Sent: Monday, July 23, 2018 1:38 PM To: Ali Taher <ataher(a)vanrise.com> Subject: Re[4]: [SR-Users] NATHELPER issue Original INVITE contains private IP in c= parameter, so you need to use STUN or TURN or ICE, something which is supported in your system or in your softphone. See my comments in the previous mail. Best regrads Понедельник, 23 июля 2018, 12:25 +03:00 от Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >: Hi Konstantin, INVITE: INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> X-AUTH-IP: Y.Y.Y.Y Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0 Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Max-Forwards: 69 Contact: <sip:967123456@192.168.26.3:31556;alias=Y.Y.Y.Y~31556~1> To: "9613159370"<sip:9613159370@X.X.X.X:5065> From: "+967123456"<sip:967123456@X.X.X.X:5065>;tag=5269152d Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 434 v=0 o=- 5 2 IN IP4 192.168.26.3 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.26.3 t=0 0 m=audio 8602 RTP/AVP 0 8 101 a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602 a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602 a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602 a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF 200 OK 200 OK Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> From: "+967123456" <sip:967123456@X.X.X.X:5065>;tag=5269152d To: "9613159370" <sip:9613159370@X.X.X.X:5065>;tag=agtHmrp3Dccrg Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Contact: <sip:9613159370@X.X.X.X:5060;transport=udp> User-Agent: FIKAR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 Remote-Party-ID: "Outbound Call" <sip:506251539613159370@X.X.X.X <mailto:506251539613159370@X.X.X.X> >;party=calling;privacy=off;screen=no v=0 o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X s=FIKAR c=IN IP4 X.X.X.X t=0 0 m=audio 24496 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 where: X.X.X.X is Kamailio IP Y.Y.Y.Y is origination public IP Regards, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru <mailto:piligrim_pk@mail.ru> > Sent: Monday, July 23, 2018 12:08 PM To: Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> > Subject: Re[2]: [SR-Users] NATHELPER issue Hello Ali, Could you show your latest INVITE and OK? Check you SDP bodies in those messages, if you still have there private IPs then you need to solve usual NAT traversal task in your media software (not proxy). Kamailio doesn't process media as you know. Of course you can try fix_nated_sdp function from nathelper module but it will not work for symmetric NAT because you fix SDP with public IP toward proxy but not toward UA. Your media software or hardware should support STUN or ICE protocols to work correctly via NAT. Best regards. Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> >: Thank you for your reply. ACK is now sent correctly to public IP , yet I have another issue with RBT where it’s still sent to private IP .Yet the call media is sent correctly to the public IP. There is something strange , as I hear two short different ringing tones then complete silence till the called party answer the call and RTP flow normally. How can I handle this case? Thanks, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru <file://e.mail.ru/compose/%3fmailto=mailto%253apiligrim_pk@mail.ru> > Sent: Friday, July 20, 2018 3:01 PM To: sr-users(a)lists.kamailio.org <file://e.mail.ru/compose/%3fmailto=mailto%253asr%252dusers@lists.kamailio.org> ; Ali Taher <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> > Subject: Re: [SR-Users] NATHELPER issue Hello Ali, ACK is sent by UAC to the Contact which is received in OK from UAS. So from my point of view you need to fix that Contact in OK on your proxy. My NATDETECT looks like: if (nat_uac_test("19")) { fix_nated_contact(); } I call it for requests and responses. Best regards. Konstantin Message: 23 Date: Fri, 20 Jul 2018 11:13:42 +0300 From: "Ali Taher" <ataher(a)vanrise.com <mailto:ataher@vanrise.com> > To: <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > Subject: [SR-Users] NATHELPER issue Message-ID: <070901d42001$939be2b0$bad3a810$(a)vanrise.com <mailto:070901d42001$939be2b0$bad3a810$@vanrise.com> > Content-Type: text/plain; charset="utf-8" Hello, I'm using Kamailio 4.2 as proxy with nathelper enabled. Yet , the ACK packet sent from the proxy to the origination's private IP. The ACK is sent as reply on the following 200 OK sent from the origination : 8m2EJN41BN/6WSIP/2.0 200 OK From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= X.X.X.X,SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <mailto:CbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 INVITE Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE P-Charging-Vector: icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX ;orig-ioi=MXXXXXXXXXXXXXX Content-Type: application/sdp Contact: <sip:172.16.45.65:5060;transport=UDP> Content-Length: 268 v=0 o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX s=- c=IN IP4 172.16.45.144 t=0 0 a=sendrecv m=audio 47588 RTP/AVP 18 96 c=IN IP4 172.16.45.144 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=maxptime:20 Following is the header of the sent ACK packet: Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 Via: SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc Max-Forwards: 69 From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <mailto:CbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 ACK Content-Length: 0 Where X.X.X.X is Kamailio server public IP. Following is part of my config file : route { route(NATDETECT); record_route(); if(!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } # Maybe some sanity_check() here. if(has_totag()) { if(loose_route()) { route(DLGURI); if(!t_relay()) sl_reply_error(); exit; } else { if(is_method("ACK")) { route(DLGURI); if(t_check_trans()) { t_relay(); } } else sl_send_reply("403", "Forbidden"); } exit; } ..... } route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } Can you please check why the ACK is still sent on private IP ? Thanks Ali Taher "Ali Taher" <ataher(a)vanrise.com <https://e.mail.ru/compose?To=ataher@vanrise.com> > С уважением, Константин Поляков. С уважением, Константин Поляков.

1 0

messages dumped in wrong order sometime
by Philip Kucheev 23 Jul '18

23 Jul '18

Hello there! Using kamailio 5.0.1 and msilo module for storing offline messages I noticed that sometimes, especially when a big number of messages dumping for one user, some of them can arrive to receiver in wrong order. Lets say we have 20 messages and the order of receiving could be next 1..18..17..19..20. That could be a reason of that ? Thanks, Philip

1 0

Forwarding to backup server
by Pali Rohár 23 Jul '18

23 Jul '18

Hello, can somebody help me how to configure Kamailio to route/forward requests to different (backup) server when Kamailio itself is not able to deliver them? I have working SIP gateway (on different host/port) which deliver received SIP messages to XMPP network and I want to achieve: Kamailio should try to send SIP messages directly to foreign SIP domains and when it fails (e.g. user not available or foreign SIP is not running), then forward message to that my SIP <--> XMPP gateway which will try to deliver it via XMPP. I guess that Kamailio should be able to configure for such custom routing/forwarding but I have not figured yet how to do it. I'm using Kamailio 4.4 with slightly modified default kamailio.cfg config. -- Pali Rohár pali.rohar(a)gmail.com

2 1

Re: [SR-Users] NATHELPER issue
by Ali Taher 23 Jul '18

23 Jul '18

Hi Konstantin, INVITE: INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> X-AUTH-IP: Y.Y.Y.Y Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0 Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Max-Forwards: 69 Contact: <sip:967123456@192.168.26.3:31556;alias=Y.Y.Y.Y~31556~1> To: "9613159370"<sip:9613159370@X.X.X.X:5065> From: "+967123456"<sip:967123456@X.X.X.X:5065>;tag=5269152d Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 434 v=0 o=- 5 2 IN IP4 192.168.26.3 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.26.3 t=0 0 m=audio 8602 RTP/AVP 0 8 101 a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602 a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602 a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602 a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF 200 OK 200 OK Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> From: "+967123456" <sip:967123456@X.X.X.X:5065>;tag=5269152d To: "9613159370" <sip:9613159370@X.X.X.X:5065>;tag=agtHmrp3Dccrg Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Contact: <sip:9613159370@X.X.X.X:5060;transport=udp> User-Agent: FIKAR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 Remote-Party-ID: "Outbound Call" <sip:506251539613159370@X.X.X.X>;party=calling;privacy=off;screen=no v=0 o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X s=FIKAR c=IN IP4 X.X.X.X t=0 0 m=audio 24496 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 where: X.X.X.X is Kamailio IP Y.Y.Y.Y is origination public IP Regards, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru> Sent: Monday, July 23, 2018 12:08 PM To: Ali Taher <ataher(a)vanrise.com> Subject: Re[2]: [SR-Users] NATHELPER issue Hello Ali, Could you show your latest INVITE and OK? Check you SDP bodies in those messages, if you still have there private IPs then you need to solve usual NAT traversal task in your media software (not proxy). Kamailio doesn't process media as you know. Of course you can try fix_nated_sdp function from nathelper module but it will not work for symmetric NAT because you fix SDP with public IP toward proxy but not toward UA. Your media software or hardware should support STUN or ICE protocols to work correctly via NAT. Best regards. Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >: Thank you for your reply. ACK is now sent correctly to public IP , yet I have another issue with RBT where it’s still sent to private IP .Yet the call media is sent correctly to the public IP. There is something strange , as I hear two short different ringing tones then complete silence till the called party answer the call and RTP flow normally. How can I handle this case? Thanks, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru <mailto:piligrim_pk@mail.ru> > Sent: Friday, July 20, 2018 3:01 PM To: sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> ; Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> > Subject: Re: [SR-Users] NATHELPER issue Hello Ali, ACK is sent by UAC to the Contact which is received in OK from UAS. So from my point of view you need to fix that Contact in OK on your proxy. My NATDETECT looks like: if (nat_uac_test("19")) { fix_nated_contact(); } I call it for requests and responses. Best regards. Konstantin Message: 23 Date: Fri, 20 Jul 2018 11:13:42 +0300 From: "Ali Taher" <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> > To: <sr-users(a)lists.sip-router.org <file://e.mail.ru/compose/%3fmailto=mailto%253asr%252dusers@lists.sip%252drouter.org> > Subject: [SR-Users] NATHELPER issue Message-ID: <070901d42001$939be2b0$bad3a810$(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253a070901d42001$939be2b0$bad3a810$@vanrise.com> > Content-Type: text/plain; charset="utf-8" Hello, I'm using Kamailio 4.2 as proxy with nathelper enabled. Yet , the ACK packet sent from the proxy to the origination's private IP. The ACK is sent as reply on the following 200 OK sent from the origination : 8m2EJN41BN/6WSIP/2.0 200 OK From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= X.X.X.X,SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 INVITE Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE P-Charging-Vector: icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX ;orig-ioi=MXXXXXXXXXXXXXX Content-Type: application/sdp Contact: <sip:172.16.45.65:5060;transport=UDP> Content-Length: 268 v=0 o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX s=- c=IN IP4 172.16.45.144 t=0 0 a=sendrecv m=audio 47588 RTP/AVP 18 96 c=IN IP4 172.16.45.144 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=maxptime:20 Following is the header of the sent ACK packet: Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 Via: SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc Max-Forwards: 69 From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 ACK Content-Length: 0 Where X.X.X.X is Kamailio server public IP. Following is part of my config file : route { route(NATDETECT); record_route(); if(!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } # Maybe some sanity_check() here. if(has_totag()) { if(loose_route()) { route(DLGURI); if(!t_relay()) sl_reply_error(); exit; } else { if(is_method("ACK")) { route(DLGURI); if(t_check_trans()) { t_relay(); } } else sl_send_reply("403", "Forbidden"); } exit; } ..... } route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } Can you please check why the ACK is still sent on private IP ? Thanks Ali Taher "Ali Taher" <ataher(a)vanrise.com <https://e.mail.ru/compose?To=ataher@vanrise.com> > С уважением, Константин Поляков.

1 0

Re: [SR-Users] NATHELPER issue
by Ali Taher 23 Jul '18

23 Jul '18

Hi Konstantin, INVITE: INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> X-AUTH-IP: Y.Y.Y.Y Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0 Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Max-Forwards: 69 Contact: <sip:967123456@192.168.26.3:31556;alias=Y.Y.Y.Y~31556~1> To: "9613159370"<sip:9613159370@X.X.X.X:5065> From: "+967123456"<sip:967123456@X.X.X.X:5065>;tag=5269152d Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 434 v=0 o=- 5 2 IN IP4 192.168.26.3 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.26.3 t=0 0 m=audio 8602 RTP/AVP 0 8 101 a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602 a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602 a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602 a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF 200 OK 200 OK Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> From: "+967123456" <sip:967123456@X.X.X.X:5065>;tag=5269152d To: "9613159370" <sip:9613159370@X.X.X.X:5065>;tag=agtHmrp3Dccrg Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Contact: <sip:9613159370@X.X.X.X:5060;transport=udp> User-Agent: FIKAR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 Remote-Party-ID: "Outbound Call" <sip:506251539613159370@X.X.X.X>;party=calling;privacy=off;screen=no v=0 o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X s=FIKAR c=IN IP4 X.X.X.X t=0 0 m=audio 24496 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Regards, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru> Sent: Monday, July 23, 2018 12:08 PM To: Ali Taher <ataher(a)vanrise.com> Subject: Re[2]: [SR-Users] NATHELPER issue Hello Ali, Could you show your latest INVITE and OK? Check you SDP bodies in those messages, if you still have there private IPs then you need to solve usual NAT traversal task in your media software (not proxy). Kamailio doesn't process media as you know. Of course you can try fix_nated_sdp function from nathelper module but it will not work for symmetric NAT because you fix SDP with public IP toward proxy but not toward UA. Your media software or hardware should support STUN or ICE protocols to work correctly via NAT. Best regards. Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >: Thank you for your reply. ACK is now sent correctly to public IP , yet I have another issue with RBT where it’s still sent to private IP .Yet the call media is sent correctly to the public IP. There is something strange , as I hear two short different ringing tones then complete silence till the called party answer the call and RTP flow normally. How can I handle this case? Thanks, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru <mailto:piligrim_pk@mail.ru> > Sent: Friday, July 20, 2018 3:01 PM To: sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> ; Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> > Subject: Re: [SR-Users] NATHELPER issue Hello Ali, ACK is sent by UAC to the Contact which is received in OK from UAS. So from my point of view you need to fix that Contact in OK on your proxy. My NATDETECT looks like: if (nat_uac_test("19")) { fix_nated_contact(); } I call it for requests and responses. Best regards. Konstantin Message: 23 Date: Fri, 20 Jul 2018 11:13:42 +0300 From: "Ali Taher" <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> > To: <sr-users(a)lists.sip-router.org <file://e.mail.ru/compose/%3fmailto=mailto%253asr%252dusers@lists.sip%252drouter.org> > Subject: [SR-Users] NATHELPER issue Message-ID: <070901d42001$939be2b0$bad3a810$(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253a070901d42001$939be2b0$bad3a810$@vanrise.com> > Content-Type: text/plain; charset="utf-8" Hello, I'm using Kamailio 4.2 as proxy with nathelper enabled. Yet , the ACK packet sent from the proxy to the origination's private IP. The ACK is sent as reply on the following 200 OK sent from the origination : 8m2EJN41BN/6WSIP/2.0 200 OK From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= X.X.X.X,SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 INVITE Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE P-Charging-Vector: icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX ;orig-ioi=MXXXXXXXXXXXXXX Content-Type: application/sdp Contact: <sip:172.16.45.65:5060;transport=UDP> Content-Length: 268 v=0 o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX s=- c=IN IP4 172.16.45.144 t=0 0 a=sendrecv m=audio 47588 RTP/AVP 18 96 c=IN IP4 172.16.45.144 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=maxptime:20 Following is the header of the sent ACK packet: Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 Via: SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc Max-Forwards: 69 From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 ACK Content-Length: 0 Where X.X.X.X is Kamailio server public IP. Following is part of my config file : route { route(NATDETECT); record_route(); if(!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } # Maybe some sanity_check() here. if(has_totag()) { if(loose_route()) { route(DLGURI); if(!t_relay()) sl_reply_error(); exit; } else { if(is_method("ACK")) { route(DLGURI); if(t_check_trans()) { t_relay(); } } else sl_send_reply("403", "Forbidden"); } exit; } ..... } route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } Can you please check why the ACK is still sent on private IP ? Thanks Ali Taher "Ali Taher" <ataher(a)vanrise.com <https://e.mail.ru/compose?To=ataher@vanrise.com> > С уважением, Константин Поляков.

1 0

KEMI: accessing "keyword" variables from core
by Enrico Bandiera 23 Jul '18

23 Jul '18

Hi, anyone how to access keyword variables like dst_port or src_port from kemi? (specifically python in our case). Thanks, Enrico.

2 3

Re: [SR-Users] NATHELPER issue
by Konstantin Polyakov 23 Jul '18

23 Jul '18

Hello Ali, ACK is sent by UAC to the Contact which is received in OK from UAS. So from my point of view you need to fix that Contact in OK on your proxy. My NATDETECT looks like: if (nat_uac_test("19")) { fix_nated_contact(); } I call it for requests and responses. Best regards. Konstantin >Message: 23 >Date: Fri, 20 Jul 2018 11:13:42 +0300 >From: "Ali Taher" < ataher(a)vanrise.com > >To: < sr-users(a)lists.sip-router.org > >Subject: [SR-Users] NATHELPER issue >Message-ID: < 070901d42001$939be2b0$bad3a810$(a)vanrise.com > >Content-Type: text/plain; charset="utf-8" > >Hello, > > > >I'm using Kamailio 4.2 as proxy with nathelper enabled. > > > >Yet , the ACK packet sent from the proxy to the origination's private IP. > > > >The ACK is sent as reply on the following 200 OK sent from the origination : > > > > >8m2EJN41BN/6WSIP/2.0 200 OK > >From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze > >To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 > >Via: SIP/2.0/UDP >X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= >X.X.X.X,SIP/2.0/UDP >X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN > >Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> > >Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX > >CSeq: 125698370 INVITE > >Accept: application/sdp > >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE > >P-Charging-Vector: >icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX >;orig-ioi=MXXXXXXXXXXXXXX > >Content-Type: application/sdp > >Contact: <sip:172.16.45.65:5060;transport=UDP> > >Content-Length: 268 > > > >v=0 > >o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX > >s=- > >c=IN IP4 172.16.45.144 > >t=0 0 > >a=sendrecv > >m=audio 47588 RTP/AVP 18 96 > >c=IN IP4 172.16.45.144 > >a=rtpmap:18 G729/8000 > >a=fmtp:18 annexb=yes > >a=rtpmap:96 telephone-event/8000 > >a=fmtp:96 0-15 > >a=maxptime:20 > > > >Following is the header of the sent ACK packet: > > > >Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 > >Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> > >Via: SIP/2.0/UDP >X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 > >Via: SIP/2.0/UDP >X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc > >Max-Forwards: 69 > >From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze > >To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 > >Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX > >CSeq: 125698370 ACK > >Content-Length: 0 > > > >Where X.X.X.X is Kamailio server public IP. > > > >Following is part of my config file : > > > >route { > >route(NATDETECT); > >record_route(); > > > > if(!mf_process_maxfwd_header("10")) { > > sl_send_reply("483", "Too Many Hops"); > > exit; > > } > > > > # Maybe some sanity_check() here. > > > > if(has_totag()) { > > > > if(loose_route()) { > > route(DLGURI); > > if(!t_relay()) > > sl_reply_error(); > > > > exit; > > } else { > > if(is_method("ACK")) { > > route(DLGURI); > > if(t_check_trans()) { > > t_relay(); > > } > > } else > > sl_send_reply("403", "Forbidden"); > > } > > exit; > > } > > > >..... > >} > > > > > >route[NATDETECT] { > >#!ifdef WITH_NAT > > force_rport(); > > if (nat_uac_test("19")) { > > if (is_method("REGISTER")) { > > fix_nated_register(); > > } else { > > add_contact_alias(); > > } > > setflag(FLT_NATS); > > } > >#!endif > > return; > >} > > > >route[DLGURI] { > >#!ifdef WITH_NAT > > if(!isdsturiset()) { > > handle_ruri_alias(); > > } > >#!endif > > return; > >} > > > >Can you please check why the ACK is still sent on private IP ? > > > >Thanks > >Ali Taher > > "Ali Taher" < ataher(a)vanrise.com >

2 1

Selectively enable / disable cdrs using acc module
by M S 23 Jul '18

23 Jul '18

Hi, I have a kamailio v5.0 setup where cdrs are enabled using acc module. Is there a way to enable / disable cdrs on per call basis in kamailio script? For example i want to allow cdrs only for offnet calls and disable them for onnet calls? May be a dialog flag configurable in module parameters to signal acc module whether or not to generate cdr for current dialog at the end of call?

2 5

Options for SIP traffic duplicating to third party tcp port
by Максим Баранов 23 Jul '18

23 Jul '18

Hello, what are the options for duplicating sip traffic to arbitrary tcp port in Kamailio?

3 5

forbidden domain not served
by eyas barhouk 22 Jul '18

22 Jul '18

Dears i'm using kamilio 5.2.0 as IMS with Zoiper client and RTP Engine when i'm trying to terminate a video call ,BYE message does not arrived to destination with error from S-cscf side "forbidden domain not served " has any one had the same issue before ? and is there any ideas about how to solve it ? ? thanks in advance

2 3

dialog module with dmq
by E. Schmidbauer 22 Jul '18

22 Jul '18

Is anyone using dialog module with DMQ to share dialogs across multiple kamailio instances? i've been testing it and it seems unreliable. sometimes the dialog data is sent from nodeA to nodeB, sometimes it is not. im wondering if im missing some settings or it's a bug. I tested latest master branch and version 5.1.4 Thanks! Emmanuel

3 2

$var(name) - Private memory variables
by Sergey Safarov 21 Jul '18

21 Jul '18

Hello I have faced when case when private var defined as $var(json_request) is share between different SIP messages. In my case first time this var is used from registration processing route. Then var with same name is used PUBLISH messge processing. And i be able to this value defined from registration processing. I expecting that vars is not shared between threads and message handling routes for var defined as $var(name). Is that not true? How i can prevent race condition between different message processing routes when two thread simultaneously change same variable? Sergey

3 5

uri==myself
by Karthik Srinivasan 20 Jul '18

20 Jul '18

Hello, Can someone explain to me why statement A returns True and statement B returns False. Statement A: is_myself("$ru") Statement B: uri == myself Isn't uri and $ru referencing the same data? With my current config i am expecting the is_myself to return True; i was expecting the same for Statement B but unfortunately it is returning False. Thanks, Karthik

2 11

how to configure ims in kamailio
by vinod mn 20 Jul '18

20 Jul '18

Hi what are the requirements to configure the ims to work with kamailio. -- Thanks and regards Vinod.M.N

3 2

NATHELPER issue
by Ali Taher 20 Jul '18

20 Jul '18

Hello, I'm using Kamailio 4.2 as proxy with nathelper enabled. Yet , the ACK packet sent from the proxy to the origination's private IP. The ACK is sent as reply on the following 200 OK sent from the origination : 8m2EJN41BN/6WSIP/2.0 200 OK From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= X.X.X.X,SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX CSeq: 125698370 INVITE Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE P-Charging-Vector: icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX ;orig-ioi=MXXXXXXXXXXXXXX Content-Type: application/sdp Contact: <sip:172.16.45.65:5060;transport=UDP> Content-Length: 268 v=0 o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX s=- c=IN IP4 172.16.45.144 t=0 0 a=sendrecv m=audio 47588 RTP/AVP 18 96 c=IN IP4 172.16.45.144 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=maxptime:20 Following is the header of the sent ACK packet: Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 Via: SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc Max-Forwards: 69 From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX CSeq: 125698370 ACK Content-Length: 0 Where X.X.X.X is Kamailio server public IP. Following is part of my config file : route { route(NATDETECT); record_route(); if(!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } # Maybe some sanity_check() here. if(has_totag()) { if(loose_route()) { route(DLGURI); if(!t_relay()) sl_reply_error(); exit; } else { if(is_method("ACK")) { route(DLGURI); if(t_check_trans()) { t_relay(); } } else sl_send_reply("403", "Forbidden"); } exit; } ..... } route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } Can you please check why the ACK is still sent on private IP ? Thanks Ali Taher

1 0

The problem of get_aor_hash function
by tyd 20 Jul '18

20 Jul '18

Dear all, I'm trying Kamailio 5.1.4 and IMS module. When registering to Kamailio IMS using the same ip and port for 30 users (different contact user part), the P-CSCF get the same hash number and aor value. The function get_aor_hash(_d, &_ci->via_host, _ci->via_port, _ci->via_prot) in ims_usrloc_pcscf pcontact.c seems the problem. Because hashing with host, port, prot will get the same hash value. Anyone can help this ?

3 3

dialog module : questions about dialog states
by Abdoul Osséni 19 Jul '18

19 Jul '18

Hello list, When the dialog state for a call is < 4, is it possible to use dlg_bye() function to end the call? Best regards Abdoul OSSENI Ingénieur DevOps chez Néo-Soft Co-Fondateur de ON SERVICES Tél : +33 601 135 167

2 5

Locking htable entries
by Alex Balashov 19 Jul '18

19 Jul '18

Hi, Sorry if this question has been answered before, but I'm rather puzzled as to the intent behind sht_lock() / sht_unlock(). Presumably, "thread"-safe / shm mutex locking is already implemented in htable query and mutation operations, otherwise htable would be unusable in a route script context. So what is the purpose of this higher-level lock? Is it to create a way to block config script execution in multiple processes as to serialise operations? Why would one want to do this under any ordinary circumstances? Perhaps it is a failure of my imagination... Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 4

executing python routine on demand
by Walter Martín Villalba 18 Jul '18

18 Jul '18

Hello, TL;DR How can I execute a routine in my Kamailio python module (kamailio_python.py) on demand? For example, using kamcmd from the command line. I know you can execute *app_python.reload *but I don't need to reload the whole python module, and furthermore, docs for this call say it's unstable and can even crash your kamailio instance. So my goal is to execute one single python routine on demand. https://www.kamailio.org/docs/modules/5.1.x/modules/app_python.html#app_pyt… Full explanation: I'm currently implementing OIDC token authentication for Kamailio. I have a python routine that extracts the token from a custom SIP header, and validates the token as per the OIDC spec (I'm using pyjwt plus other custom checks for this). In order to do offline validation of the token, you need the Identity Provider (IdP) Server's certificate, which I get in advance and cache in a local file on my Kamailio server. I also have an external python script which I can run to get the latest certificate from the IdP server and update it locally. The part I'm missing is how to tell Kamailio to reload the value of the certificate from the local cache, WITHOUT stopping kamailio or interrupting any sessions. This is needed in case the certificate is updated (e.g. due to expiration or compromise) on the IdP server; if you don't update it so that Kamailio can use the latest one, all token verification will fail (tokens signed by IdP server using latest private key corresponding to latest certificate, but Kamailio trying to validate the signature using the public key extracted from previous certificate -- this results in a signature error, as Kamailio needs to be using the public key corresponding to the private one that was used to sign the token). By being able to execute one single python routine on demand (e.g. using kamcmd), I can tell Kamailio to reload the certificate value from the local cache, which I just updated by running my other (external and unrelated to Kamailio software) python script. Hope this makes sense, and hopefully I can get some tips on how to achieve this. Thanks, Martín.

1 0

in-dialog requests
by Ali Taher 18 Jul '18

18 Jul '18

Hello, I'm using Kamailio as a proxy behind freeswitch to alter some packets (183 packets precisely). Here is my scenario: (---Customer---) --> ((port 5065)---Kamailio---) --> ((port5060)---Freeswitch---) --> (---Supplier---) Here is my request route : route{ if (is_method("INVITE")) { xlog("L_INFO","INSIDE request route $si \n"); insert_hf("X-AUTH-IP: $si\r\n"); } route(FWD); } reply_route { #xlog("L_INFO","INSDE reply \n"); if (status == "183") { change_reply_status(180, "Ringing"); exit; } } route[FWD] { #xlog("L_INFO","INSIDE FWD \n"); forward(X.X.X.X,5060); } Where X.X.X.X is the Freeswitch IP This script is working normally as long as the customer is sending the requests(INVITE,BYE,ACK,CANCEL..) . But when the supplier or freeswitch initiated the request , above script causes loop where the packet is bounced between kamailio and freeswitch. As I know , loose_route is used to handle such requests , but I'm not sure how to forward these packets to the customer after receiving them from freeswitch. I'm not sure also if what's I'm doing is statefull or stateless. Any help is appreciated. Thanks, Ali Taher

2 4

Kamailio dialog presence (BLF)
by Michal Škuta 18 Jul '18

18 Jul '18

Hello, I am trying make presence module work with Kamailio. Right now I am testing these modules: presence, presence_xml, presence_dialoginfo, pua and pua_dialoginfo. I had some success, the BLF is working only when the monitored extension is calling. When the monitored extension is calling, then the BLF changes its color (red). After end of the call the BLF changes color to green. My problem is when a User comes on-line then it sends SUBSCRIBE with *Event: dialog* and don't get notified of its subscribers state and BLF is not changed. I found similar problem in this mail list : https://lists.kamailio.org/pipermail/sr-users/2017-March/096314.html But is not working for me. Second question is about BLF and offline state. I can not found anything in RFC, there is no offline state, no way to turn off the BLF lamp from green to OFF. I found only this pdf from grandstream: http://www.grandstream.com/sites/default/files/Resources/GXP21x0_Eventlist_… mentioning offline state on page 8. Does anyone have experience with it?

2 2

Using the SIP related functions in a lib
by Bobby Mah 18 Jul '18

18 Jul '18

Hello all, Is it worth my time to try and create a SIP lib out of the SIP functionality available in Kamailio? Thanks for all your input, Bobby

2 1

SIP re-invite Packets
by Ali Taher 18 Jul '18

18 Jul '18

Hello, I'm using below routing logic to change the 183 reply to 180 and send it back to the origination. Noting that Kamailio is listening on 5065. route{ if (is_method("INVITE")) { xlog("L_INFO","INSIDE request route $si \n"); insert_hf("X-AUTH-IP: $si\r\n"); } route(FWD); } reply_route { #xlog("L_INFO","INSDE reply \n"); if (status == "183") { change_reply_status(180, "Ringing"); exit; } } route[FWD] { #xlog("L_INFO","INSIDE FWD \n"); forward(x.x.x.x,5060); } Below is the scenario , where the customer is sending packets to Kamailio , and Kamailio is forwarding them the supplier (with IP x.x.x.x) , then the supplier reply back to kamailio and kamailio relay the reply to the customer ----Customer---- --> ----Kamailio ---> ---Supplier--- I have a case where the supplier sends in some cases an in-dialog invite (Re-Invite) , which is considered I think as SIP-Request and thus as per above routing logic , kamailio is sending it back to the supplier , yet I need to send it to the customer. How can I solve this issue? Thanks Ali Taher

3 6

Error on Kamailio 4.3.7 and dialog
by Laura 18 Jul '18

18 Jul '18

Hi guys, on a test server with kamailio 4.3.7 statefull configuration with dialog I see a lot of messages like this.. Jul 16 15:20:26 vm-nextaitz-01 /usr/sbin/kamailio[31288]: ERROR: dialog [dlg_req_within.c:340]: send_bye(): terminating non-confirmed dialogs not supported Jul 16 15:20:26 vm-nextaitz-01 /usr/sbin/kamailio[31288]: ERROR: dialog [dlg_req_within.c:340]: send_bye(): terminating non-confirmed dialogs not supported Jul 16 15:20:26 vm-nextaitz-01 media-dispatcher[31555]: error: failed to end dialog: 500 Operation failed I'm not find the issue and the way for fix.. Any idea ? Regards Laura

3 8

ACK Cseq incrases after 404 on uac_auth()
by Yuriy Gorlichenko 17 Jul '18

17 Jul '18

Hi. I found some additional scenario when kamailio works incorrect with ACK Cseq after succesfull auth with uac_auth but unsuccesfull call: 1.Caller via kamailio as proxy (Cseq 1) 2.Kamailio resends call to provider with Auth method (Cseq 1) 3. Provider answers 401/407 (Cseq 1) 4. Kamailio inrases Cseq and sends invite with credentians (dialog track_cseq_updates = 1, uac_auth()) (Cseq 2) 5. Provider says - 404. not found (Cseq 2) 6. Kamailio resends INVITE to another server (lets say VM). It contans not tracked yet Cseq (Cseq 1) 7. VM server says 200 (Cseq 1) 8. Kamailio resends 200 to the caller (Сseq 1) 9. Caller says ACK (Cseq 1) 10. Kamailio incrases CSeq of ACK and resends t VM server (Сseq 2) So in this case kamialio tries to track Sceq because thining that this ACK belongs to the dialog that was made to the auth server So as I see here is a misunderstanding on the dialog module Any suggestions how to avoid it or exclude resent INVITE from the dialog module tracking?

2 4

RTPEngine serial forking with early media
by Alex Balashov 16 Jul '18

16 Jul '18

Hi, I've got a scenario where call branch #1 goes to an in-call announcement server; the message is played out via unidirectional early media (183+SDP), then that branch is disposed of with a 503. A subsequent call then goes out to the real SIP provider that will ultimately answer the call. There's some trouble handling this with RTPEngine: Two subsequent calls to rtpengine_offer() in a normal request route context will result in two SDP message lumps concatenated in the second INVITE (to the provider). This can be fixed by putting the rtpengine_offer() invocations into branch routes. However, the next problem is that then the identical offer goes to the second callee with no regard for RTPEngine's internal state, so RTPEngine ends up transmitting from a different port to the one that is declared in the SDP offer, requiring NAT latching from the provider to achieve proper symmetry in the return stream. This can be fixed with: rtpengine_offer("replace-origin replace-session-connection ICE=remove via-branch=1 internal external"); And in the reply handling: rtpengine_answer("replace-origin replace-session-connection ICE=remove via-branch=2"); An alternative to using via-branch[1] is to simply delete the call before initiating branch #2: rtpengine_delete("delete-delay=0"); rtpengine_offer("replace-origin replace-session-connection ICE=remove internal external"); However, both of these lead to the problem that caller receives two distinct SDP answers on the respective branches. I am not sure offhand whether this is a protocol semantics violation. On the one hand, RFC 3261 § 13.2.1 ("Creating the Initial INVITE") says: If the initial offer is in an INVITE, the answer MUST be in a reliable non-failure message from UAS back to UAC which is correlated to that INVITE. For this specification, that is only the final 2xx response to that INVITE. That same exact answer MAY also be placed in any provisional responses sent prior to the answer. The UAC MUST treat the first session description it receives as the answer, and MUST ignore any session descriptions in subsequent responses to the initial INVITE. So, I always understood that the first answer is the final answer, absent a session-updating request cycle. On the other hand, RFC 3960 ("Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)") Section 4 says: The application server model consists of having the UAS behave as an application server to establish early media sessions with the UAC. The UAC indicates support for the early-session disposition type (defined in [2]) using the early-session option tag. This way, UASs know that they can keep offer/answer exchanges for early media (early-session disposition type) separate from regular media (session disposition type). I take this, along with RFC 3959 Section 3, to imply an amendment to 3261 § 13.2.1, but I'm not sure. Regardless, I'm not convinced all UAs will handle this situation. So, what I would most prefer is a means of recycling the same answer port on the "back" side of RTPEngine while pivoting it on the front across branches/multiple offers. Is that possible? If so, how can it be achieved? Many thanks, -- Alex [1] As best as I understood how to use it; the documentation is a bit unclear! -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

Convert 183+SDP and 180+SDP into a single reply w/SDP
by yu＠yu-boot.ru 16 Jul '18

16 Jul '18

Hello. My uplink usually gives ringback as both 183 Progress with SDP and 180 Ringing also with (different) SDP. One of my clients can't handle that behavior and it just don't plays ringback at all. Is it possible to relay only last provisional response with SDP and drop anything else 183? Any ideas plz? Can't figure out even a logic itself. Thanks. -- sent from myMail for Android

5 7

Encapsulate and de-encapsulate SIP traffic
by William Phillips 16 Jul '18

16 Jul '18

I have a need to encapsulate SIP traffic depending on destination and obviously de-encapsulate traffic incoming from those destinations. The approach to dealing with outbound SIP traffic seems pretty straight forward, I create an appropriate routing block that determines if I need to encap the SIP message, if I do, I forward it off to my module. The incoming traffic isn't particularly complicated, I'm just not 100% sure on the way to give that de-encapsulated SIP message back to kamailio to be processed. It looks like I can achieve this using the TM module with the uac_req_t struct and tm_bind.request but I really only see it used for SIP text messages in modules like XMPP and SMS. Is this the appropriate approach to achieve what I need? Will

2 1

5.1.3 issue with rtpengine.reload
by Marcin Kowalczyk 16 Jul '18

16 Jul '18

Hi, I have Kam 5.1.3 running on top of Ubuntu Xenial docker image. Today I've added new RTPEngine set and tried to reload rtpengine from db but no luck. Command executes without any errors (but way too fast) root@dfa7aae0b6ef:/# time kamcmd rtpengine.reload real 0m0.002s user 0m0.000s sys 0m0.000s root@dfa7aae0b6ef:/# Last option would be restart but I would prefer not as has multiple WSS connections active. Do you have idea what I could try to reload rtpengine set?

2 1

File transfer with Kamailio IMS??
by Ashok kumar Marra 16 Jul '18

16 Jul '18

Hi, I m using cent-os 6.2, kamailio-5.1.2 and openims-hss. I am using rtpengine to relay voice, but I am unable to send files or chat messages using it. I looked at rtpengine logs, it was saying unknown codec for chat and file-transfer requests. Then, I thought of using Kamailio msrp , tried but no success. Does current Kamailio IMS support file transfer? If so, please give me suggestions on doing it. -- With Best Regards, Ashok Kumar.

2 1

rtpengine/kamailio teardown method
by Abdoul Osséni 16 Jul '18

16 Jul '18

Hello all, There are some rtpengine parameters that permit to send an RPC command to kamailio in order to terminate a dialog. - timeout = xx - b2b-url = http://xxxx:xx/RPC - xmlrpc-format = 1 Each rpc command sent to Kamailio exited with status 256. By checking the code of rtpengine, I see rtpengine seems to use "teardown" method. I think Kamailio does not know this method. Can you help me please ? Abdoul OSSENI Ingénieur DevOps chez Néo-Soft Co-Fondateur de ON SERVICES Tél : +33 601 135 167

2 1

Communication between Kamailio Proxy and Registrar Server
by amit Kumar 16 Jul '18

16 Jul '18

Hi All, As per our requirement, we want to configure one Kamailio as a " Kamailio PROXY" and 2nd Kamailio as a " Kamailio REGISTRAR". Whenever any INVITE request come at " Kamailio PROXY" we want to get the UA information from " Kamailio REGISTRAR " and send INVITE to particular UA. Could you please let us know how achieve this. Thanks, Amit

2 1

SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM)
by Aqs Younas 13 Jul '18

13 Jul '18

Greeting list, I am using two async functions(http_async_query) in configuration one for authentication and other one for push notifications. I am also using tsilio module to suspend the transaction and resumed it. I am resetting this flag 'T_ASYNC_SUSPENDED' before suspending the transaction. Everything is working fine except. When caller send CANCEL on the call, kamailio successfully completes sip flow at callee(B party) side but send back below packet towards caller. 2018/07/10 19:47:41.125893 172.31.21.87:5060 -> 39.38.156.111:59969 SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM ) Via: SIP/2.0/UDP 39.38.156.111:59969 ;branch=z9hG4bK-524287-1---91c6114330fe6293;rport=59969 To: <sip:923335550609@sip.talkhomeappcall.com ;transport=UDP>;tag=9fe2d4c84d97a4af81aa65ca8fc51b3d-62cb From: <sip:923135082344@sip.talkhomeappcall.com;transport=UDP>;tag=d72eaf4c Call-ID: VkAwxpSgkKSGn5dXaXgn5A.. CSeq: 2 INVITE Content-Length: 0 root@sip:~# kamailio -V version: kamailio 5.1.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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: unknown compiled on 08:06:26 May 2 2018 with gcc 4.9.2 Could someone please help me out how to fix this. Cancels are handles automatically so, i cannot hardcore anything to resolve it. Best Regards, Aqs Younas

3 8

Re: [SR-Users] sr-users Digest, Vol 158, Issue 12
by Ne Mo 13 Jul '18

13 Jul '18

Thanks Daniel! I'm folow: https://www.kamailio.org/docs/modules/stable/modules/ dispatcher.html#dispatcher.ex.install but my kamailio get errors: parameter <dstid_avp> of type <1> not found in module <dispatcher> parameter <attrs_avp> of type <1> not found in module <dispatcher> parameter <sock_avp> of type <1> not found in module <dispatcher> parameter <cnt_avp> of type <1> not found in module <dispatcher> I have no infor how to fix it Can you show me more detail about how to do: - *Fix error* * - Set the socket attribute to your private ip socket to do bridge from external to internal networks. * (just add to "attrs(str,opt)", and in my situation, it is my private IP of Kamailio server1 ? forexample: 1 sip:<PublicIP>:<Port> 0 0 duid=kamilioserver1;socket=<protocol>:<KamailioServer1_PrivateIP>:<Port>;my=xyz) (I don't understand what my=xyz mean..) * - Do i have to do something in Kamailio server2 and KamailioServer3 (or just create kamailio database each server and synchronyze kamailio database between them?)* *Thanks!* And thanks Mojtaba! I had read your recomment tutorial with OpenSIP, but do not know how to apply to my system. Have you got an other more clearly tutorial for this? On Thu, Jul 12, 2018 at 5:00 PM, <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. UAC register (Ben Hood) > 2. Re: UAC register (Daniel-Constantin Mierla) > 3. Re: UAC register (Daniel-Constantin Mierla) > 4. dispatcher module complete tutorial (SamSung TV) > 5. Re: UAC register (Ben Hood) > 6. Re: UAC register (Ben Hood) > 7. Re: dispatcher module complete tutorial (Mojtaba) > 8. Re: SIP/2.0 500 I'm terribly sorry, server error occurred > (1/TM) (Aqs Younas) > 9. Re: dispatcher module complete tutorial (Daniel-Constantin Mierla) > 10. Re: UAC register (Daniel-Constantin Mierla) > 11. Re: UAC register (Daniel-Constantin Mierla) > 12. Re: SIP/2.0 500 I'm terribly sorry, server error occurred > (1/TM) (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 11 Jul 2018 12:36:36 +0100 > From: Ben Hood <ben(a)relops.com> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] UAC register > Message-ID: <7B1A15CF-7BCF-4BCC-8DF0-20E2906BA1DA(a)relops.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > I’m wondering if there is a flag somewhere in the UAC module config that > prevents remote registration from happening. > > It looks like I’ve loaded the uacreg table but I can’t see any REGISTER > methods being sent from Kamailio to the remote peer. > > The remote registrants are loaded using this config: > > listen=udp:172.16.176.150:5060 > > loadmodule "db_postgres.so" > loadmodule "kex.so" > loadmodule "corex.so" > loadmodule "tm.so" > loadmodule "tmx.so" > loadmodule "sl.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "cfg_rpc.so" > loadmodule "counters.so" > loadmodule "uac.so" > > modparam("uac", "reg_db_url", DBURL) > modparam("uac", "reg_contact_addr", “172.16.176.150:5060") > modparam("uac", "reg_db_table", "uacreg") > modparam("uac", "reg_timer_interval", 10) > modparam("uac", "reg_retry_interval", 10) > > request_route { > route(RELAY); > } > > route[RELAY] { > > t_on_reply("MANAGE_REPLY"); > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > > And it looks like this has been consumed by Kamailio: > > $ kamcmd uac.reg_dump > { > l_uuid: 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f > l_username: 126231943099723776 > l_domain: 172.16.176.150 > r_username: 1000 > r_domain: 172.16.176.129 > realm: 172.16.176.129 > auth_username: 1000 > auth_password: password > auth_proxy: sip:172.16.176.129:5080 > expires: 0 > flags: 0 > diff_expires: -1531235076 > timer_expires: 0 > reg_init: 1531234841 > reg_delay: 0 > } > > Doing a refresh successfully reloads the uacreg data > > $ kamcmd uac.reg_refresh 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f > > I can get Kamailio to forward an OPTIONS message to the remote peer () to > verify network connectivity, e.g. : > > $ru = "sip:1000@172.16.176.129:5080"; > $fs = "udp:172.16.176.150:5060"; > > But Kamailio does not appear to generate REGISTER messages either after > first boot or as a result of the manual refresh. > > This is using an instance of Kamailio 5.0.1. Should I try with a newer > version? > > Is there a way to get extra debugging of the uac module? > > If I increase the debug verbosity, I get quite granular detail for the DB > traffic, but the uac module doesn’t appear to emit much diagnostics. > > Many thanks, > > Ben > > > > > >

2 1

UAC register
by Ben Hood 12 Jul '18

12 Jul '18

Hi, I’m wondering if there is a flag somewhere in the UAC module config that prevents remote registration from happening. It looks like I’ve loaded the uacreg table but I can’t see any REGISTER methods being sent from Kamailio to the remote peer. The remote registrants are loaded using this config: listen=udp:172.16.176.150:5060 loadmodule "db_postgres.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "counters.so" loadmodule "uac.so" modparam("uac", "reg_db_url", DBURL) modparam("uac", "reg_contact_addr", “172.16.176.150:5060") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 10) modparam("uac", "reg_retry_interval", 10) request_route { route(RELAY); } route[RELAY] { t_on_reply("MANAGE_REPLY"); if (!t_relay()) { sl_reply_error(); } exit; } And it looks like this has been consumed by Kamailio: $ kamcmd uac.reg_dump { l_uuid: 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f l_username: 126231943099723776 l_domain: 172.16.176.150 r_username: 1000 r_domain: 172.16.176.129 realm: 172.16.176.129 auth_username: 1000 auth_password: password auth_proxy: sip:172.16.176.129:5080 expires: 0 flags: 0 diff_expires: -1531235076 timer_expires: 0 reg_init: 1531234841 reg_delay: 0 } Doing a refresh successfully reloads the uacreg data $ kamcmd uac.reg_refresh 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f I can get Kamailio to forward an OPTIONS message to the remote peer () to verify network connectivity, e.g. : $ru = "sip:1000@172.16.176.129:5080"; $fs = "udp:172.16.176.150:5060"; But Kamailio does not appear to generate REGISTER messages either after first boot or as a result of the manual refresh. This is using an instance of Kamailio 5.0.1. Should I try with a newer version? Is there a way to get extra debugging of the uac module? If I increase the debug verbosity, I get quite granular detail for the DB traffic, but the uac module doesn’t appear to emit much diagnostics. Many thanks, Ben

2 3

dispatcher module complete tutorial
by SamSung TV 12 Jul '18

12 Jul '18

I want to use kamailio as load balancing server my topo: (Private IP) Kamailio server2 / / (public iP)Kamailio server1(load balance)(Private IP) \ \ (Private IP) Kamailio server1 Any body can recomment me some tutorial to do this? Thanks any advance!

3 2

Kamailio, forking and multiple answer
by Marino Mileti 10 Jul '18

10 Jul '18

Hi guys, i've a Kamailio with forking scenario using addbranch technique.Now I've a problem; i know it's very very very rare but it could be possible so it's better for me to find a solution. A -> B,C,D in forking parallel mode, so B C and D rings simultaneulsy. What is happen if B anser the call, and at the same time C decline the call?It seems that, using my cfg scripts, C sends DECLINE but this message never arrive to A because it receives a 200OK of B. How I can prevent this? I would like that if DECLINE is the first message, all other answers will be ignored (first answer win). Any help or comment? ThanksMarino

2 1

TLS CRL configuration
by Amarnath Kanchivanam 10 Jul '18

10 Jul '18

Hi All, I'm trying to configured kamailio as TLS server with below configuration (tls.cfg) and TLS server is started successfully. [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = ./sip/server.key certificate = ./sip/server.crt ca_list = ./bundle.crt crl = ./sip_crl.pem verify_depth = 9 [client:default] verify_certificate = no require_certificate = no TLS connection works fine. Later i have updated the sip_crl.pem with server certificate revoked details and performed tls.reload command to load the latest update. After this I expect any TLS client trying to establish TLS connection should fail, as the client and server certificates are signed by same authority and server certificate is revoked. But the clients are able to establish TLS connection without any errors. I'm not getting any traces to confirm CRL validation has been performed before accepting the TLS connection. Any advice would be help to proceed with evaluating CRL functionality. -Amar

3 5

CPS
by Sergiu Pojoga 10 Jul '18

10 Jul '18

Hi folks, Trying to implement CPS throttling based on this <https://lists.kamailio.org/pipermail/sr-users/2013-September/079713.html>article, having some trouble. modparam("htable", "htable", "rhs=>size=32;initval=0;autoexpire=300;" modparam("htable", "htable", "rhm=>size=32;initval=0;autoexpire=1800;") route[CPS] { $var(rateHashSec) = $si+":sec:"+$timef(%Y/%m/%d_%H_%M_%S); $var(rateHashMin) = $si+":min:"+$timef(%Y/%m/%d_%H_%M_00); xlog("L_INFO", "var(rateHashSec)=$var(rateHashSec); var(rateHashMin)=$var(rateHashMin)\n"); $var(ratePerSec) = $shtinc(rhs=>$var(rateHashSec)); $var(ratePerMin) = $shtinc(rhm=>$var(rateHashMin)); xlog("L_INFO", "var(ratePerSec)=$var(ratePerSec); var(ratePerMin)=$var(ratePerMin)\n"); $var(limitPerSec) = 1; $var(limitPerMin) = 1; if ($var(ratePerSec) > $var(limitPerSec) || $var(ratePerMin) > $var(limitPerMin)) { xlog("L_INFO", "CPS Limit on IP $si\n"); send_reply("503", "CPS Limit on IP $si"); exit; } } Logs show hashes are created but *$var(ratePerSec/Min)* won't increment no matter how many calls I place per second or minute: Jul 9 14:35:52 proxy /usr/local/sbin/kamailio[1484]: INFO: <script>: var(rateHashSec)=65.39.XX.XX:sec:2018/07/09_14_35_52; var(rateHashMin)=65.39.XX.XX:min:2018/07/09_14_35_00 Jul 9 14:35:52 proxy /usr/local/sbin/kamailio[1484]: INFO: <script>: var(ratePerSec)=0; var(ratePerMin)=0 Much obliged. --Sergiu

2 3

Delay in 100 Trying response for INVITE when the load is 70 INVITE's per sec
by vivek＠advaitamtech.com 09 Jul '18

09 Jul '18

Hi All, When we are using the kamailio to handle around 70 INVITE's per second, kamailio is sending 100 Trying with the delay of around 3-4sec for few INVITE's. How can I reduce the delay in 100 Trying? Also, I tried increasing the number of threads for SIP routing processes. This doesnt seem to help. Also tried "modparam("tm", "auto_inv_100", 0)" . This is not working. Please help. Thank you, Vivek.

3 2

RTPEngine-Recording-daemon
by Mojtaba 09 Jul '18

09 Jul '18

Hello, In RTPEngine recording-daemon, I enabled "forward-to" option to unix_socket. All things work right, I got metadata at first, then media packets (RTP-UDP-IP). The metadata for each calls is received just at first packet, but i want to know, if i have more than one concurrent active call in RTPEngine, How can i distinct which media packet was from which call? Thanks with Regards. --Mojtaba Esfandiari.S

2 2

ul.add | path & socket not correctly set
by Laurent Schweizer 09 Jul '18

09 Jul '18

Dear all, I have kamailio 5.0.6 and I have an issue to add permanent location When I add a location with : kamcmd ul.add location_ch demo(a)peoplefone.ch sip: demo@95.128.80.122:5060 0 0.0 . 0 128 6111 . udp:95.128.80.3:5060 then I get the following insertion: kamcmd ul.lookup location_ch demo@ { AoR: demo Contacts: { Contact: { Address: sip:demo@95.128.80.122:5060 Expires: permanent Q: 0.000000 Call-ID: dfjrewr12386fd6-343(a)kamailio.mi CSeq: 1 User-Agent: SIP Router MI Server Received: [not set] Path: . State: CS_SYNC Flags: 0 CFlags: 128 Socket: [not set] Methods: 6111 Ruid: ulcx-5b243cb5-67c6-31 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: 0 Keepalive: 0 Last-Keepalive: 0 Last-Modified: 0 } } As you can see the Path value is set to "." But it must be set to [not set] The Socket is also not set. Any idea ? BR Laurent

2 3

RFC: database unsigned number types
by Daniel-Constantin Mierla 07 Jul '18

07 Jul '18

Hello, starting here a discussion about an issue with the database API and signed/unsigned number type to see how to address it. So far, the database API supported only signed types for numbers, respectively DB1_INT (int in c) and DB1_BIGINT (long long in c). However, many database table columns are defined as UNSIGNED INT or UNSIGNED BIGINT. The database connector modules are mapping UNSIGED values retrieved from database over signed fields in DB API structure. The other way around is done as well: signed values are the inserted in the unsigned columns. There is no issue if the value in C is a positive number, however, if the value goes over MAX_UINT/2 (over 2147483647), it becomes negative and inserting the value in database results in an exception and 0 being stored instead. So far, I guess the issue was rarely exposed, if at all, because no report on it, even these data types for DB1 are since the SER project was started in 2001. As I looked at database definition schema, most of unsigned columns are for internal flags or ids (e.g., lcr_id), where I guess no large values were used or needed so far. However, it can bite at any time and needs to be addressed. So far, two solutions come in mind: 1) drop using UNSIGNED INT for those db columns, use only INT and do unsigned cast in the C code when reading and cast to int when writing. 2) update the db connector modules to support unsigned types -- I added support for them in DB API, but each db_* module has to be updated. The also each module that uses UNSIGNED DB columns must be updated 1) should be simpler, 2) more work but better in long term Any other ideas? Which of the options you prefer to go for? Not to forget: somehow related, probably we have to switch from int to long for PV number values, otherwise timestamp variables can go negative once unix timestamp approaches MAX_UINT/2 (still plenty of time, but should not be delayed for long ...). Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

2 1

sdpops / keep_codecs_by_name not modifying sdp unless i call msg_apply_changes
by Enrico Bandiera 06 Jul '18

06 Jul '18

Hi, I'm encountering an issue where calling keep_codecs_by_name does not modify the SDP on the INVITE unless I call the "evil function" msg_apply_changes Any help is welcome, Thanks, Enrico.

2 2

rtpengine installation dependencies
by Aqs Younas 05 Jul '18

05 Jul '18

Greetings list. This might not be related to Kamailio but I thought someone would be able to give a hand in resolving dependency during the installation of rtpengine on Debian 8 Jessie. It is complaining about below dependencies. root@debian-769mb-miami-01:/usr/src/rtpengine# dpkg-buildpackage dpkg-buildpackage: source package ngcp-rtpengine dpkg-buildpackage: source version 6.3.0.0+0~mr6.3.0.0 dpkg-buildpackage: source distribution unstable dpkg-buildpackage: source changed by Sipwise Jenkins Builder < jenkins(a)sipwise.com> dpkg-buildpackage: host architecture amd64 dpkg-source --before-build rtpengine dpkg-source: info: using options from rtpengine/debian/source/options: --extend-diff-ignore=.gitreview dpkg-checkbuilddeps: *Unmet build dependencies: debhelper (>= 10~) libbcg729-dev* dpkg-buildpackage: warning: build dependencies/conflicts unsatisfied; aborting dpkg-buildpackage: warning: (Use -d flag to override.) I tried to install debhelper from deb but complained from dh-autoreconf and that itself complained abount debhelpler. So, no one is being installed. Also, If someone can help me how to resolve dependency for libbcg729-dev too. Any pointer or hint or link to some installation guide would be more than welcome. Br, Aqs.

9 17

RTPEngine Recording does not work
by Mojtaba 05 Jul '18

05 Jul '18

Hi all, I installed RTPEngine (Version: 6.4.0.0+0~mr6.4.0.0 git-master-4eb80da) in my VM machine (Debian GNU/Linux 8 (jessie), i also installed kamailio 5.1. In kamailio routes, i have: route { ... #rtpengine_manage("record-call") rtpengine_manage(); start_recording(); ... } Here is all other configuration files: In /etc/default/ngcp-rtpengine-daemon file: RUN_RTPENGINE=yes CONFIG_FILE=/etc/rtpengine/rtpengine.conf CONFIG_SECTION=rtpengine PIDFILE=/var/run/ngcp-rtpengine-daemon.pid MANAGE_IPTABLES=yes TABLE=0 In /etc/default/ngcp-rtpengine-recording-daemon file: RUN_RTPENGINE_RECORDING=yes CONFIG_FILE=/etc/rtpengine/rtpengine-recording.conf CONFIG_SECTION=rtpengine-recording PIDFILE=/var/run/ngcp-rtpengine-recording-daemon.pid MUST_NFS=no NFS_HOST=192.168.1.1 NFS_REMOTE_PATH=/var/recordings NFS_LOCAL_MOUNT=/var/lib/rtpengine-recording # must match output-dir if used NFS_OPTIONS=hard,intr,tcp In /etc/rtpengine/rtpengine.conf file: [rtpengine] table = 0 interface = 192.168.122.200 listen-ng = 127.0.0.1:2223 recording-dir = /var/spool/rtpengine/ recording-method = proc In [rtpengine-recording] table = 0 # output-storage = db (use default) # output-format = mp3 (use default) # output-mixed = 1 spool-dir = /var/spool/rtpengine/metadata output-dir = /var/spool/rtpengine/recording But the recording is not work, I have these issues in syslog: [1530178531.860171] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command 'start recording' from 127.0.0.1:38654 [1530178531.860191] NOTICE: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Turning on call recording. [1530178531.860239] WARNING: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Call recording through /proc interface requested, but kernel table not open [1530178531.860262] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860269] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860275] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860281] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860287] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860292] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860303] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start recording' from 127.0.0.1:38654 (elapsed time 0.000112 sec) [1530178531.969040] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command 'start recording' from 127.0.0.1:39374 [1530178531.969069] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start recording' from 127.0.0.1:39374 (elapsed time 0.000004 sec) [1530178537.176906] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command 'delete' from 127.0.0.1:40258 [1530178537.177003] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Scheduling deletion of call branch '9052b05a' (via-branch '') in 30 seconds [1530178537.177026] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'delete' from 127.0.0.1:40258 (elapsed time 0.000084 sec) [1530178537.178368] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command 'start recording' from 127.0.0.1:40258 [1530178537.178392] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start recording' from 127.0.0.1:40258 (elapsed time 0.000004 sec) ^C[1530178543.752740] INFO: Version 6.4.0.0+0~mr6.4.0.0 git-master-4eb80da shutting down Also when i start RTPEngine, I have FATAL error about module xt_RTPENGINEer in syslog: Jun 28 05:44:58 debian8 systemd[1]: Starting NGCP RTP/media Proxy Daemon... Jun 28 05:44:58 debian8 ngcp-rtpengine-iptables-setup[10203]: modprobe: FATAL: Module xt_RTPENGINE not found. Jun 28 05:44:58 debian8 ngcp-rtpengine-iptables-setup[10203]: iptables: No chain/target/match by that name. Jun 28 05:44:58 debian8 ngcp-rtpengine-iptables-setup[10203]: ip6tables: No chain/target/match by that name. Jun 28 05:44:58 debian8 systemd[1]: Started NGCP RTP/media Proxy Daemon. Jun 28 05:44:58 debian8 rtpengine[10219]: [1530179098.989322] INFO: Generating new DTLS certificate Jun 28 05:44:59 debian8 rtpengine[10219]: [1530179099.033499] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED J Let me know what is the problem and how can i solve it? -- --Mojtaba Esfandiari.S

4 7

Video call recording and mobile push notifications
by Jeremy Renner 05 Jul '18

05 Jul '18

Hi, I would like to setup up my SIP server / PBX for my business, now we have some candidates: 1. Open source solution: - Asterisk PBX, - Freeswitch PBX - Kamailio - OpenSIPS 2. Business solution: - Brekeke PBX(https://www.brekeke.com - Vodia PBX(https://www.vodia.com) - 3CX PBX(https://www.3cx.com) - PortSIP PBX(https://www.portsip.com/portsip-pbx) *Below features are mandatory for our project:* - Video call recording (For the finance industry, the video recording is necessary) - Push notifications for mobile app - Multi-tenant support - Both Linux and Windows support (at 1st stage, we would like to run it on Windows server and migrate it to Linux server in the future if users increased), the Linux support is required, the Windows support is preferred. We have some questions: 1. Does the the Kamailio can works as a PBX ? 2. If yes, does the Kamailio support push notifications and video recording ? 3. Does the Kamailio can works for Multi-tenant ? 4. Does Kamailio support Windows ? So far according to our research, with the business solution: - The Vodia PBX, PortSIP PBX and brrekeke all are support Multi-tenant, the 3CX is not. - The 3CX and PortSIP support push notifications, - The PortSIP also provide client SDK, with 3CX we only see the 3CX provide client apps, does 3CX has client SDK provided ? - It's seems all these PBX are support video recording ? - The PortSIP PBX and 3CX both support Linux. Please help me to make the decision, base on your experiences, which one (open source or business solution) is good to us ? I'm really new to VoIP... Thanks in advance. Best regards,

3 3

Re-write From URI Domain ($fd) with source IP or relayed message
by Marrold 04 Jul '18

04 Jul '18

Hi All, For interoperability reasons I need to re-write the From URI Domain ($fd) in requests proxied by Kamailio to the source IP (socket) that Kamailio will send them via. I have experimented with onsend_route and can successfully discover the snd_ip, however it seems I am unable to rewrite the $fd, presumably because I am too late in the logic. Is this possible? I can hardcode IPs if required but would rather keep it dynamic if I can. Thanks in advance, Matthew

2 2

Unable to insert impu into table
by eyas barhouk 04 Jul '18

04 Jul '18

Hello community , on kamailio server i'm using the module "ims_usrloc_scsc" with the following configuration : ......................................................... modparam("ims_usrloc_scscf", "db_url", "mysql://eyas:eyas@localhost/scscf") modparam("ims_usrloc_scscf", "db_mode", 1) modparam("ims_usrloc_scscf", "enable_debug_file", 0) modparam("ims_usrloc_scscf", "matching_mode", 0) modparam("ims_registrar_scscf", "error_reply_code", 503) .......................................................... when i'm running kamailio and try to register i found the following : Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: ims_usrloc_scscf [usrloc_db.c:235]: db_insert_impurecord(): Unable to insert impu into table [sip:09876993998148@NET.SYRIATEL] Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: ims_usrloc_scscf [udomain.c:580]: insert_impurecord(): error inserting contact into db Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: ims_usrloc_scscf [impurecord.c:974]: update_impurecord(): Unable to insert new IMPU for <sip:09876993998148@NET.SYRIATEL> Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: ims_registrar_scscf [save.c:809]: update_contacts(): Unable to update impurecord for <sip:09876993998148@NET.SYRIATEL> Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: ims_registrar_scscf [cxdx_sar.c:253]: async_cdp_callback(): Error processing REGISTER Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: INFO: ims_registrar_scscf [cxdx_sar.c:84]: create_return_code(): created AVP successfully : [saa_return_code] - [-1] Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: <script>: SAR failure - error response sent from module Jul 04 16:04:04 S-CSCF [S-CSCF][3697]: INFO: ims_auth [cxdx_mar.c:79]: create_return_code(): created AVP successfully : [maa_return_code] - [-2] Jul 04 16:04:04 S-CSCF [S-CSCF][3814]: INFO: ims_auth [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed finding avp (avp_code = 297, vendor_id = 0) Jul 04 16:04:04 S-CSCF [S-CSCF][3814]: INFO: ims_auth [cxdx_mar.c:79]: create_return_code(): created AVP successfully : [maa_return_code] - [1] Jul 04 16:04:04 S-CSCF [S-CSCF][3694]: INFO: ims_auth [authorize.c:834]: authenticate(): uri=sip:NET.SYRIATEL;transport=TCP nonce=cae12e7c256b8a052ad9824ebb91fa07 response=ee82f8f8ad6bf4530d210144e Jul 04 16:04:04 S-CSCF [S-CSCF][3694]: INFO: ims_auth [authorize.c:886]: authenticate(): UE said: ee82f8f8ad6bf4530d210144eaf3c413 and we expect ee82f8f8ad6bf4530d210144eaf3c413 ha1 fd5e9c83ecc96ee Jul 04 16:04:04 S-CSCF [S-CSCF][3694]: INFO: ims_registrar_scscf [cxdx_sar.c:84]: create_return_code(): created AVP successfully : [saa_return_code] - [-2] Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: INFO: ims_registrar_scscf [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed finding avp Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: ERROR: ims_usrloc_scscf [usrloc_db.c:235]: db_insert_impurecord(): Unable to insert impu into table [sip:09876993998048@NET.SYRIATEL] Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: ERROR: ims_usrloc_scscf [udomain.c:580]: insert_impurecord(): error inserting contact into db Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: ERROR: ims_usrloc_scscf [impurecord.c:974]: update_impurecord(): Unable to insert new IMPU for <sip:09876993998048@NET.SYRIATEL> Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: ERROR: ims_registrar_scscf [save.c:809]: update_contacts(): Unable to update impurecord for <sip:09876993998048@NET.SYRIATEL> Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: ERROR: ims_registrar_scscf [cxdx_sar.c:253]: async_cdp_callback(): Error processing REGISTER could any one tell me where was my wrong and why it can't write on the DB & thanks in advance

1 0

Kamailio Anycast
by Mike Little 03 Jul '18

03 Jul '18

Hello, Sorry if this has already been answered but can Kamailio be configured to support clustering and use Anycast? We are looking to use Kamailio as a front end to Asterisk as the registar and presence server. If this is supported, are there any documents that explain how to configure this type of setup? Thanks!

2 1

IMS usage of Kamailio
by Akib Sayyed 03 Jul '18

03 Jul '18

Dear All I am willing to try Kamailio as IMS. before i try i would like to know following points - in Kamailio i see diameter support and authentication I-CSCF, P-CSCF, S-CSCF charging, QOS, ISC implemented but what about TAS do i need TAS seperately - I see that there is HOW-TO wiki , in how to wiki you are using FHoSS as HSS , can we use commercial/any other hss (afaik its using diameter so any HSS will do)? - Is there any configuration online available kamailio as IMS? - Are all IMS modules of Kamailio can be installed on one server or need each module on same machine (i am using IMS only for testing 1 or 2 calls at a time)

1 0

Having trouble getting presence working with Polycom VVX phones
by Justin Foley 03 Jul '18

03 Jul '18

Hey everyone. I've set up a kamailio server on a local dev box and running the phones on the same subnet as it to avoid any NAT issues. Seems to work fine, phones register & can call eachother and hear voice but the issue I'm running into is I can't seem to get presence to work. I've enabled it in the kamailio.cfg file with #!WITH_PRESENCE but if I go into "do not disturb" or try to call on phone1, it's not indicated on phone2. Also running ngrep on port 5060, I'm not seeing the anything regarding presence being sent over the SIP protocol.

2 2

$rs printed null
by Andrew Chen 03 Jul '18

03 Jul '18

Hi community, So I'm running 4.1.5 Kam version and I wanted to print out $rs variable. It printed out <null> right after the 183 session. Should it print something? What our server side sent is this: INVITE 100 183 488 or 487 And this is the output in kam logs: Jun 28 15:05:49 sjomainfnsg51 /usr/local/sbin/*kamailio*[15215]: INFO: <script>: [4975Fz-ses1-414b02f5R68422169(a)192.168.0.22][2][CSeq 1] fuze_for_mac/99.99.99999 Reply status <null> detected... any thoughts? Thanks -- Andy Chen -- *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 4

Multiple aliases in the Request Message
by Borislav Gergov 03 Jul '18

03 Jul '18

Hello, what is the proper way for kamailio nathelper module to handle multiple aliases in the request . For example We have ACK message that looped in the kamailio because is not forward to user phone: ACK sip:xxxxxx@10.10.10.32:38416;alias=192.168.129.135~5060~1;alias=192.168.205.32~38416~1 SIP/2.0 Via: SIP/2.0/UDP 192.168.146.174:46248 ;branch=z9hG4bK-524287-1---5b14eec905adf6ce Max-Forwards: 70 Route: <sip:mo@192.168.129.133;lr=on;ftag=c7d62d68;did=8a1.bcf2;rm=8> Route: <sip:mo@192.168.129.135;lr=on;ftag=c7d62d68;did=8a1.121> Route: <sip:mt@192.168.129.135;lr=on;ftag=c7d62d68;did=8a1.221> Route: <sip:mt@192.168.129.133;lr=on;ftag=c7d62d68;did=8a1.ccf2;rm=7> Contact: <sip:yyyyy@10.153.146.174:46248;transport=UDP> To: <sip:xxxxxx@domain.net;transport=UDP>;tag=dd9aaf1c From: <sip:yyyyy@domain.net;transport=UDP>;tag=c7d62d68 Call-ID: 5Zogf8chhpYlQRdpaZM4CA.. CSeq: 2 ACK User-Agent: Softphone Content-Length: 0

2 1

Auth and default kami config
by yu＠yu-boot.ru 03 Jul '18

03 Jul '18

Hello. Have a question on auth and default kamailio.cfg. Default behavior of kamailio on auth is: Auth initial requests, delete auth headers before forwarding to next hop. In-dialog requests are forwarded "as-is" so all auth related info is propagated to uplinks or just another Kami users. How should it be done correctly or "kamailio-way"? As for now, I just delete auth headers for all in-dialog requests, but I'm not sure it is correct. I looked how it works on softswitches like Yate. Registers and initial Invites are authenticated, ACK is not authenticated, but all other requests like reINVITE and BYE are. How is should be done? Thanks. -- sent from myMail for Android

2 1

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