- sr-users - kamailio.org

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'.

2 1

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

2 14

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?

2 2

Record Routes
by Wilkins, Steve 11 Sep '18

11 Sep '18

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

2 4

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

2 9

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

3 3

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!

4 5

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,

2 5

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

1 0

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

1 1

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

2 5

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

4 11

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

5 13

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

3 2

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

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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

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