November 2014 - sr-users

Kamailio does not delete expired contacts
by Bui Minh Yen 24 Nov '14

24 Nov '14

Hi everyone, I am using Kamailio-4.1.5. I met a problem that Kamailio does not delete expired contact. I enable log file for module usrloc and see that, about 1 first hour, timer callback of usrloc module working fine (delete all expired contacts). But after that, it is not called --> Expired contacts in shared memory and database are not deleted. I configured usrloc module like that: # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "db_ops_ruid", 1) Could you help me to fix this problem! Thanks so much!

2 1

SNMP stats problem
by Muszalski Michał - Hurt 24 Nov '14

24 Nov '14

Good evening, I'm quite new in the Kamailio community so at the beginning I just want to say "Hello" to everyone. I have a following problem with SNMP in Kamailio. I want to collect data from Kamailio by SNMP. I've installed snmpd on the machine (Linux RHEL) and it's working. I've load snmpstats module in Kamailio and it starts up without error. I've configured Kamailio to use agentXsocket and enable agentx in SNMP. Port 705 is opened and netstat shows connection established between Kamailio and SNMP. Wireshark shows some packets going through. The problem is that there is no data during snmpwalk from Kamailio. Just standard data from SNMP, nothing from Kamailio. I've copied Kamailio MIBs to the proper directory, added it in the snmp.conf. Moreover I've tried to add -m +KAMAILIO-REG-MIB to snmpwalk. Still no data from Kamailio during snmpwalk. Any ideas? Best Regards, Michal Muszalski

3 2

RTP Between end points behind NAT or edit headers's sdp?
by Javier Ricke 24 Nov '14

24 Nov '14

------phona A-------kamailio---------asterisk----- OPTION 1: configure asterisk or kamailio i used asterisk, and install kamailio for traffic RTP can be send between end points that behind NAT router, and do not have to go through RTP proxy,, plz help! i think to the moment install kamailio, headrs'sdp fix IP private, but no!, how can fix it plz!!? regards! or OPTION 2: edit sip/sdp mi sip/sdp is [code] <--- SIP read from UDP:152.74.21.12:5060 ---> ACK sip:1001@152.74.21.12:6112 SIP/2.0 Via: SIP/2.0/UDP 152.74.21.12;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 190.164.204.227:41553 ;branch=z9hG4bK-d8754z-84f73c7e042445de-1---d8754z-;rport=41553 Max-Forwards: 16 Contact: <sip:JavierTren@190.164.204.227:41553;transport=UDP> To: <sip:1001@152.74.21.12;transport=UDP>;tag=as6e487bf1 From: <sip:JavierTren@152.74.21.12;transport=UDP>;tag=2b8fa52c Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY. CSeq: 2 ACK Proxy-Authorization: Digest username="JavierTren",realm="152.74.21.12",nonce="VGFnblRhZkIpJRpScSaEi795VKe4uof0",uri=" sip:1001@152.74.21.12 ;transport=UDP",response="116bf459c22231d0a770534d674b768d",algorithm=MD5 User-Agent: Zoiper r27147 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- set_destination: Parsing <sip:152.74.21.12;lr=on;ftag=2b8fa52c> for address/port to send to set_destination: set destination to 152.74.21.12:5060 Audio is at 12064 Adding codec 100002 (gsm) to SDP Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 152.74.21.12:5060: INVITE sip:JavierTren@190.164.204.227:41553;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 152.74.21.12:6112;branch=z9hG4bK3992226d;rport Route: <sip:152.74.21.12;lr=on;ftag=2b8fa52c> Max-Forwards: 70 From: <sip:1001@152.74.21.12;transport=UDP>;tag=as6e487bf1 To: <sip:JavierTren@152.74.21.12;transport=UDP>;tag=2b8fa52c Contact: <sip:1001@152.74.21.12:6112> Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY. CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.0 Session-Expires: 1800;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 256 v=0 o=root 1842142539 1842142540 IN IP4 192.168.1.8 s=Asterisk PBX 11.13.0 c=IN IP4 192.168.1.8 t=0 0 m=audio 8000 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [/code] and the field o and c i need to IP public y no private,, plz any?

2 1

database scheme generation and synchronization
by Måns Nilsson 21 Nov '14

21 Nov '14

Greetings, I'm fighting a number of errors in the 'acc' table like so: Nov 21 01:43:05 landsort postgres[53327]: [4-1] ERROR: column "time" of relation "acc" does not exist at character 78 That is, I believe, the default column name. Thing is, though, that the DDL that I used to build the table, is the one in in scripts/ and it does not does not contain that string: (on master but it is the same in 4.2 which is what I've got in production) $ for d in mysql oracle postgres ; do egrep '\btime\b' scripts/${d}/*create* ; done $ How are those files generated? How do I come about to submitting an useful patch? What is the desired datatype for the time column? -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 Edwin Meese made me wear CORDOVANS!!

2 2

rtpengine flags UDP/TLS/RTP/SAVP and UDP/TLS/RTP/SAVPF
by Juha Heinanen 21 Nov '14

21 Nov '14

Richard Fuchs wrote: > The command flags "RTP/AVP" etc are translated into the key > "transport-protocol=$X" within the control protocol. If you wish to > force usage of one of the UDP/TLS/... protocol, you should be able to do > so by spelling it out in your command. I believe this is not documented, > but should work. i tried and the offer and audio worked accordingly: Nov 21 05:37:15 rautu /usr/bin/sip-proxy[11113]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1 transport-protocol=UDP/TLS/RTP/SAVP trust-address) for consistency, it might be a good idea to make the dtls flags known to rtpengine module. -- juha

1 0

Kamailio creates tons of REDIRECT packages...
by Arnau Riba 20 Nov '14

20 Nov '14

Hello! I'm having this issue and I'm not able to solve it. I have searched it many times on the Internet but I found nothing, that's why I am asking here... I'm sorry I do not wanna cause any troubles but I need to get this done... This is my scenario, I want to make a call through a Kamailio server that redirects to another server. I have an User A registered to Server A and a User B registered to Server B, what I want to do is the following: User A calls User B using Server A, because Server A doesn't have the User B the Server A will redirect to Server B and make the call. I'm using Ekiga and of course Kamailio (and I want to make it work both ways). This is the code I'm using to do this in kamailio.cfg, I replaced this lines of code: *$avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } }* For this ones: *$avp(oexten) = $rU; if (!lookup("location")) { sl_send_reply("300",("REDIRECT")); rewritehostport("10.80.129.177"); forward(uri:host,uri:port); break; }* And it didn't work, then I putted the lines above as the following: *$avp(oexten) = $rU; if (!lookup("location")) { * * forward(uri:host,uri:port); rewritehostport("10.80.129.177");* * sl_send_reply("300",("REDIRECT")); break; }* And it worked but if I do a Wireshark Capture on the Server I get a lot of REDIRECT packets and my teacher says this is wrong, that I should only get 1. He did not help me and said I must learn and I want but I'm not able to understand what's going on, I need to do this because I want to pass the exam and not to fail... Again I'm sorry for taking your time, I'm asshamed. Thank you very much

2 1

Re: [SR-Users] rtpengine UDP/TLS/RTP/SAVP <=> RTP/AVP re-INVITE
by Richard Fuchs 20 Nov '14

20 Nov '14

On 11/18/2014 08:29 PM, Juha Heinanen wrote: > Richard, > > Enclosed find syslog that includes rtgpengine log at level 7 and also > pcap of the call. I made re-invite from baresip to sems once I saw this > in syslog: > > Nov 19 03:22:14 rautu rtpengine[29718]: [5315d797a9a5e7ce port 50761] DTLS-SRTP successfully negotiated > > I didn't include the to this message due to the pcap. Looking at your logs, I think what happened is that you've included the "trust-address" flag in the original answer, but haven't included it in the answer to the re-invite. This means that in the second answer, rtpengine erroneously saw a different endpoint than before (using the source address of the SIP packet), in response to this it also opened a new endpoint on its own side, which was then sent to the DTLS-SRTP client. This client then saw the new endpoint and correctly started a new DTLS connection, meanwhile rtpengine was still using the old keys and (I'm guessing) neither expected nor processed the new DTLS handshake, thus en/decrypted SRTP using the old keys, resulting in mangled RTP packets. So the short-term fix would be to include trust-address to avoid switching endpoints. As for DTLS restarts in rtpengine, this is something that I was just working on and DTLS restarts on changed endpoints should be working soon. cheers

2 1

Thread safety of shared variables
by Alex Balashov 19 Nov '14

19 Nov '14

Does setting $shv()s in script require lock()ing, or is it inherently thread-safe? Thanks! -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Kamailio Does NOT Forward Registration Requests To Asterisk.
by Mahmoud Ramadan Ali 19 Nov '14

19 Nov '14

Hi Dears, I'm trying to configure Kamailio as SBC in multi home mode for Asterisk by authenticating the inbound SIP registration requests,i'm following this tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb to achieve this goal. i have modified the necessary changes like the Asterisk DB URL and the SIP table name and Username and password column and verified the connection. My topology like this *Asterisk (192.168.100.10) <----Internal:192.168.100.1---->Kamailio<---External:192.168.50.1-----> SIP Phone (192.168.50.2)* But when trying to register a SIP phone Kamailio does NOT forward the authentication request to Asterisk and sends 401 Unauthorized error message.I've attached my config file if any one wants to check it and thanks in advance. Best Regards U 192.168.50.2:37297 -> 192.168.50.1:5060 REGISTER sip:192.168.50.1;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.50.2:37297 ;branch=z9hG4bK-d8754z-a46e0c7c9d98fe52-1---d8754z-;rport;transport=UDP. Max-Forwards: 70. Contact: <sip:1001@192.168.50.2:37297 ;rinstance=1d7c44dbcb8a7a2f;transport=UDP>. To: <sip:1001@192.168.50.1;transport=UDP>. From: <sip:1001@192.168.50.1;transport=UDP>;tag=1d222e19. Call-ID: NTc2NDBjMGQ2YWFmZjdmNWI0MzVmN2Y4NzYyODJlMTc.. CSeq: 2 REGISTER. Expires: 70. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE. Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri. User-Agent: Z 3.2.21357 r21367. Authorization: Digest username="1001",realm="192.168.50.1",nonce="VGfAuFRnv4wMvoTG7wA9tqYD9fgZDe3D",uri="sip:192.168.50.1;transport=UDP",response="8bbd01d879250585eafee4f510689f73",algorithm=MD5. Allow-Events: presence, kpml. Content-Length: 0. # U 192.168.50.1:5060 -> 192.168.50.2:37297 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.50.2:37297 ;branch=z9hG4bK-d8754z-a46e0c7c9d98fe52-1---d8754z-;rport=37297;transport=UDP. To: <sip:1001@192.168.50.1 ;transport=UDP>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe8b. From: <sip:1001@192.168.50.1;transport=UDP>;tag=1d222e19. Call-ID: NTc2NDBjMGQ2YWFmZjdmNWI0MzVmN2Y4NzYyODJlMTc.. CSeq: 2 REGISTER. WWW-Authenticate: Digest realm="192.168.50.1", nonce="VGfAuFRnv4wMvoTG7wA9tqYD9fgZDe3D". Server: kamailio (4.1.6 (i386/linux)). Content-Length: 0.

2 6

rtpengine and sdes
by Daniel-Constantin Mierla 19 Nov '14

19 Nov '14

Hello, I saw that rtpengine docs still advertise support for SDES SRTP and I was wondering if anyone was (is still) using it for decryption/encryption of this type of SRTP, mainly in scenarios like SRTP from a classic sip phone (e.g., snom) to another SIP endpoint wihtout srtp support. As I cc-ed Richard, another question is about the plans for the future, if at this moment is properly working/maintained, will it be kept for the future or the plan is to deprecate/remove/don't invest any effort in its maintenance? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Training/Tutorials?
by Jordan Rinke 19 Nov '14

19 Nov '14

For the life of me I can't find any in depth online resources on setting up and configuring Kamailio. Do anyone know of some good resources? Specifically I want to put this in front of a couple Asterisk servers so I need to learn how to connect a trunk to it, and then based on the call destination route it to the right Asterisk servers. I feel like this shouldn't be complicated I just don't understand where/how Kamailio is configured coming from only using Asterisk previously.

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rtpengine UDP/TLS/RTP/SAVP <=> RTP/AVP re-INVITE
by Juha Heinanen 19 Nov '14

19 Nov '14

when i make call from UDP/TLS/RTP/SAVP baresip to RTP/AVP sems, rtpengine gets called on initial invite/200 ok like this and audio works fine: Nov 18 02:46:06 rautu /usr/bin/sip-proxy[926]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1 RTP/AVP trust-address) Nov 18 02:46:06 rautu /usr/bin/sip-proxy[878]: INFO: ===== rtpengine_answer(ICE=force via-branch=2 trust-address) however, when baresip makes re-invite, rtpengine gets called using the same offer/answer flags, but sems clears the call. on syslog, i see this: Nov 18 02:46:59 rautu /usr/bin/sip-proxy[919]: INFO: Routing in-dialog INVITE <sip:127.0.0.1:5090;transport=udp> from <sip:jh@test.tutpro.com> Nov 18 02:46:59 rautu /usr/bin/sip-proxy[879]: INFO: ===== rtpengine_answer(ICE=force via-branch=2 replace-session-connection replace-origin) Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50444] SRTP output wanted, but no crypto suite was negotiated Nov 18 02:46:59 rautu /usr/bin/sip-proxy[919]: INFO: Routing in-dialog ACK <sip:127.0.0.1:5090;transport=udp> from <sip:jh@test.tutpro.com> Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50462] Discarded invalid SRTP packet: authentication failed Nov 18 02:46:59 rautu sems[20439]: [#b7193b70] [receive, AmRtpAudio.cpp:212] ERROR: decode() returned -4 Nov 18 02:46:59 rautu /usr/bin/sip-proxy[878]: INFO: ===== rtpengine_delete() Nov 18 02:46:59 rautu /usr/bin/sip-proxy[878]: INFO: Routing in-dialog BYE <sip:jh-0x9a95b00@192.98.102.30:5066;transport=tcp> from <sip:jh@as.test.tutpro.com> to <sip:192.98.102.30:46718;transport=TCP> based on gruu Nov 18 02:46:59 rautu rtpengine[29718]: [abd2bcf75f71af57 port 50463] Error parsing RTCP header: invalid packet type and on baresip console this: dtls_srtp: ---> DTLS-SRTP complete (audio/RTCP) Profile=AES_CM_128_HMAC_SHA1_80 dtls_srtp: incoming DTLS connect from 192.98.102.30:50524 dtls_srtp: verified sha-1 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 srtp: recv: failed to decrypt RTCP-packet (Unknown error 217) srtp: recv: failed to decrypt RTCP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) srtp: recv: failed to decrypt RTP-packet (Unknown error 217) sip:jh@as.test.tutpro.com: session closed: Connection reset by peer could it be that at some point during the re-invite, sems gets srtp audio and therefore clears the call? if so, is it a bug in rtpengine? -- juha

2 2

reREGISTER problems: contact is inserted instead updated
by Klaus Darilion 18 Nov '14

18 Nov '14

Hi! Kamailio 4.1.6 with default settings for usrloc and registrar modul and db_mode=1 (write through). Fritzbox sends a reREGISTER and Kamailio tries to insert the contact into the database, but this fails as this contact for this AoR already exists. So I think there are 2 scenarios: Kamailio false INSERTs instead of UPDATEs the contact, or the contact was deleted from usrloc, but not from the database. Frankly I see the in the mysql logfile that immediately after the failed INSERT the record is deleted from DB. Does anybody have a pointer what might be the problem here? AFAIS the client always uses the same Contact and the same call-id. only Cseq increases. Thanks Klaus

1 1

rtpengine flags UDP/TLS/RTP/SAVP and UDP/TLS/RTP/SAVPF
by Juha Heinanen 18 Nov '14

18 Nov '14

rtpengine README lists these rtp protocol and profile flags: + RTP/AVP, RTP/SAVP, RTP/AVPF, RTP/SAVPF rtpengine source code, however, seems to know also these DTLS ones: daemon/call.c: .name = "UDP/TLS/RTP/SAVP", daemon/call.c: .name = "UDP/TLS/RTP/SAVPF", can they be used in rtpengine_offer/answer calls even when they are missing from README? -- juha

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Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE
by Dan Christian Bogos 17 Nov '14

17 Nov '14

Hey Paul, Thanks for your feedback! In this case the call will come in via UDP (so it has 2 record route headers). I can still play tricks like reading the record-route headers but, I was in some way hoping that is the responsibility of loose_route() function to properly use the right protocol out of route headers since we are a proxy. Cheers, DanB On 14.11.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > Message: 3 > Date: Thu, 13 Nov 2014 16:55:18 +0530 > From: Varghese Paul<varghesepaul87(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE > Message-ID: > <CAM33Ch4furD1FM-ZNKfbQ0nXNQd88BmzW73gjjb=cVyuC0K=8w(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi Dan, > > You can check the transport value in the RURI and relay the SIP message as > tcp. > > if ($*rP* == "tcp"){ > t_relay_to_tcp() > } > > Regards > > Varghese Paul

3 4

Siremis Installation Issues
by Daniel Shoemaker 17 Nov '14

17 Nov '14

This is my first time posting to the mailing list. I'm currently running Kamailio 4.1, per the instructions on the website I got the tar bal for Siremis 4.1 s well. I have followed the install instructions at http://kb.asipto.com/siremis:install41x:main and I get all the way down to the bottom where I need to login to the web interface and when I try to use the default I get a very length openbiz error: [{"target":"ERROR","content":" \n[2014-11-17 11:51:15 (GMT)] An exception occurred while executing this script: \nError message: #8192, Non-static method BizSystem::dbConnection() should not be called statically, assuming $this from incompatible context<\/font> \nScript name and line number of error: \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php:423<\/font> \n *function:<\/b> errorHandler ( 8192, \"Non-static method BizSystem::dbConnection() should not be called...\", \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 423, Array(0) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/sysheader.inc 117\nfunction:<\/b> userErrorHandler ( 8192, \"Non-static method BizSystem::dbConnection() should not be called...\", \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 423, Array(0) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php 423\nfunction:<\/b> getDBConnection ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Lite.php 406\nfunction:<\/b> _run_search ( Array(2) ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/data\/BizDataObj_Lite.php 345\nfunction:<\/b> fetchRecords ( \"[username]='admin' and status='1'\", Array(0), 1 ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/service\/authService.php 98\nfunction:<\/b> authDBUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/service\/authService.php 66\nfunction:<\/b> authenticateUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/modules\/user\/form\/LoginForm.php 92\nfunction:<\/b> Login ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 310\nfunction:<\/b> invoke ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 110\nfunction:<\/b> dispatchRequest ( ) @ \/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php 32\nfunction:<\/b> include_once ( \"\/var\/www\/siremis-4.1.0\/openbiz\/bin\/BizController.php\" ) @ \/var\/www\/siremis-4.1.0\/siremis\/bin\/controller.php 6<\/div>\n------------------------------Please ask system administrator for help...<\/div>\n"}]* My web server is configured to use apache and I've checked the ownership on the files and folder. I did a chmod 775 starting at the siremis-4.1.0/ directory. I did a search and found that the openbiz folder needs to be right under the siremis-4.1.0/ folder which it is. I have tried reinstalling several times only to get the same result. I'm sure I'm missing something simple I just need to know what it is. I'm currently running CentOS 7, Kernel 3.10.0-123.9.3, php 5.6.2-3, mariadb 10.0.14-1, & httpd 2.4.6-18. Dan --

1 0

Kamailio TCP Problems
by Andrew Mason 17 Nov '14

17 Nov '14

Hi All, Looking for some help with a TCP issue in Kamailio. I am running load testing through the proxy with a topology of UAC > Proxy > UAS. With UDP it works fine (using SIPp scripts) - when I turn to TCP as the transport i get the following error in the kamailio logs: ERROR: <core> [tcp_main.c:4159]: handle_tcpconn_ev(): connect 192.168.0.19:5060 failed The IP address stated is the UAC which is sending an INVITE to the Proxy, which seems to send back a 100 TRYING to the UAC which is followed by repeated INVITE messages. It looks like the leg from the Proxy to the UAS is never established using TCP - however if I send the SIP messages to the Proxy via TCP and set the UAS to receieve in UDP Kamailio completes with TCP > Kamailio > UDP. When the UAC is set to send on TCP and UAS set to receive on TCP netstat shows the ports as listening. Can anyone please help?

1 0

Kamailio 4.2 TLS and csipsimple
by nadie 16 Nov '14

16 Nov '14

Hello. I installed K 4.2 with TLS. I can login users with csipsimple in TLS. but users can't call to other. in 5060 all correct. Thanks.

1 0

kamailio with pipelimit crashes when CPU load average >2
by Julia Boudniatsky 16 Nov '14

16 Nov '14

Hello, Sometimes kamailio with pipelimit crashes with segmentation fault, when CPU load average >2. The problem is not reproduced in the LAB (we used stress –c ). Please find attached core file. Thank you, Julia

1 0

401 vs 407 UAC auth issue
by Satish Patel 15 Nov '14

15 Nov '14

We have opensip configured with UAC auth to register SIP provide. I have configured two provider and both got registered but very interesting thing happened. Provider "A" sending me 407 challenge for authentication - Working Provider "B" sending me 401 for authentication - its failed to auth what is the solution here? does UAC module work with 401?

1 1

Secure WebSocket connection is getting dropped
by Kamrul Khan 14 Nov '14

14 Nov '14

Hi, I have a wired problem in my Kamailio. Over WSS, Kamailio is dropping the websocket connection right after it is getting the BYE from other side. So the client side is not getting any BYE. Over normal http it is working fine. Any idea why it might happen or where to look at? Thanks in advanced!

1 0

Non standard TLS port
by elektau＠yandex.ru 14 Nov '14

14 Nov '14

Hello All, I'm using Kamailio 4.2 running TLS only connection on non standard port 5065. I added the following line to the kamailio.cfg: listen=tls:111.222.333.444:5065 Domain SRV record as follows: _sips._tls.domain.com port 5065 When I try to connect using a softphone to domain.com login/password - "error while registering" When I connect using domain.com:5065 login/password - connection established successfully. So the question is: why I don't get redirected automatically to the specified port? If I use the default port for TLS (5061) in kamailio.cfg and in SRV, I can successfully connect using domain.com login/password Please help. Many Thanks!

4 5

cannot register UA when domain is not valid, using usrloc, auth_db and registrar modules
by Vik Killa 14 Nov '14

14 Nov '14

Hello, I'm trying to register 1004(a)3.local (yes i know 3.local is not a valid domain) and kamailio is responding with SIP/2.0 478 Unresolvable destination (478/SL) Why does kamailio require a valid domain? Is there any way to disable kamailio from looking up the domain (i don't see a reason to look it up since it's just a REGISTER request) Here is my cfg: if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "1"); exit; } And kamilio is responding with: Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: tm [t_fwd.c:1778]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: tm [t_fwd.c:1778]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: tm [t_fwd.c:1778]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Here is SIP ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.742489 192.168.207.162:63582 -> 192.168.207.156:5060 REGISTER sip:3.local SIP/2.0 Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport;branch=z9hG4bKPjbea638d5e7054e03bc4dfee6b6872275 Max-Forwards: 70 From: "1004" <sip:1003@3.local>;tag=ff09e20e549b4127a88ae6a6ec34e88b To: "1004" <sip:1003@3.local> Contact: <sip:98524760@192.168.16.120:63582 >;+sip.instance="<urn:uuid:7d4ed3bd-451a-4406-a8e2-80315408b7c5>" Call-ID: a7454fb3539740ffb385c146303a926f CSeq: 1 REGISTER Expires: 600 Supported: gruu User-Agent: Blink 0.9.1.2 (Windows) Content-Length: 0 ------------------end of packet ----------------------------- ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.743505 192.168.207.156:5060 -> 192.168.207.162:63582 SIP/2.0 478 Unresolvable destination (478/SL) Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport=63582;branch=z9hG4bKPjbea638d5e7054e03bc4dfee6b6872275;received=192.168.207.162 From: "1004" <sip:1003@3.local>;tag=ff09e20e549b4127a88ae6a6ec34e88b To: "1004" <sip:1003@3.local>;tag=98beed43ea484d57160c0a87164dfce1.ff17 Call-ID: a7454fb3539740ffb385c146303a926f CSeq: 1 REGISTER Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 ------------------end of packet ----------------------------- ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.743625 192.168.207.156:5060 -> 192.168.207.162:63582 SIP/2.0 478 Unresolvable destination (478/TM) Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport=63582;branch=z9hG4bKPjbea638d5e7054e03bc4dfee6b6872275;received=192.168.207.162 From: "1004" <sip:1003@3.local>;tag=ff09e20e549b4127a88ae6a6ec34e88b To: "1004" <sip:1003@3.local>;tag=b9d45a0d611ee9eea95e269f12755a03-ff17 Call-ID: a7454fb3539740ffb385c146303a926f CSeq: 1 REGISTER Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 ------------------end of packet ----------------------------- ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.750939 192.168.207.156:5060 -> 192.168.207.162:63582 SIP/2.0 478 Unresolvable destination (478/SL) Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport=63582;branch=z9hG4bKPj186bbf75b3964f06893742b853533302;received=192.168.207.162 From: "1004" <sip:1003@3.local>;tag=8b337cc975ab41c3b431a335a898128a To: "1004" <sip:1003@3.local>;tag=98beed43ea484d57160c0a87164dfce1.10fc Call-ID: fb19ff0b98f542849a249cd96dcd5f8c CSeq: 1 PUBLISH Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0

2 2

Allow Re-invite mid call
by Eric Koome 13 Nov '14

13 Nov '14

How do I process re-invite from sip provider mid call which check connection. At the moment Kamailio replies 404 not here, and call is dropped. Eric

2 3

avoid OPTION requests without authentication
by Julien 13 Nov '14

13 Nov '14

Hello, I received in my SIP clients an OPTION request forwarded by kamailio. The user agent of this request is 'sipvicious'. I want to avoid these kind of requests from unauthenticated users How can I do that ? -- Julien Barreau

2 3

Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE
by Dan Christian Bogos 13 Nov '14

13 Nov '14

Thanks Alex for your confirmation. In some way I was afraid about this ;). Have a good one! DanB

5 11

Kamailio - group presence
by Marino Mileti 13 Nov '14

13 Nov '14

Hi guys, I've to servers K1 e K2. On K2 I've registered two users on K1 and 2 users on K2. I would like that a user on K1 can subscribe to status of K2.... when there's a call on K2, it has to publish "On-call" status; if there isn't any ongoin call it has to publish "Online" status. I've tried to use uac module and send "PUBLISH" from scripts and it works but there are a lot of issue with SIP-If-Match header..so on K1 presentity table on db everytime a row is added and the old will be deleted when expires. I've tried also with PUA_MI...but using fifo on K2...so user on K1 should subscribe for a "virtual" user on K2. This isn't a good idea in case of network problem. Is there anyother way or do you have any suggests? ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-group-presence-tp132512.ht… Sent from the Users mailing list archive at Nabble.com.

1 0

messages delivered by MSILO are not cleared immediately
by Prasun Bheri 13 Nov '14

13 Nov '14

Hi Group, I have configured MSILO with following modparams. modparam("msilo", "db_url",DBURL) modparam("msilo", "offline_message","$rU is offline!") modparam("msilo", "extra_hdrs", "Msilo-msg: true\r\n") modparam("msilo", "outbound_proxy", MSILO_OUTBOUND_PROXY) modparam("msilo", "content_type_hdr","Content-Type: message/cpim\r\n") modparam("msilo", "extra_hdrs", "Content-Type: message/cpim\r\n") modparam("msilo", "add_date", 0) MSILO dumps all the messages to destination immediately after recipient registers and gets 200 ok for sent messages. The delivered messages are not cleared immediately and takes a minute to delete each message. Could some one explain why the delay is observed? In case you need to know I am using Kamailio 4.0.0 on CentOS 6.5 Thanks & Regards, Prasun

1 0

Kamailio Routing Logic
by Zaka Ul Isam 13 Nov '14

13 Nov '14

Hello Friends and Gurus: I am struggling with routing and need a bit of clarification WRT following: route { if ($fU=="601"&&is_method("INVITE")) record_route(); rewritehost("10.0.10.111"); //SS IP route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } * In the above stateful forwarding what is the rationale of calling route (1) within main route block? Could the t_relay statement afyer rewriting the host part of URI not have performed the stateful forwarding? ** In route[1] there is is only an expression abt logical not condition about t_relay() (if methods in this function are not invoked??) I wonder where exactly is stateful forwarding taking place? 1- Is it the default action? 2- Or in function sl_reply_error()? Thanks a lot in advance! Zaka ________________________________ Albtelecom & Eagle Mobile invite you to protect together the Environment. Please do not print the e-mail unless really necessary. Albtelecom & Eagle Mobile iu ftojnë të mbrojmë së bashku Mjedisin. Lutemi të mos e printoni këtë komunikim elektronik nëse nuk është vërtet i nevojshëm. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the addressee/s. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited. If you receive this e-mail in error, please notify the sender immediately and delete it! Albtelecom makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message may belong to sender alone and may not necessarily reflect the opinions of Albtelecom. Albtelecom shall bear no liability for any loss or damage caused by software or e-mail viruses. Ky mesazh dhe çdo informacion i transmetuar në përmbajtje te këtij mesazhi është konfidencial dhe është i destinuar vetëm për marrësin e destinuar. Nëse nuk jeni marrësi i destinuar, Ju bëjmë me dije se çdo përhapje, transmetim, kopjim apo përdorim i çdo informacioni është i ndaluar. Nëse e merrni këtë mesazh gabimisht, ju lutem kontaktoni urgjentisht nisësin e tij dhe fshijeni atë. Albtelecom nuk jep asnjë garanci për saktësinë apo plotësinë e informacionit në përmbajtje të këtij mesazhi dhe nuk mban asnjë përgjegjësi për informacionin e përmbajtur, transmetimin, marrjen, ruajtjen apo përdorimin e tij në çfarëdolloj mënyre. Mendimet e shprehura në këtë mesazh mund t'i përkasin vetë nisësit dhe nuk mund të reflektojnë domosdoshmërisht qëndrimet e Albtelecom. Albtelecom nuk do të mbaje asnjë përgjegjësi për humbje ose dëme te shkaktuara nga programet apo viruset.

2 4

result of param.value transformation when param does not exist?
by Juha Heinanen 13 Nov '14

13 Nov '14

wiki transformations page on param.value does not mention, what is the result of the transformation when named param does not exist. same with many other transformations. what is the value if referenced value does not exit? should it be mentioned on the wiki page? -- juha

2 4

topoh and uac_replace_from()
by Alex Balashov 13 Nov '14

13 Nov '14

Hi, If I use uac_replace_from() to alter the User part of the From URI: uac_replace_from("sip:$xavp(call=>src)@$fd"); it overwrites the domain concealment that is otherwise imparted by topoh, i.e. restores the original From domain. However, uac_replace_from() does not appear to offer the option to provide a replacement user part only, so I have to specify the domain. As far as I know, there is no means of accessing the modifications topoh has made, as this is transparent from the point of view of the config. Any way around this problem? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 3

kamailio does DNS lookup of from host when using TCP
by Vik Killa 12 Nov '14

12 Nov '14

Hello, I'm wondering why kamailio will try doing a DNS lookup of the from host with a REGISTER packet when using TCP? It doesn't do this with UDP. I always get this error since the host is not a valid domain: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Here is the full log with debug (error is last line) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [tcp_main.c:2294]: tcpconn_send_put(): tcp_send: found fd in cache ( 10, 0x7f860bd18938, 1) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f860bd18938 n=639 fd=10 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf=#012SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/TCP 192.168.16.120:53972;received=192.168.213.162;rport=53972;branch=z9hG4bKPjac894585917043d28da015d397178725;alias#015#012From: "1003" <sip:1003@3.local>;tag=dbc7a9087cb84006b65d891e5ad1f5ea#015#012To: "1003" <sip:1003@3.local>;tag=ey7D0K9DX49DF#015#012Call-ID: ee72194a1b8e4effa2948e237d804b90#015#012CSeq: 1 REGISTER#015#012User-Agent: AMPSwitch#015#012Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE#015#012Supported: timer, path, replaces#015#012WWW-Authenticate: Digest realm="3.local", nonce="cdf32dde-ab14-4265-b491-bee703e83255", algorithm=MD5, qop="auth"#015#012Content-Length: 0#015#012#015#012 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: tm [t_reply.c:1663]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3026]: DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0xa2c420, 6, 2, 0x7f860bd2dbe0), fd_no=1 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKPj19019d19feaa4a06ac1d6cdc31a2e05d>; state=6 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 237, <alias> = <n/a>; state=16 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts... Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=10 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [27]; uri=[sip:1003@3.local] Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body ["1003" <sip:1003@3.local>#015#012] Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> <PUBLISH> Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=757 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: textops [textops.c:2478]: has_body_f(): content type is 196613 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: siputils [checks.c:103]: has_totag(): no totag Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=10 global id=9 T start=0xffffffffffffffff Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=12001, isACK=0 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=10 global id=10 T end=(nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=10 , global msg id=10 , T on entrance=(nil) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=12001, isACK=0 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [md5utils.c:67]: MD5StringArray(): DEBUG: MD5 calculated: d71ddc300035dfaa15e181fe4d6a9aa7 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:566]: _dns_hash_find(): dns_hash_find(_sip._tcp.3.local(17), 33), h=867 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [resolve.c:770]: get_record(): get_record: lookup(_sip._tcp.3.local, 33) failed Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:894]: dns_cache_mk_bad_entry(): dns_cache_mk_bad_entry(_sip._tcp.3.local, 33, 60, 1) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:827]: dns_cache_add(): dns_cache_add: adding _sip._tcp.3.local(17) 33 (flags=1) at 867 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:566]: _dns_hash_find(): dns_hash_find(3.local(7), 1), h=184 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [resolve.c:770]: get_record(): get_record: lookup(3.local, 1) failed Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:894]: dns_cache_mk_bad_entry(): dns_cache_mk_bad_entry(3.local, 1, 60, 1) Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: DEBUG: <core> [dns_cache.c:827]: dns_cache_add(): dns_cache_add: adding 3.local(7) 1 (flags=1) at 184 Nov 12 15:45:19 ua01-proxy01 /usr/sbin/kamailio[3025]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local"

2 4

UDP-TCP bridging in-dialog SUBSCRIBE
by Dan Christian Bogos 12 Nov '14

12 Nov '14

Hey Guys, I have the following scenario: UA1 (UDP) -> Kamailio Proxy -> (TCP) UA2 As per the signaling sample, I would expect that in-dialog SUBSCRIBEs are bridged between UDP and TCP network but for some reason this does not happen (the subscribe comes in over UDP and leaves still over UDP although tcp is present in the route headers). Are we somehow forced to mangle contact header and add transport parameter to it or the record-route headers should serve for this purpose? Any tip would be highly appreciated. Ta, DanB Bellow the traces for initial SUBSCRIBEs as well as in-dialog ones (info mangled to protect privacy). ##################################### Normally routed request (non-dialog one, correctly bridged) ##################################### """ # U 2014/11/12 19:04:51.293887 pub_ip_ua:54705 -> pub_ip_ep:5060 SUBSCRIBE sip:user1@fakedomain.com:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.243.82:5062;branch=z9hG4bK1114302228. From: "user2" <sip:user2@fakedomain.com>;tag=1003326118. To: <sip:user1@fakedomain.com>. Call-ID: 1273081718. CSeq: 2 SUBSCRIBE. Contact: <sip:user2@192.168.243.82:5062>. Accept: application/dialog-info+xml. Max-Forwards: 70. User-Agent: Tiptel IP 284 6.70.13.18 0015651b574f. Expires: 60. Event: dialog. Content-Length: 0. . # T 2014/11/12 19:04:51.295706 pub_ip_ep:34052 -> 192.168.51.168:5060 [AP] SUBSCRIBE sip:user1@192.168.51.168:5060;transport=tcp SIP/2.0. Record-Route: <sip:pub_ip_ep;transport=tcp;r2=on;lr;nat=yes>. Record-Route: <sip:pub_ip_ep;r2=on;lr;nat=yes>. Via: SIP/2.0/TCP pub_ip_ep;branch=z9hG4bKe26f.3dce66bd1976447b88f6096e926ce68c.0. Via: SIP/2.0/UDP 192.168.243.82:5062;rport=54705;received=pub_ip_ua;branch=z9hG4bK1114302228. From: "user2" <sip:user2@fakedomain.com>;tag=1003326118. To: <sip:user1@fakedomain.com>. Call-ID: 1273081718. CSeq: 2 SUBSCRIBE. Contact: <sip:user2@192.168.243.82:5062;alias=pub_ip_ua~54705~1>. Accept: application/dialog-info+xml. Max-Forwards: 65. User-Agent: Tiptel IP 284 6.70.13.18 0015651b574f. Expires: 60. Event: dialog. Content-Length: 0. . # T 2014/11/12 19:04:51.300350 192.168.51.168:5060 -> pub_ip_ep:34052 [AP] SIP/2.0 202 Accepted. Via: SIP/2.0/TCP pub_ip_ep;branch=z9hG4bKe26f.3dce66bd1976447b88f6096e926ce68c.0;rport=34052. Via: SIP/2.0/UDP 192.168.243.82:5062;rport=54705;received=pub_ip_ua;branch=z9hG4bK1114302228. Record-Route: <sip:pub_ip_ep;transport=tcp;r2=on;lr;nat=yes>. Record-Route: <sip:pub_ip_ep;r2=on;lr;nat=yes>. From: "user2" <sip:user2@fakedomain.com>;tag=1003326118. To: <sip:user1@fakedomain.com>;tag=PFxfnedC0jP7. Call-ID: 1273081718. CSeq: 2 SUBSCRIBE. Contact: <sip:user1@192.168.51.168:5060>. Expires: 60. User-Agent: Sipean-AS 1.2.10. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: active;expires=60. Content-Length: 0. . ## U 2014/11/12 19:04:51.300745 pub_ip_ep:5060 -> pub_ip_ua:54705 SIP/2.0 202 Accepted. Via: SIP/2.0/UDP 192.168.243.82:5062;rport=54705;received=pub_ip_ua;branch=z9hG4bK1114302228. Record-Route: <sip:pub_ip_ep;transport=tcp;r2=on;lr;nat=yes>. Record-Route: <sip:pub_ip_ep;r2=on;lr;nat=yes>. From: "user2" <sip:user2@fakedomain.com>;tag=1003326118. To: <sip:user1@fakedomain.com>;tag=PFxfnedC0jP7. Call-ID: 1273081718. CSeq: 2 SUBSCRIBE. Contact: <sip:user1@192.168.51.168:5060>. Expires: 60. User-Agent: Sipean-AS 1.2.10. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: active;expires=60. Content-Length: 0. """ ###################### NOT BRIDGED, IN-DIALOG requests: ###################### """ # U 2014/11/12 19:46:18.448550 pub_ip_ua:54705 -> pub_ip_ep:5060 SUBSCRIBE sip:user1@192.168.51.168:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.243.82:5062;branch=z9hG4bK459444805. Route: <sip:pub_ip_ep;r2=on;lr;nat=yes>. Route: <sip:pub_ip_ep;transport=tcp;r2=on;lr;nat=yes>. From: "user2" <sip:user2@fakedomain.com>;tag=1902678123. To: <sip:user1@fakedomain.com>;tag=2518urgZJFet. Call-ID: 2104228355. CSeq: 3 SUBSCRIBE. Contact: <sip:user2@192.168.243.82:5062>. Accept: application/dialog-info+xml. Max-Forwards: 70. User-Agent: Tiptel IP 284 6.70.13.18 0015651b574f. Expires: 60. Event: dialog. Content-Length: 0. . # U 2014/11/12 19:46:18.449161 pub_ip_ep:5060 -> 192.168.51.168:5060 SUBSCRIBE sip:user1@192.168.51.168:5060 SIP/2.0. Via: SIP/2.0/UDP pub_ip_ep;branch=z9hG4bK7de2.a9aed4070706f93b25b4d2d3d7963a19.0. Via: SIP/2.0/UDP 192.168.243.82:5062;rport=54705;received=pub_ip_ua;branch=z9hG4bK459444805. From: "user2" <sip:user2@fakedomain.com>;tag=1902678123. To: <sip:user1@fakedomain.com>;tag=2518urgZJFet. Call-ID: 2104228355. CSeq: 3 SUBSCRIBE. Contact: <sip:user2@192.168.243.82:5062;alias=pub_ip_ua~54705~1>. Proxy-Authorization: Digest username="user2", realm="fakedomain.com", nonce="VGOr71RjqsMDoDyEqMnELO4fotFJLYhV", uri="sip:user1@192.168.51.168:5060", response="12fbfa50428b37bd55dbb4b51f8d5cba", algorithm=MD5. Accept: application/dialog-info+xml. Max-Forwards: 65. User-Agent: Tiptel IP 284 6.70.13.18 0015651b574f. Expires: 60. Event: dialog. Content-Length: 0. """

2 1

TLS/TCP sipping
by Petr Wozniak 11 Nov '14

11 Nov '14

Hello, Base on my testing OPTION pings can be sent only to UDP nated clients. So it is impossible with kamailio to send OPTION pings to TCP/TLS nated clients? I found doc for some version of nathelper module where the natping _tcp parameter can be set by modparam("nathelper", "natping_tcp", 1) Unfortunately, this parameter cannot be used in default nathelper module in kamailio 4.2.(doc is probably from OpenSIPS) I would like to send OPTION pings to TLS nated and individualy selected user agents. Does exist some possibility to do it with kamailio 4.2? If not by default does exist the possibility to compile and build kamailio with the nathelper module mentioned above (with natping_tcp parameter)? Or does exist some other solution? Thank you in advance Petr

1 0

Kamailio v4.2.1 Released
by Daniel-Constantin Mierla 11 Nov '14

11 Nov '14

Hello, Kamailio SIP Server v4.2.1 stable release is out. This is a maintenance release of the latest stable branch, 4.2, that includes fixes since release of v4.2.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.1. For more details about version 4.2.1 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-2-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

header manipulation in locally generated SIP OPTIONS requests
by Klaus Feichtinger 11 Nov '14

11 Nov '14

Hello, does kamailio offer any possibility for adding / replacing SIP headers in locally built SIP requests? In detail: SIP OPTIONS requests that are built by the DISPATCHER module for probing the configured targets. I have to extend these messages with "Accept" SIP header fields (which are marked with m* in RFC3261 - so they SHOULD be present), as the dispatcher target is requiring this information for answering these requests.... The event_route is principally accepting the append_hf() function and is adding the configured header fields. But the answer to that request (which is including these additional header fields) is confusing the dispatcher module. Dispatcher module is reacting with this ERROR message: DEBUG: dispatcher [dispatch.c:2406]: probing set #1, URI sip:10.10.10.10 DEBUG: dispatcher [dispatch.c:2345]: OPTIONS-Request was finished with code 500 (to , group 1) ERROR: dispatcher [dispatch.c:2358]: Setting the state failed (, group 1) When I comment the textops functions out again and use the original OPTIONS messages, dispatcher is happy and knows the states of the probed targets. Therefore, I´ll ask again: does kamailio offer any way for manipulating locally built SIP requests? kr Klaus

3 4

Releasing v4.2.1
by Daniel-Constantin Mierla 11 Nov '14

11 Nov '14

Hello, I am considering to release first minor version out of branch 4.2 as v4.2.1 next week, by Tuesday or Wednesday. If there is any comment regarding it, write to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

1 1

ALIAS_DB replacing From and Contact fields
by Martin Schiøtz 11 Nov '14

11 Nov '14

Hi I'm using the ALIAS_DB module. username = 101 alias_username = 44444444 (fictive public phone number) 'alias_db_lookup("dbaliases")' is working fine for incoming calls from PSTN. Calling 44444444 R-URI is replaced with 101 SIP uri. But when 101 is calling to a PSTN number how do I replace the From and Contact fields with 44444444? Cheers, Martin

1 0

RTP end points
by Javier Ricke 10 Nov '14

10 Nov '14

i used asterisk, and install kamailio for traffic RTP can be send between end points, and do not have to go through RTP proxy, any work this form?, and client behind to NAT.. regards!

2 2

beginner's problem with sipping
by Petr Wozniak 10 Nov '14

10 Nov '14

Dear sr-users, I am beginner and apologize for bothering you with such a trivial problem. I would like to test sip UDP and TCP keepalive pinging. Kamailio and client are on the same network (192.168.1.44 and 192.168.1.20) that implies not behind NAT. So in kamailio.cfg I have set modparam("nathelper", "ping_nated_only", 0) My simple kamailio.cfg please you find enclosed. >From logfiles on client's side implies that neither UDP nor OPTIONS pings are sending from kamailio. If somebody would be so kind to help me I will be very glad. Thank you in advance Regards Petr

2 3

Counting xavp stack
by Alex Balashov 10 Nov '14

10 Nov '14

It would be great if there were some way to count the depth of a stack of XAVPs, e.g. $xavp(root=>num) = 1; $xavp(root[0]=>num) = 2; $xavp(root[0]=>num) = 3; ... $var(depth) = $(xavp(root=>num){avp.count}); Right now, the only way to do this that I know of is something like: $var(i) = 0; while(defined $xavp(root[$var(i)]=>num)) $var(i) = $var(i) + 1; $var(i) = $var(i) - 1; Is there a secret I don't know about? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 2

Setting up a IMS Server Platform using Kamailio 4.2.x code
by Austin Einter 10 Nov '14

10 Nov '14

Hi All Last few days we have tried to setup Kamailio for IMS server setup. We have not been successful. Now we are planning to use Kamailio 4.2.0 code base (Latest) to setup Kamailio IMS setup. I am looking for a link/document, that gives proper step by step instructions how to setup Kamailio as IMS servers (P-CSCF, I-CSCF and S-CSCF). Can somebody kindly provide the official link or any other working link, that I can follow to setup P-CSCF, I-CSCF ,S-CSCF and HSS. Many thanks Austin

2 1

Fwd: Extension and carrier same ip problem
by Martin Snelgoedkoop 10 Nov '14

10 Nov '14

> Hi, we use Kazoo which loads acl and dial plan out a DB. > We run in to smth that seems freeswitch default behavior so that's why I write here. > > We have a carrier that needs to register a extension to our platform as well. > They use the same ip. And that creates a problem as when I place a call with the device using the extension it gets identified as carrier and the route is then not correct. > > The same happens the other way arround, if a call comes in via the carrier the ip is recognized as an extension ip and then it won't use the carrier route. > > I was wondering if I could use some flags or a way of changing the ip used or doing some magic to make it work. I would prefer not to use a proxy. > > The setup is kamailio>freeswitch > > > Met vriendelijke groet, Martin > > Team snelgoedkoop

3 2

Kamilio verify online servers
by Marino Mileti 10 Nov '14

10 Nov '14

Hi guys, I've an architecture with multiple Kamailio. I would like to implement some check that allow me to verify if a server is online or not so for example i can use msilo to save chat message and relay it when the server come's up. Is it possible to use UAC module to send a REGISTER message from one server to another so i can "dump" all message in very simple way? Is there any example ? Many thanks ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamilio-verify-online-servers-tp132… Sent from the Users mailing list archive at Nabble.com.

2 1

htable locking
by Alex Balashov 10 Nov '14

10 Nov '14

The documentation for the htable module gives no guidance on whether sht_lock()/unlock() should be used in all situations when writing or reading from the htable. Should it be? I was under the impression (not supported by anything said in the documentation) that read and write access to the htable is fundamentally thread-safe, i.e. it is safe to both read and write to the htable even if the potential exists for another worker process to do the same concurrently. Is that not the case? If not, should there not be strong admonitions to use sht_lock/sht_unlock around concurrent operations--and it sounds like nearly all useful applications of the htable are essentially concurrent? Thanks! -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 6

Generate REFER msg
by Agiftel 10 Nov '14

10 Nov '14

Hi all, i need some advice on how to solve a problem. I've a scenario like following: [SIP PBX] <---->CUCM (Call Manager v 10.5 ) User A: belong to CUCM User B: belong to SIP PBX User C: belong to SIP PBX User A calls User B. After some conversation, User A wants to transfer call to User C. CUCM transfer the call with re-INVITE; in this case on SIP PBX 2 channel SIP are used. CUCM does not support transfer, on Trunk, using REFER method. My idea is using Kamailio beetwen SIP-PBX and CUCM and handles call generating REFER msg in some way. Is this idea a good one? Or there is a better way? Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Generate-REFER-msg-tp132317.html Sent from the Users mailing list archive at Nabble.com.

2 1

rr problem: sips instead of sip
by Jon Bonilla (Manwe) 07 Nov '14

07 Nov '14

Hi all I'm experiencing a problem while using a mobile client on a NGCP mr3.5.1 (based on kamailio 4.1.6) The problem is as follows: A custom mobile client B blink for linux The path of a call from B to A would be this one: B --> Kamailio LB --> Kamailio proxy --> Sems --> Kamailio LB --> A Kamailio LB is a stateless proxy. Kamailio proxy is a stateful proxy and registrar and sems acts as B2BUA. Kamailio LB listens tcp,udp and tls to the outside and speaks udp to the inside. Kamailio proxy and sems only listen udp localhost (ports 5062 ans 5080). My problem comes when the ACK from B to A (to the INVITE-200) doesn't leave the Kamailio LB to the proxy. The problem is that the kamailio proxy added itself as sips in the record-route header: U 2014/11/06 19:22:20.447717 127.0.0.1:5062 -> 127.0.0.1:5080 INVITE sips:test1@10.0.1.5:41956;rinstance=145658D8;transport=tcp SIP/2.0' Record-Route: <sips:127.0.0.1:5062;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw-->' Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>' Record-Route: <sip:XX.XX.XX.XX;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>' Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKc36a.b42be7e4a4db27ac699a52fda7fb5fe0.0' Route: <sip:lb@127.0.0.1;lr;received=sip:YY.YY.YY.YY:41956%3Btransport%3Dtls;socket=sip:XX.XX.XX.XX:5061>' Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc36a.a58b9955ce93726b17c0240a555011f2.0' Via: SIP/2.0/UDP 10.10.10.4:47879;received=ZZ.ZZ.ZZ.ZZ;rport=47879;branch=z9hG4bKPj6UmIovzyxDRCgcuz-VoXHEyJHRpA712x' When the ACK arrives to the LB from B, it can't relay it to the Kamailio proxy: NOTICE: <script>: New request on lb - M=ACK R=sip:ngcp-lb@XX.XX.XX.XX;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31 NOTICE: <script>: Relaying request, du='sips:127.0.0.1:5062;lr;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw--', fs='udp:127.0.0.1:5060' - R=sip:127.0.0.1:5080;prxroute=1 WARNING: <core> [forward.c:264]: get_send_socket2(): WARNING: get_send_socket: protocol/port mismatch (forced udp:127.0.0.1:5060, to tls:127.0.0.1:5062) I think the problem might be in the way the client registers? This is the registration info where it uses sips in the contact: contact: sips:test1@10.0.1.5:41956;rinstance=145658D8;transport=tcp If I try the same calling blink, which has the following contact in the registration it works without issues as the proxy doesn't insert sips in the Record-Route header: contact: sip:91807432@10.10.10.4:57160;transport=tls Any thoughts? cheers, Jon

3 2

Kamailio v4.0.7 Released
by Daniel-Constantin Mierla 07 Nov '14

07 Nov '14

Hello, Kamailio SIP Server v4.0.7 stable release is out. This is a maintenance release of the 3rd old stable branch, 4.0, that includes fixes since release of v4.0.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.x. Deployments running previous v4.0.x versions are strongly recommended to be upgraded to v4.0.7, or even better, to 4.1.x or 4.2.x, as the focus from now on will be on maintenance of these release series. For more details about version 4.0.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-0-7-released/ Note: the latest stable branch is 4.2, at this moment with its latest release v4.2.0. See more details about it at: * http://www.kamailio.org/w/2014/10/kamailio-v4-2-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Releasing v4.0.7
by Daniel-Constantin Mierla 07 Nov '14

07 Nov '14

Hello, I am planning to release v4.0.7 based on latest source code in branch 4.0. This branch is going out of official maintenance, the current latest stable branches being 4.2 and 4.1. If you have any remarks regarding it, write back to mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

1 2

cdp.so segment fault
by Tom Secula 06 Nov '14

06 Nov '14

I am just wondering if anyone has seen this type of segfault on 4.2 ? I'm trying to run kamailio as an scscf on centos 6.6 ? Nov 6 14:17:20 scscf kernel: kamailio[6246]: segfault at 0 ip 00f3f425 sp bfa0e9f4 error 6 in cdp.so[f08000+96000]

1 0

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sr-users November 2014