- 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,

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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 ?

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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/ >

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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

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