sr-users September 2018

sr-users@lists.kamailio.org

72 participants
122 discussions

by Karlheinz Knapp

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

5 years, 8 months

Event: Kamailio Developers Meeting, Sep 27-28, 2018, in Dusseldorf, Germany

by Daniel-Constantin Mierla

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

5 years, 8 months

kamailio with evapi crashing on high volume

by Jayesh Nambiar

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

5 years, 8 months

Kamailio with CGNAT

by Amar Tinawi

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

5 years, 8 months

dlg.end_dlg on RTP timeout

by Igor Olhovskiy

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 years, 8 months

call_stats

by David Villasmil

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

5 years, 8 months

Via header line missing

by Karlheinz Knapp

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

5 years, 8 months

null $au

by David Villasmil

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

5 years, 8 months

Via header line missing

by Karlheinz Knapp

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

5 years, 8 months

dispatcher local socket

by Евгений Голей

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

5 years, 8 months

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sr-users September 2018