- sr-users - kamailio.org

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!

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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

Re: [SR-Users] RPC uac.reg_info hangs second time
by Dan Christian Bogos 06 Nov '14

06 Nov '14

Hey Daniel, Thanks for your fast reply! I am using the 4.2.0 out of debian packages (not nightly). Will shift on nightly and retest. Not a big deal regarding 150 seconds, maybe it should be a good reason for async reloads (reply with the delay number of seconds when reload will happen). Reloads can be always implemented outside of kamailio but since everybody would need it I assume a more centralized approach would be useful. Anyway, thanks again! DanB On 24.10.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > Hello, > > are you using branch 4.2 or some tgz/package from 4.2.0 release? There > was a fix related to such issue, so 4.2 branch must be used (or debian > nightly builds for 4.2). > > I will have to check the code how flexible that is, but afaik, the limit > on reload interval was thought to avoid sending again registration > requests while retransmissions from previous one can occur. > > Cheers, > Danel

2 7

Kamailio v4.1.7 Released
by Daniel-Constantin Mierla 06 Nov '14

06 Nov '14

Hello, Kamailio SIP Server v4.1.7 stable release is out. This is a maintenance release of the previous stable branch, 4.1, that includes fixes since release of v4.1.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.7. For more details about version 4.1.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-1-7-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

how to check if reply is retransmission?
by Juha Heinanen 06 Nov '14

06 Nov '14

is there a means to check if reply to t_relay()ed request is retransmission? there is t_is_retr_async_reply(), but that does not apply to regular transactions. i would like to avoid double calling of rtpengine_answer(). -- juha

4 5

AT&T and UAC cancel after 200 OK
by Yuriy Gorlichenko 06 Nov '14

06 Nov '14

Hello. I use UAC module for trunks and have trouble when call to mobile endpoint of AT&T provider. When endpoint pickup call ans 200 OK reply come into kamailio, kamailio send CANCEL. so at the endpoint I see than call is dropped and at kamailio client I hear voicemail speech form AT&T endpoint. Provider say that his trace is OK... My sip trace bellow: IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1129 E.......@.l. ...6........q..INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 651 E.......*.Z.6... .........}.SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com >;tag=04e2a294d0728b89107e081a19babab4.43a7 Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", qop="auth" Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 410 E.......@.o. ...6.........j%ACK sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com >;tag=04e2a294d0728b89107e081a19babab4.43a7 Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 ACK Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402 E.......@.k. ...6..........`INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri=" sip:98765432100@phone.myprovider.com", qop=auth, nc=00000001, cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3", algorithm=MD5 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402 E.......@.k. ...6.........._INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri=" sip:98765432100@phone.myprovider.com", qop=auth, nc=00000001, cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3", algorithm=MD5 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529 E..-....*.[.6... .........xiSIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102;rport=1024;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529 E..-....*.[.6... .........xkSIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102;rport=1024;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 998 E.......*.Y16... .........^.SIP/2.0 180 Ringing Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 986 E.......*.Y<6... ...........SIP/2.0 180 Ringing Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=UrBQrZp9vSm3p Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Contact: <sip:98765432100@173.203.60.50:5060;transport=udp> User-Agent: myprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1294 E..*....*.X.6... .........:.SIP/2.0 183 Session Progress Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1282 E.......*.X.6... ........ ..SIP/2.0 183 Session Progress Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=UrBQrZp9vSm3p Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Contact: <sip:98765432100@173.203.60.50:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415020637 1415020638 IN IP4 173.203.60.50 s=FreeSWITCH c=IN IP4 173.203.60.50 t=0 0 m=audio 25430 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W.6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 412 E.......@.o. ...6.........j.CANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6...........ACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.0ec383b6dc8fbb1861dfe644f92ee9ca.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK010bb317;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W~6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........jACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.6a1e71bf5fcd88f137733279fa31770c.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK1f95579c;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 246 E.......*.\.6... ...........OPTIONS sip:68.34.22.11:1024 SIP/2.0 Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0 From: sip:keepalive@myprovider.com;tag=29ebf5d To: sip:68.34.22.11:1024 Call-ID: 6360a4bb-53bebf7-5686ca6(a)10.202.129.254 CSeq: 1 OPTIONS Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 330 E..f....@.p. ...6........R.<SIP/2.0 405 Method not allowed Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0;rport=5060;received=34.43.4.23 From: sip:keepalive@myprovider.com;tag=29ebf5d To: sip:68.34.22.11:1024;tag=c6894543322e2a4942d921e4298ce904.1ec7 Call-ID: 6360a4bb-53bebf7-5686ca6(a)10.202.129.254 CSeq: 1 OPTIONS Server: MS Lync Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W|6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........eACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.47488c128d3518b837410e363acc963e.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK5e2d222b;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W{6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........;ACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.00885667802b30315ce7f8b5db31e2bd.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK6081ac89;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200

1 0

how to use mqueue.
by 张顺通 06 Nov '14

06 Nov '14

whle (mq_fetch("QUEUE")) { if (yes) { do something } else { mq_add("QUEUE"); add back to queue. } } can mq_add use here? I tried it, it do not work. and How to determine the end?

3 6

Releasing v4.1.7
by Daniel-Constantin Mierla 06 Nov '14

06 Nov '14

Hello, I plan to release a new version (4.1.7) of the previous stable branch 4.1. Most probably on Thursday this week. If anyone has remarks/issues about it, write back to mailing lists to discuss. 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

benchmark module
by bob sacamano 05 Nov '14

05 Nov '14

hello, i was wondering where does this module write the logs? Nothing shows up in my syslog. If i replace the benchmark parts in the route with xlog for example xlog( "L_INFO", "$Ts " ); this shows up fine in syslog heres what u have in my config file: modparam("benchmark", "enable", 1) modparam("benchmark", "granularity", 100) modparam("benchmark", "loglevel", 3) if(is_method("REGISTER")) { bm_start_timer("test"); save("location"); bm_log_timer("test"); exit; } I have been searching but cant find any info on it

2 5

Kamailio with media proxy and two network interfaces
by Julien 05 Nov '14

05 Nov '14

Hello, I have a problem with Kamailio and a setup with two network interfaces (eth0 and eth1) Devices on eth0 CANNOT see devices on eth1 and vice versa, there is no route between both interfaces. Only Kamailio and the media proxy can access both. +--------------------------+ | kamailio + rtpproxy | eth0 | + openvpn | eth1 +---------+ +-----------+ | | | | | +--------------------------+ | +-+----+ +----+-+ | A | | B | | | | | +------+ +------+ A user connected at eth0 shall be able to get the presence status of users on eth1 and users shall be able to call each other. What I did so far: I added listen statements for both interfaces to the Kamailio config file: -listen=udp:eth0_IP:5060 -listen=udp:eth1_IP:5060 What's not working: When I try to make a call from user in A domain to another user in B domain, the call established correctly, but no stream running. On B device I see an incoming call from 'sip:user@A_Domain'. B user doesn't have route for A Domain, so that doesn't work. Is what I want technically possible with Kamailio. If yes, what do I have to change in my config? Do you have any suggestions? Thanks in advance. -- Julien Barreau

3 3

how you make Kamailio with mariadb
by ANTHONY HESNAUX 05 Nov '14

05 Nov '14

hello Daniel I want to install Kamailio with mariadb on CENTOS 7 I already install Kamailio on CENTOS 6.5 without error but on CENTOS 7 with mariadb, What modules do you include for make cfg: make include_modules="db_mysql" cfg thank you best regard Anthony HESNAUX

3 8

UAC error at reply : t_should_relay_response:
by Yuriy Gorlichenko 05 Nov '14

05 Nov '14

ERROR: t_should_relay_response: status rewrite by UAS: stored: 491, received: 200 Hello. I use UAC module for trunks form 4.2 master branch. I have an error when endpoit at the trunk picked up. OK reply never replied to caller because I see next error: ERROR: t_should_relay_response: status rewrite by UAS: stored: 491, received: 200 So as I see before reply 200 sended to kamailio (from callee) I see packet with 491 SIP code. It is strange because at the another server (same configuration) OK successfully sended to caller and session is Ok...

1 0

rare crash , race condition between 183 and 408 ?
by Dragos Oancea 05 Nov '14

05 Nov '14

Hi all & hi Daniel , We had a crash with version 4.0.5 today . I looks like it happens due to a race condition between a provisional response (183) and the generation of a 408 . We send the INVITE, get a `100 Trying` from upstream , then we get a `183 Session Progress` from upstream , then after exactly 1 minute , we get another `183 Session Progress` from upstream. At this moment kamailio crashed. Some tm params we have : modparam("tm", "fr_timer", 5000) modparam("tm", "fr_inv_timer", 60000) <- our final reply timer is 1 minute . modparam("tm", "restart_fr_on_each_reply", 1) GDB output and logs here: kamailio 4.0.5 crash - in fake_req() - race cond - Pastebin.com kamailio 4.0.5 crash - in fake_req() - race cond - Paste... (gdb) bt #0 0x00007f4d93129a32 in memcpy () from /lib64/libc.so.6 View on pastebin.com Preview by Yahoo In fake_req() from t_reply.c , the pointer shmem_msg->new_uri.s which is passed to memcpy becomes invalid. memcpy( faked_req->new_uri.s, shmem_msg->new_uri.s,faked_req->new_uri.len); (gdb) p shmem_msg->new_uri $3 = {s = 0x0, len = 0} If this bug is not supposed to be fixed in 4.2.0 , we can help with debugging . Regards, Dragos

2 1

AliasDB module & Forking
by Marino Mileti 05 Nov '14

05 Nov '14

Hi guys, i've implemented a parallel forking using aliasdb module. Everything is working good if the first client (first row in dbaliases table) in online. When I try to make a call to a specific number three clients rings together. If the first client isn't online no call is forwarded to other clients :( I know that with alias_db_lookup() the first record from DB is the new R-URI and the others are added like branches. But if the R-URI in unreachable how can try to call the other branches? This is location script.. if(alias_db_lookup("dbaliases")){ $sht(a=>$ci) = $sel(contact.uri); } $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: xlog("L_INFO","No contact found\n"); exit; case -3: send_reply("404", "Erorr"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } Any suggest? My script still log "No contanct found" until timeout :( Many thanks ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/AliasDB-module-Forking-tp132128.html Sent from the Users mailing list archive at Nabble.com.

2 3

DTMF Tones question
by Anthony Martin 05 Nov '14

05 Nov '14

HI, Please, excuse me for my bad English Just a question, kamailio does support DTMF Tones? i have a big while searching for documentation about that but without good results. I want to make a server side call transfer method whit DTMF Tones... Thanks so much for your good work Greetings from Venezuela

3 2

DBURL
by shmick＠riseup.net 05 Nov '14

05 Nov '14

[i have no idea about email subject nor anything else for this except prior i configured rtpproxy to work with kamailio on a unix socket and updated tls.cfg to use my cert/key] running kamailio from deb http://deb.kamailio.org/kamailio jessie main $ sudo service kamailio restart [FAIL] Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed! [....] 0(15228) INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... 0(15228) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 353, column 30-34: syntax error 0(15228) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 353, column 30-34: Invalid arguments 0(15228) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 353, column 35: ERROR: bad config file (3 errors) 0(15228) WARNING: <core> [ppcfg.c:219]: pp_ifdef_level_check(): different numbe[FAILpreprocessor directives: N(#!IF[N]DEF) - N(#!ENDIF) = 1 ... failed! failed! /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", *DBURL*) <--------not happy with DBURL here modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif

2 3

dnssec module
by shmick＠riseup.net 05 Nov '14

05 Nov '14

running kamailio from deb http://deb.kamailio.org/kamailio jessie main when i add the dnssec module to /etc/kamailio/kamailio.cfg kamailiop fails to start and gives: [FAIL] Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed! [....] 0(8839) ERROR: <core> [sr_module.c:597]: load_module(): could not open module </usr/lib/x86_64-linux-gnu/kamailio/modules/dnssec.so>: /usr/lib/x86_64-linux-gnu/libval-threads.so.14: undefined symbol: SSL_get_SSL_CTX 0(8839) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /etc/kamailio/kam[FAIL.cfg, line 239, column 12-22: failed to load module ERROR: bad config file (1 errors) ... failed! failed! $ sudo ls -la /usr/lib/x86_64-linux-gnu/kamailio/modules/dnssec.so -rw-r--r-- 1 root root 22912 Oct 17 09:04 /usr/lib/x86_64-linux-gnu/kamailio/modules/dnssec.so any help appreciated thanks

2 4

REGISTRAR location replication (pua_reginfo)
by lejarreta.e＠sarenet.es 04 Nov '14

04 Nov '14

Good morning. We've got a problem with location memory and database replication between "n" kamailios as REGISTRAR servers. 1.- WHAT WE WANNA DO: - To have "n" Kamailios with their DB and memory "user location" replicated. - Each Kamailio owns its own DB.(It could be also with a common DB). - Each kamailio has a PUA_REGINFO, PRESENCE, and REGISTRAR roles. - USRLOC Module with db_mode = 2 (Write-Back scheme). 2.- WHAT WE DON'T WANNA DO: - USRLOC Module with db_mode != 2 (Write-Back scheme). 3.- BACKGROUND: - RFC3265 - RFC3680 - RFC3903 - I attach a small architecture graph. 4.- PROBLEMS: 4.1 Deadlock: Each REGISTER involves a NOTIFY to the other side. Each NOTIFY involves another NOTIFY to the other side. There should be a mechanism to avoid PUBLISH reginfo_handle_notify() function calls but to PUBLISH save() calls on pua_reginfo module. Any idea would be appreciated (also to workaround our problem). 4.2 PUA_REGINFO module notify.c Just FYI. We think It doesn't comply to RFC3680. It should observe "unkown-param" object to aid "Contact" header params such as "Path", "Received", "User_agent" and so on. We've changed code but for a fast startup we've also avoided "unkown-param" and now we're able to see replicated "Path", "Received ... params on each DB. Thanks and best regards. Eduardo Lejarreta.

2 1

how to configure multi-proxy mode for kamailo 4.2.x
by andrew 04 Nov '14

04 Nov '14

hi By default, kamailio.cfg is used for one proxy. How to set up kamailio to support multi-proxy, so that serveral kamaiio servers work together, and each one is responsible for sip clients in some specific area? B.R. Andrew

2 1

"Help with dialog $dlg_var(cseq_diff)"
by Yuriy Gorlichenko 03 Nov '14

03 Nov '14

Hello. I need to increment CSeq value for INVITE with Auth params when use UAC_AUTH for outgoing calls to provider. Kamailio 4.2 may increment this using dialog module http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for… Now I experements with this and var $dlg_var(cseq_diff) and at transaction way int have NULL value. I can not understand why My config is: # ----- dialog params modparam("dialog", "db_url",DBURL) modparam("dialog", "db_mode", 1) modparam("dialog","table_name","dialog") modparam("dialog", "dlg_flag", 4) modparam("dialog", "initial_cbs_inscript", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "default_timeout", 60) modparam("dialog", "track_cseq_updates", 1) route config is at request route if(is_method("INVITE") && !has_totag()){ $dlg_ctx(timeout_route) = "DIALOG_END"; $dlg_ctx(timeout_bye) = 1; dlg_manage(); xlog("L_INFO","Dialog manage is {$ct}\n"); } } t_relay(); next I handle failure reply because 407 reply is recieved failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_check_status("401|407")){ xlog("L_INFO", "Reply from provider on failure: $tU"); xlog("L_ERR","401/407 - Unauthorized. ($ci .) ($rm) from ($fu) (IP:$si:$sp) to ($Ri:$Rp). Must be authorized with digest Auth."); avp_print(); xlog("L_INFO", "CSeq diff: $dlg_var(cseq_diff)"); uac_auth(); xlog("L_INFO", "UAC_AUTH(): $tU"); append_branch(); t_relay(); } if (t_is_canceled()) { exit; } } As you may see i logging $dlg_var(cseq_diff) value and now it NULL. So I can not understand why? What wrong I do? Thanks for advice.

2 3

Problems using htable iterator
by Glenn Larsen 03 Nov '14

03 Nov '14

Hi, I have a hard time using the htable iterator... I have this simple setup: modparam("htable", "htable", "h1=>size=4") ... sht_iterator_start("i1", "h1"); while (sht_iterator_next("i1")) { xlog("L_ERR", " SS: $shtitkey(i1) and $shtitval(i1)\n"); } sht_iterator_end("i1"); ... Now to put some simple data, I use: # sudo kamcmd htable.sets h1 test 123 # sudo kamcmd htable.dump h1 { entry: 8 size: 1 slot: { item: { name: test value: 123 } } } When the route is "executed", I get this in my log file: Nov 3 15:16:47 beast /sbin/kamailio[24865]: ERROR: SS: <null> and <null> The $shtitkey(i1) and $shtitval(i1) is always null. Is it a bug? Or did I manage to do something wrong? -- Best regards, Glenn O. Larsen -- Best regards, *Glenn O. Larsen* *CTO* *SimCom AS* Professor Birkelands vei 24c N-1081 Oslo Direct: +47 21 09 34 12 Mobile: +47 997 00 000 Online profile: http://simcom.no/glenn.larsen *We simplify your communication ...*

2 1

rtpengine does not send RTP packets out
by Juan Perez 03 Nov '14

03 Nov '14

Hi, I have kamilio-4.2 and rtpengine running on the same machine.I have this scenario: softphone --> Kamailio with Rtpengine --> AsteriskThe softphone initiates the call, it is sent to the Asterisk. I can see the SDPs being re-written with the new IP/Ports provided by rtpengine: Invite from Kamailio to Asterisk200 Ok from Kamailio to Softphone However, I take a signaling/media capture on the server where the kamailio/rtpengine are running and see the RTP coming from both endpoints (softphone and asterisk) to the correct ports but there is no packets coming out from the proxy to either direction. I see these 2 lines on the rtpengine log and make me think that something prevents the rtpengine to stream out to the 2 endpoints: Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Write error on RTP socket Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Final packet stats: this is the log from the rtpengine: Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: Got valid command from 127.0.0.1:46686: offer - { "sdp": "v=0#015#012o=- 314474204173996793 6663595733338147866 IN IP4 99.153.161.148#015#012s=-#015#012c=IN IP4 99.153.161.148#015#012t=0 0#015#012m=audio 16386 RTP/AVP 114 0 8 101#015#012a=rtpmap:114 ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8 pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012", "direction": [ "pub", "pub" ], "replace": [ "origin", "session-connection" ], "call-id": "0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [ "IP4", " ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: ... 99.153.161.148" ], "from-tag": "73rH43p60NeKm", "command": "offer" } Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Creating new call Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Opened ports 35022..35023 for media relay Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Opened ports 35038..35039 for media relay Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Returning to SIP proxy: d3:sdp529:v=0#015#012o=- 314474204173996793 6663595733338147866 IN IP4 54.173.60.13#015#012s=-#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35022 RTP/AVP 114 0 8 101#015#012a=rtpmap:114 ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8 pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:35023#015#012a=ice-ufrag:oPm3I7OE#015#012a=ice-pwd:WjZnh7dnhfdkfoXDoqDbnCLLr7N7#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35022 typ host#015#012a=candidate: ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] ... NZ1IXXQHYxOora9j 2 UDP 2130706430 54.173.60.13 35023 typ host#015#0126:result2:oke Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: Got valid command from 127.0.0.1:46686: answer - { "sdp": "v=0#015#012o=root 439969655 439969655 IN IP4 205.10.248.62#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 205.10.248.62#015#012t=0 0#015#012m=audio 11142 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=silenceSupp:off - - - -#015#012a=ptime:20#015#012a=sendrecv#015#012", "direction": [ "pub", "pub" ], "replace": [ "origin", "session-connection" ], "call-id": "0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [ "IP4", "205.10.248.62" ], "fr ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: ... om-tag": "73rH43p60NeKm", "to-tag": "as57e7b8e3", "command": "answer" } Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Returning to SIP proxy: d3:sdp500:v=0#015#012o=root 439969655 439969655 IN IP4 54.173.60.13#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35038 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=silenceSupp:off - - - -#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:35039#015#012a=ice-ufrag:cBL9SIw9#015#012a=ice-pwd:QHw7VHp83nMsLpD8ZoSqtDxGchN4#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35038 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP 213070 ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] ... 6430 54.173.60.13 35039 typ host#015#0126:result2:oke Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Write error on RTP socket Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Final packet stats: Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] --- Tag 'as57e7b8e3', created 0:00 ago, in dialogue with '73rH43p60NeKm' Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35022 <> 205.10.248.62:11142, 0 p, 0 b, 0 e Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35023 <> 205.10.248.62:11143 (RTCP), 0 p, 0 b, 0 e Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] --- Tag '73rH43p60NeKm', created 0:00 ago, in dialogue with 'as57e7b8e3' Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35038 <> 99.153.161.148:16386, 0 p, 0 b, 1 e Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35039 <> 99.153.161.148:16387 (RTCP), 0 p, 0 b, 0 e Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Got valid command from 127.0.0.1:46686: delete - { "call-id": "0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [ "IP4", "205.10.248.62" ], "from-tag": "as57e7b8e3", "command": "delete" } Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] [0866b358-dc9c-1232-1399-3767db69b8dd] Call-ID to delete not found Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Returning to SIP proxy: d7:warning38:Call-ID not found or tags didn't match6:result2:oke this is the log from kamailio (showing only the lines that include the word rtpengine): Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: INFO: <script>: mylog: TO_FS: Starting rtpengine session. Invite Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:request_route=[TO_FS] c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=441 a=25 n=rtpengine_offer Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine_funcs.c:148]: check_content_type(): type <application/sdp> found valid Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]: rtpp_function_call(): proxy reply: d3:sdp529:v=0#015#012o=- 314474204173996793 6663595733338147866 IN IP4 54.173.60.13#015#012s=-#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35022 RTP/AVP 114 0 8 101#015#012a=rtpmap:114 ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8 pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:35023#015#012a=ice-ufrag:oPm3I7OE#015#012a=ice-pwd:WjZnh7dnhfdkfoXDoqDbnCLLr7N7#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35022 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP 2130706430 54.173.60.13 35023 typ host#015#0126:result2:oke Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: INFO: <script>: mylog: Starting rtpengine session. Answer Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:onreply_route=[TO_FS] c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=476 a=25 n=rtpengine_answer Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine_funcs.c:148]: check_content_type(): type <application/sdp> found valid Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]: rtpp_function_call(): proxy reply: d3:sdp500:v=0#015#012o=root 439969655 439969655 IN IP4 54.173.60.13#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35038 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=silenceSupp:off - - - -#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:35039#015#012a=ice-ufrag:cBL9SIw9#015#012a=ice-pwd:QHw7VHp83nMsLpD8ZoSqtDxGchN4#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35038 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP 2130706430 54.173.60.13 35039 typ host#015#0126:result2:oke Nov 1 18:59:36 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:request_route=[LOOSE_ROUTE] c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=488 a=24 n=rtpengine_delete Nov 1 18:59:36 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]: rtpp_function_call(): proxy reply: d7:warning38:Call-ID not found or tags didn't match6:result2:oke

2 4

signaling for NON SIP environment
by fb111＠web.de 03 Nov '14

03 Nov '14

2 1

question
by Javier Ricke 03 Nov '14

03 Nov '14

i work with asterisk, but i have problem at the moment to send audio rtp, because ip is private, kamailio will could solution this problem? regards

2 1

mysql db error
by shmick＠riseup.net 03 Nov '14

03 Nov '14

2 1

Address already in use error when starting Kamailio service
by Mahmoud Ramadan Ali 03 Nov '14

03 Nov '14

Hiiiiii everyone, I'm receiving this error when starting Kamailio saying that it can not resolve my ip addresses used by Kamailio and another "127.0.0.1:5060 : Address already in use " error message...so how can i fix this please ? Nov 2 15:39:26 Kamailio /usr/local/sbin/kamailio[7692]: CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 35 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.100.1 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.50.1 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.100.1 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.50.1 Nov 2 15:39:44 Kamailio /usr/local/sbin/kamailio[7726]: ERROR: <core> [tcp_main.c:2876]: tcp_init(): bind(b, 0x7faebd135ca4, 16) on 127.0.0.1:5060 : Address already in use

3 3

Kamailio behind NAT - Wrong VIA Header
by Marko Seidenglanz 03 Nov '14

03 Nov '14

Hello, We are using Kamailio 4.2 and RTPEngine 3.3 hosted on a VM behind NAT in Google Cloud Engine. Asterisk 11 + Kamailio + RTPEngine are on the same machine and Kamailio is used as WebRTC Gateway. INVITES, that are rewritten by Kamailio look like the following: INVITE sip:eTden7KoNSdMVCUNYIwmz@wh2.24dial.com SIP/2.0 Record-Route: <sip:0.0.0.0:5064;lr=on;nat=yes> Via: SIP/2.0/UDP 0.0.0.0:5064 ;branch=z9hG4bK86f5.88fb7f8810c91758b5f3962420d3a4ef.0 Via: SIP/2.0/UDP 10.240.1.95:5060 ;received=127.0.0.1;branch=z9hG4bK05952ee6;rport=5060 Max-Forwards: 69 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as7900b63f To: <sip:eTden7KoNSdMVCUNYIwmz@wh2.24dial.com> Contact: <sip:anonymous@10.240.1.95:5060;alias=127.0.0.1~5060~1> Call-ID: 0662d336694c60ed60ed5310571a1aee@10.240.1.95:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.1 Date: Mon, 03 Nov 2014 09:01:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 676 P-hint: outbound As you can see, Kamailio uses 0.0.0.0 for Record-Route as well as for VIA Header. It uses the address configured in kamilio.cfg (listen=udp:0.0.0.0:5064) I know, that we could use the advertise parameter to tell kamailio the external IP. The problem is, that IPs in GCE are dynamically assigned to the VMs. Thus static configuration of external IP is not sufficient for our setup. Is there a way, to let Kamailio gather it's external IP dynamically. (E.g. using a STUN server or something like that?) Another possibility would be to tell Kamailio it's external IP via a start parameter. Is that possible? Any help would be grateful. Kind regards, Marko Seidenglanz

2 1

using time stamp
by bob sacamano 03 Nov '14

03 Nov '14

hi, I am trying to see how long kamailio takes to process messages I saw the commands: set_time_stamp diff_time_stamp in older versions of kamailio but they dont seem to be available. I have been searching for info but have been unable to find anything. Is the command no longer available or is there some other way to use? thanks

2 3

FIFO Error When Running Kamctl dispatcher reload
by Mahmoud Ramadan Ali 02 Nov '14

02 Nov '14

Hiiiii Everyone I have encountered that error whenever i try to use the #Kamctl dispatcher reload even thought that i have the "modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")" and loadmodule "mi_fifo.so" configured...any ideas how to solve this issue ?! root@Kamailio:/usr/local/etc/kamailio# kamctl dispatcher reload ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module.

2 1

Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error
by vivi 01 Nov '14

01 Nov '14

Hi all.. I have solve the problem . Cheers, vivi From: vivi [mailto:vivi.hilton@gmail.com] Sent: Saturday, June 06, 2009 11:29 PM To: 'users(a)lists.kamailio.org' Subject: /usr/local/sbin/kamdbctl create error Hi all I have specify the wanted db type (DBENGINE=MYSQL) in the /usr/local/etc/kamailio/kamctlrc then I using "/usr/local/sbin/kamdbctl create" to create MySQL database, but I got this error: ERROR: database engine not specified, please setup one in the config script root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim /usr/local/etc/kamailio/kamctlrc root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR: Creating core database and grant privileges failed! Cheers, vivi

2 1

Trouble with LCR removing $ru after upgrading to 4.1.6
by Will Ferrer 01 Nov '14

01 Nov '14

Hi We are having an odd issue with our LCR. This started after we upgraded from kamailio 4.1.0 to 4.1.6. Here is the situation: We populate our gw_uri_avp (stored in: $avp(i:709)) as follows: 16|1||||[gateway ip address]||5060||1|1. The $ru before running next gateway shows: sip:[destination number]@[our domain that the call came into]. This is how it should look before lcr occurs. Then we run next_gw(). We get these message in the logs during the running of next_gw DEBUG: lcr [lcr_mod.c:2156]: generate_uris(): r_uri <*1*[destination number]@[gateway ip address]:5060*1*>, dst_uri <> DEBUG: lcr [lcr_mod.c:2437]: next_gw(): added ruri_user_avp <[destination number]> DEBUG: lcr [lcr_mod.c:2473]: next_gw(): added flags_avp <1> DEBUG: lcr [lcr_mod.c:2480]: next_gw(): added tag_avp <> DEBUG: lcr [lcr_mod.c:2488]: next_gw(): added defunct_gw_avp <1253613379> The main thing to notice there is that there is now a 1 at the start and end of the r_uri shown in generate_uris. I am fairly certain those are new, and maybe related to what happens next. Immediately following the next_gw function running the $ru becomes: <null>, so of course the call doesn't go through. I have looked at the code for the lcr module and it looks like it wasn't changed in version 4.1.6. I found alot of other notes in the change log that had to do with the way uris are handled so it is possible one of those changes caused the problem. I am stumped currently. Any help any one can offer me would be very appreciated. All the best. Will Ferrer Swithsoft Inc

2 4

Installing rtpengine on CentOS 6.5
by Juan Perez 31 Oct '14

31 Oct '14

Using Kamailio 4.2, CentOS 6.5, as per the documentation here: https://github.com/sipwise/rtpengine "Running make will compile the binary, which will be called rtpengine." When trying to install rtpengine I got the following error when building "daemon" [root@ip-10-0-2-68 daemon]# make Usage: awk [POSIX or GNU style options] -f progfile [--] file ... Usage: awk [POSIX or GNU style options] [--] 'program' file ... POSIX options: GNU long options: -f progfile --file=progfile -F fs --field-separator=fs -v var=val --assign=var=val -m[fr] val -O --optimize -W compat --compat -W copyleft --copyleft -W copyright --copyright -W dump-variables[=file] --dump-variables[=file] -W exec=file --exec=file -W gen-po --gen-po -W help --help -W lint[=fatal] --lint[=fatal] -W lint-old --lint-old -W non-decimal-data --non-decimal-data -W profile[=file] --profile[=file] -W posix --posix -W re-interval --re-interval -W source=program-text --source=program-text -W traditional --traditional -W usage --usage -W use-lc-numeric --use-lc-numeric -W version --version To report bugs, see node `Bugs' in `gawk.info', which is section `Reporting Problems and Bugs' in the printed version. gawk is a pattern scanning and processing language. By default it reads standard input and writes standard output. Examples: gawk '{ sum += $1 }; END { print sum }' file gawk -F: '{ print $1 }' /etc/passwd Makefile:88: .depend: No such file or directory make: *** No rule to make target `aux.c', needed by `.depend'. Stop. [root@ip-10-0-2-68 daemon]# I was able to build the iptables-extension, the kernel failed because I dont have the dependencies but the daemon is required. I think that I have all the dependencies: [root@ip-10-0-2-68 daemon]# rpm -qa | egrep "pcre|pkgconfig|xml|glib|zlib|openssl|curl" openssl-1.0.1e-30.el6_6.2.x86_64 glibc-common-2.12-1.132.el6_5.4.x86_64 zlib-1.2.3-29.el6.x86_64 curl-7.19.7-37.el6_5.3.x86_64 glibc-devel-2.12-1.132.el6_5.4.x86_64 libcurl-devel-7.19.7-37.el6_5.3.x86_64 openssl-devel-1.0.1e-30.el6_6.2.x86_64 glibc-2.12-1.132.el6_5.4.x86_64 pcre-7.8-6.el6.x86_64 libxml2-2.7.6-14.el6_5.2.x86_64 libcurl-7.19.7-37.el6_5.3.x86_64 python-pycurl-7.19.0-8.el6.x86_64 pkgconfig-0.23-9.1.el6.x86_64 glib2-2.26.1-7.el6_5.x86_64 dbus-glib-0.86-6.el6.x86_64 glibc-headers-2.12-1.132.el6_5.4.x86_64 zlib-devel-1.2.3-29.el6.x86_64 libxml2-devel-2.7.6-14.el6_5.2.x86_64 pcre-devel-7.8-6.el6.x86_64 [root@ip-10-0-2-68 daemon]# Am I missing something here?, thank you jp

2 4

pipelimit rate question
by Alex Balashov 31 Oct '14

31 Oct '14

Hi, There does not seem to be a way to update the limits of pipes dynamically. If I don't use DB backing, there is no MI/RPC command to flush existing pipes and cause them to be recreated with reconstituted limits. If I use DB backing and add pipes to the pl_pipes table, an SQL update to the limit column does not influence existing pipes that have been loaded from the DB on startup. Am I missing something, or does the existing feature set of the module simply not allow this? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 4

async workers causing unexplained high load average
by Alex Balashov 31 Oct '14

31 Oct '14

Another thing I have found is that having a certain amount of async workers running, even if they are not doing anything, appears to cause unexplained CPU load, even if the Kamailio instance is completely idle and not processing any calls. Here is the baseline load with no async workers: [root@centosity6 ~]# uptime 14:06:28 up 2 min, 2 users, load average: 0.01, 0.00, 0.00 If I start Kamailio with 1 async worker... modparam("async", "workers", 1) async_workers=1 14:07:59 up 3 min, 2 users, load average: 0.00, 0.00, 0.00 Still nothing. Now I try 2: 14:08:54 up 4 min, 2 users, load average: 0.45, 0.11, 0.04 14:08:59 up 4 min, 2 users, load average: 0.49, 0.12, 0.04 14:09:04 up 4 min, 2 users, load average: 0.53, 0.14, 0.04 14:09:09 up 4 min, 2 users, load average: 0.57, 0.15, 0.05 14:09:14 up 5 min, 2 users, load average: 0.60, 0.17, 0.05 14:09:19 up 5 min, 2 users, load average: 0.63, 0.18, 0.06 14:09:31 up 5 min, 2 users, load average: 0.69, 0.21, 0.07 14:09:37 up 5 min, 2 users, load average: 0.71, 0.22, 0.07 14:10:35 up 6 min, 2 users, load average: 0.89, 0.36, 0.13 14:11:07 up 6 min, 2 users, load average: 0.93, 0.42, 0.16 ... I don't have time right now to see if it will rise to infinity, but it just keeps going up. There are no SIP messages being received at all, and there is absolutely nothing going on. The ascent of the load average is much higher with >= 4 threads. I have 8 "CPUs" available from /proc/cpuinfo (quad-core processor with HyperThreading). When I attach strace to the PID of the Async Worker, there's nothing going on except the normal recvfrom() blocking wait: [root@centosity6 kamailio]# kamctl ps | grep Async which: no greadlink in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/) Process:: ID=11 PID=1559 Type=Async Task Worker Process:: ID=12 PID=1560 Type=Async Task Worker [root@centosity6 kamailio]# strace -p 1559 Process 1559 attached - interrupt to quit recvfrom(9, [root@centosity6 ~]# strace -p 1560 Process 1560 attached - interrupt to quit recvfrom(9, Nor do the other Kamailio threads seem to be doing anything unusual. 'ps aux' and 'top' show 0.0% CPU load on the Kamailio processes themselves, and 'iowait' shows the system as 98-99% idle. I suppose it's possible that this is a peculiarity of the testing environment: I am running this on CentOS 6.5 inside VirtualBox, which I think is KVM-based. I have not had a chance to duplicate this experiment in a different kind of execution environment yet. It's possible that the load average is simply being improperly indicated because of the VM environment. -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 6

append_hf on kamctl fifo dlg_bridge destroys ACK
by Kristian F. Høgh 31 Oct '14

31 Oct '14

Hi, Tested on 4.0.6 (+contact header REFER patch) and trunk. When appending header on originating INVITE, the ACK from kamailio gets messed up. I generate call, using myself as proxy kamctl fifo dlg_bridge sip:201@[domain] sip:202@[domain] sip:[kamailio-ip]:5060 In route[tm:local-request], i use append_hf("P-Asserted-Identity:...") If I remove append_hf, the ACK looks OK. Regards, Kristian Høgh Trace on local interface: INVITE sip:201@[domain] SIP/2.0 Via: SIP/2.0/UDP [kamailio-ip];branch=z9hG4bK9d54.ef433c83000000000000000000000000.0 To: <sip:201@[domain]> From: <sip:controller@kamailio.org>;tag=760f7752be879bf1a1346b149d21b85f-8c89 CSeq: 10 INVITE Call-ID: 0b4f5d725846cc40-12109@[kamailio-ip] Max-Forwards: 70 Content-Length: 147 User-Agent: kamailio Contact: <sip:controller@kamailio.org:5060> Content-Type: application/sdp P-Asserted-Identity: <sip:201@[domain]> <snip SDP> SIP/2.0 200 OK Via: SIP/2.0/UDP [kamailio-ip];rport=5060;branch=z9hG4bK9d54.ef433c83000000000000000000000000.0 Record-Route: <sip:[kamailio-ip];lr;ftag=760f7752be879bf1a1346b149d21b85f-8c89;vsf=AAAAAFFfXzREVlRVXUR1WU8dAxFCHAFHQgYCbC5kaw--;did=abe.bd12> Contact: <sip:201@[peer-ip]:43017;rinstance=50924dd7a7927b2f;transport=UDP> To: <sip:201@[domain]>;tag=594df15d From: <sip:controller@kamailio.org>;tag=760f7752be879bf1a1346b149d21b85f-8c89 Call-ID: 0b4f5d725846cc40-12109@[kamailio-ip] CSeq: 10 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.0.19649 r19608 Allow-Events: presence, kpml Content-Length: 157 <snip SDP> ACK sip:201@[peer-ip]:43017;rinstance=50924dd7a7927b2f;transport=UDP SIP/2.0 Via: SIP/2.0/UDP [kamailio-ip];branch=z9hG4bK9d54.ef433c83000000000000000000000000.0 \0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00P\0xeeu\0x00\0x00\0x00\0x00\0x00$\0x12v\0x00\0x00\0x00\0x00\0x00\0xbb\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0xf0\0xf0\0xf0\0xf0\0x00\0x00\0x00\0x00\0x16\0x0075\0x13\0x01\0x00\0x00X\0xfc\0xc2Vk\0x7f\0x00\0x00\0x00\0xfd\0xc2Vk\0x7f\0x00\0x00\0x0a\0x00\0x00\0x00f-8c\0x0b\0xfd\0xc2Vomain\0x00<domain>\0x009@178X\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00To: <sip:201@[domain]>;tag=594df15d k\0x7f\0x00\0x00\0x17\0x00\0x00\0x00 ACK Route: <sip:[kamailio-ip];lr;ftag=760f7752be879bf1a1346b149d21b85f-8c89;vsf=AAAAAFFfXzREVlRVXUR1WU8dAxFCHAFHQgYCbC5kaw--;did=abe.bd12> User-Agent: kamailio Content-Length: 0 REFER ...

2 4

Susceptibility to POODLE Vulnerability?
by Rainer Piper 31 Oct '14

31 Oct '14

Hi all, is it possible to add in http://kamailio.org/docs/modules/4.2.x/modules/tls.html under the line > 9.1. |tls_method| (string) ... ... If rfc3261 conformance is desired, TLSv1 must be used. For compatibility with older clients SSLv23 is a good option. *Example 1.3. Set |tls_method| parameter* ... modparam("tls", "tls_method", "TLSv1") ... < !!! *a warning **that the use of SSLv3 **susceptibility to POODLE Vulnerability* !!! -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rainer@sip.soho-piper.de:5072 (pjsip-test) XMPP: rainer(a)xmpp.soho-piper.de

4 6

cSeq increasing
by Yuriy Gorlichenko 30 Oct '14

30 Oct '14

Does it possible increase cSeq manually (for example remove and then append headers?) for UAC module when send INVITE messages with Auth, or kamailio have pseudovar for this header?

4 8

UAC INVITE to Provider
by Yuriy Gorlichenko 30 Oct '14

30 Oct '14

Hello. I use kamailio for calling to porvider. My providr seccefuully registered from UAC module, but when I try to call through it? it back 401 Unauthorised. I send second try with Digest Auth header at INVITE and it receive me 401 too... I register this provider from asterisk and call succesfully Ok. So i get dump from asterisk This is successfull INVITE: INVITE sip:89126975590@sip.provider.com SIP/2.0 Via: SIP/2.0/UDP 17.4.28.7:50600;branch=z9hG4bK5f118681;rport Max-Forwards: 70 From: <sip:gw2@17.4.28.7:50600>;tag=as33192a38 To: <sip:89126975590@sip.provider.com> Contact: <sip:gw2@17.4.28.7:50600> Call-ID: 021088c360a8dbf023bf35560a9daf1e@17.4.28.7:50600 CSeq: 103 INVITE User-Agent: Asterisk PBX 12.6.1 Authorization: Digest username="gw2", realm="provider.com", algorithm=MD5, uri="sip:89126975590@sip.provider.com", nonce="014d80ca", response="67bad8a0c97afc2b6747b471a56bca9f" Date: Wed, 29 Oct 2014 18:50:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 253 v=0 o=root 1098729670 1098729671 IN IP4 17.4.28.7 s=Asterisk PBX 12.6.1 c=IN IP4 17.4.28.7 t=0 0 m=audio 10088 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv Then I get dump from my kamailio (unsuccessfull INVITE) INVITE sip:89126975590@sip.provider.com SIP/2.0 Record-Route: <sip:sip.myservice.com:5068;nat=yes;ftag=as4684d4b9;lr=on> Via: SIP/2.0/UDP sip.myservice.com:5068 ;branch=z9hG4bK600b.1d5ff0fd59d4f3d2a1a06d722c0daa92.2 Via: SIP/2.0/UDP my.aterisk:50600;branch=z9hG4bK2b8d9b09;rport=50600 Max-Forwards: 70 From: <sip:gw2@sip.myservice.com:5068>;tag=as4684d4b9 To: <sip:89126975590@sip.provider.com > Contact:<sip:Vebinar-gw2@sip.myservice.com:5068> Call-ID: 445a7b884aeeab125d91886210c9beb7@sip.myservice.com:50600 CSeq: 102 INVITE User-Agent: SoftSwitch Date: Wed, 29 Oct 2014 22:32:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 Authorization: Digest username="gw2", realm="provider.com", nonce="10129bde", uri="sip:89126975590@sip.provider.com ", response="6d3411b24cbb57ad72271790ec01b453", algorithm=MD5 v=0 o=root 468654998 468654998 IN IP4 1.2.3.4 s=SoftSwitch c=IN IP4 1.2.3.4 t=0 0 m=audio 30104 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30105 I see difference between packetts only at SDP (not inportant things) and at VIA and request route Headers. All other fields identical. So -why Asterisk call successull and Kamailio kall unsuccessfull? What the differense?

2 2

websocket and SIP
by Ricardo Martinez 30 Oct '14

30 Oct '14

Hello. I’m having some problems using websocket to communicate a webRTC client with the SIP world. I have a Kamailio with a websocket port running on 5062, from that socket I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length. When I made the capture on the other leg (the pure SIP side) I only see a SIP INVITE with 1500 bytes. Seems that something is fragmenting the packet but not putting all the parts together. Could this be a problem with Kamailio?. Does someone has the same problem? Hope that someone could help me. Best Regards, Ricardo Martinez.-

3 5

Verified version of RTPproxy and Kamailio
by Juan Perez 30 Oct '14

30 Oct '14

Can somebody please tell me what would be a verified combination of kamailio and rtpproxy?I mean, for example, kamailio 4.1 and rtpproxy 1.x that can be downloaded from:http://<real_url> I found at least 3 different places where one can download rtpproxy: http://sourceforge.net/p/sippy/rtpproxy/ci/master/tree/ https://github.com/miconda/rtpproxy https://github.com/sippy/rtpproxy I downloaded the one from https://github.com/sippy/rtpproxy and have tested against kamailio 4.1, 4.2 and they are not compatible.It seems that the logi inside the function "send_rtpp_command" is not compatible with the rtpproxy version from https://github.com/sippy/rtpproxy and that causes kamailio to disable rtpproxy and it becomes useless. Second questions: do the feature added to the rtpproxy (parameter -A to use external public IP for devices behind NAT) is compatible with OPENSIPS only? I am about to test with kamailio 3.3, I will update this post later onthank you in advancejp

1 0

kamailio server as Load balancing and Failover
by Nandini madhu 30 Oct '14

30 Oct '14

Dear All, Greetings, I want to implement kamailio server as failover and load balancing. Is there any configuration needed in kamailio.cfg. As I have searched in blogs, For load balancing, Virtual IP is added and also heartbeat software is needed. Please let me Know how to configure kamailio server for load balancing and failover Waiting for response Thank you, Regards, Nandini

3 2

Wrong port in via, from and contact headers
by Eric Koome 30 Oct '14

30 Oct '14

This is my setup: UAC -> firewall (1:1 nat) -> Kamailio:5060 - > asterisk:5080 . | . sip providers I'm running kamailio 4.1.x & ast 11 based on aspitos kamailio configuration tutorial. My problem is port number in via, contact and from headers is 5080 instead of 5060 for calls to provider. Public ip is fine. I have used advertised_port=5060 but no change. Also tried record_route_preset(ip:5060), but no change. I can make calls but I would imagine I would miss acks & and reinvites. Any thoughts on what I'm missing? Eric

1 0

Problem initiating a call with dlg_bridge
by Paul Smith 29 Oct '14

29 Oct '14

I seem to be going round in circles… I am trying to use dlg_bridge() from the dialog module to establish a call between two SIP endpoints. I have tested with Snom phones and linphone soft phone with the same result. I get an outbound call to the first (from) end point, I answer the phone… and then nothing obvious happens. I can get this same result when using kamcmd or from within the kamailio.cfg. For example using kamcmd: kamcmd dlg.bridge_dlg sip:105@mykamailioip sip:106@mykamailioip sip:mykamailioip:5060 I expected this to send an invite from "controller" via my KamailioIP to my registered local subscriber 105 (from), and then send a re-invite to 105 so that 105 creates a call leg to 106 (to). The kamailio.cfg is pretty much the default. I am running 4.2 and rtpengine to proxy the RTP streams. Not sure if it is necessary or useful but I call setflag() at the start of the request_route() to set the Dialog flag. What else should I have included in the kamailio.cfg to make this work? Also is there any way to control the SDP in the initial invite from dlg_bridge. By default I see RTP/AVP, with alaw and ulaw…. I’d like to offer RTP/SAVP swell or instead of RTP/AVP. Thanks

3 7

Trouble with max-forwards
by Mike Dunton 29 Oct '14

29 Oct '14

Readers, I am having issues with passing the max-forward header to my freeswitch service. Here are the sip invites Here is a call coming from a kazoo box though my carrier Into kamailio. Kamailio shows it has Max-Forward after the sanitize check, but when it reaches freeswitch no Max-Forward Header. *KAMAILIO* kamailio[2687]: ERROR: <script>: After Sanitize - [INVITE sip:+1850764####@aio07-bandwidth.tresta-aio.com:5060 <http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/> SIP/2.0 Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Accept: application/sdp Allow: INVITE,ACK,CANCEL,BYE Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.2791522.0 Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.f8621604.0 Via: SIP/2.0/UDP 67.231.9.###:5060;branch=z9hG4bK04B40f39b0e10295cd4 From: <sip:+1NUMBERSCRUBED@67.231.9.59;isup-oli=0>;tag=gK043cdaa8 To: <sip:+1850764####@67.231.8.85 <sip%3A%2B18507646071(a)67.231.8.85>> Call-ID: 537168748_70701748(a)67.231.9.# <537168748_70701748(a)67.231.9.59>## CSeq: 1289511105 INVITE *Max-Forwards: 49* Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer X-FS-Support: update_display,send_info Supported: precondition Content-Length: 359 Content-Type: application/sdp Remote-Party-ID: <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>>;privacy=off;screen=no v=0 o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59 s=SIP Media Capabilities c=IN IP4 67.231.9.72 t=0 0 m=audio 23918 RTP/AVP 9 0 18 96 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 *FREESWITCH* recv 1479 bytes from udp/[10.1.13.123]:5060 at 18:31:26.312854: ------------------------------------------------------------------------ INVITE sip:+1850764###@aio07-bandwidth.tresta-aio.com:5060 <http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/> SIP/2.0 Record-Route: <sip:10.1.13.###;lr=on;ftag=gK043cdaa8> Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Accept: application/sdp Allow: INVITE,ACK,CANCEL,BYE Via: SIP/2.0/UDP 10.1.13.123;branch=z9hG4bKc892.1b308e66.0 Via: SIP/2.0/UDP 67.231.8.195;branch=z9hG4bKc892.2791522.0 Via: SIP/2.0/UDP 67.231.8.85;branch=z9hG4bKc892.f8621604.0 Via: SIP/2.0/UDP 67.231.9.59:5060;branch=z9hG4bK04B40f39b0e10295cd4 From: <sip:+1NUMBERSCRUBED@67.231.9.# <sip%3A%2B1NUMBERSCRUBED(a)67.231.9.59>##;isup-oli=0>;tag=gK043cdaa8 To: <sip:+1850764####@67.231.8.# <sip%3A%2B18507646071(a)67.231.8.85>##> Call-ID: 537168748_70701748(a)67.231.9.# <537168748_70701748(a)67.231.9.59>## CSeq: 1289511105 INVITE MaxForwards would go here? Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Supported: precondition Content-Length: 359 Content-Type: application/sdp Remote-Party-ID: <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>>;privacy=off;screen=no X-AUTH-IP: 67.231.8.195 v=0 o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59 s=SIP Media Capabilities c=IN IP4 67.231.9.72 t=0 0 m=audio 23918 RTP/AVP 9 0 18 96 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 Here is my kamailio default.cfg http://pastebin.com/vxdFe8n0 Can anyone point me in the right direction, and tell me why freeswitch isn't being passed this header? Both kamailio and freeswitch are on the same box in this example. Thanks for your help. Mike

2 1

404 Not here as response to BYE requests
by Marko Seidenglanz 29 Oct '14

29 Oct '14

Hello, We have a setup where Kamailio 4.2 is used in front of Asterisk as WebRTC Proxy doing the encryption and NAT Traversal. Everything works as expected, except that BYE Requests sent by the WebRTC Client are not forwarded by Kamailio to Asterisk. We use record routing. Instead Kamailio responds with a "404 Not here". INVITE Headers: Kamailio --> Browser: sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com SIP/2.0 Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 Via: SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK2d4f70db Max-Forwards: 70 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as32dd9e24 To: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com> Contact: <sip:anonymous@10.240.177.13:5060> Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.1 Date: Wed, 29 Oct 2014 10:04:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 676 BYE Headers Browser --> Kamailio: BYE sip:anonymous@anonymous.invalid SIP/2.0 Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 , SIP/2.0/UDP 10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK2d4f70db From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com >;tag=2TQ878KMAVLA43TXVZHNAWCWVKU6BLPBURF3 To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as32dd9e24 Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 0 BYE Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Reason: Q.850;cause=16 BYE Response Kamailio --> Browser: SIP/2.0 404 Not here Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bK65df.6fd637d055286a45aa6f3e12c5ac873c.0 , SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK51c69f33;received=80.255.2.37 From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com >;tag=47J6E76F5EVB583A682FQR799J6XDSU46MW8 To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as0b12a022 Call-ID: 683def08655393622c55299c1a7c92d5@10.240.177.13:5060 CSeq: 0 BYE Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 In the logs I can see the following messages: [loose.c:113]: find_first_route(): No Route headers found [loose.c:929]: loose_route(): There is no Route HF Does anybody know, why Kamailio may respond with 404 Not here? Do I have to send the BYE request directly to Asterisk? Kind regards, Marko

2 3

Kamailio and RTPProxy
by Juan Perez 29 Oct '14

29 Oct '14

I installed Kamailio 4.2 and the latest rtpproxy. When starting Kamailio I see these lines: Oct 27 21:54:26 ip-10-0-2-68 kamailio: DEBUG: <core> [sr_module.c:590]: load_module(): load_module: trying to load </usr/local Oct 27 21:54:26 ip-10-0-2-68 kamailio: DEBUG: <core> [sr_module.c:357]: register_module(): register PV from: rtpproxy Oct 27 21:54:26 ip-10-0-2-68 kamailio: DEBUG: <core> [cfg.lex:1724]: pp_define(): defining id: MOD_rtpproxy Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: ERROR: rtpproxy [rtpproxy.c:1707]: send_rtpp_command(): timeout waiting reply f Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy <udp:10.0.2.68:77 Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: WARNING: rtpproxy [rtpproxy.c:1610]: rtpp_test(): support for RTP proxy <udp:10 Oct 27 21:54:37 ip-10-0-2-68 ./kamailio[3898]: ERROR: rtpproxy [rtpproxy.c:1707]: send_rtpp_command(): timeout waiting reply f Oct 27 21:54:37 ip-10-0-2-68 ./kamailio[3898]: ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy <udp:10.0.2.68:77 Oct 27 21:54:37 ip-10-0-2-68 ./kamailio[3898]: WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP I run rtpproxy like this: /usr/local/rtpproxy-install-dir/bin/rtpproxy -l <public ip here> -s udp:10.0.2.68:7722 -F -d DBUG:LOG_LOCAL3 I have these linee in the kamailio.cfg file loadmodule "rtpproxy.so" # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:10.0.2.68:7722") I have tried with localhost, with 127.0.0.1 and get the same results ( I mean configure kamailio to contact rttpproxy with localhost instead of ip and at the same time running rtpproxy with the same reference. I can see the rtpproxy running/listening [root@ip-10-0-2-68 sbin]# netstat -lnp | grep rtpproxy udp 0 0 10.0.2.68:7722 0.0.0.0:* 3608/rtpproxy I can also see these lines: [root@ip-10-0-2-68 sbin]# netstat -np | grep 7722 udp 0 0 10.0.2.68:49772 10.0.2.68:7722 ESTABLISHED 3928/./kamailio udp 0 0 10.0.2.68:36097 10.0.2.68:7722 ESTABLISHED 3897/./kamailio udp 0 0 10.0.2.68:49158 10.0.2.68:7722 ESTABLISHED 3906/./kamailio udp 0 0 10.0.2.68:37129 10.0.2.68:7722 ESTABLISHED 3904/./kamailio udp 0 0 10.0.2.68:39052 10.0.2.68:7722 ESTABLISHED 3926/./kamailio udp 0 0 10.0.2.68:38161 10.0.2.68:7722 ESTABLISHED 3919/./kamailio udp 0 0 10.0.2.68:53144 10.0.2.68:7722 ESTABLISHED 3887/./kamailio udp 0 0 10.0.2.68:39721 10.0.2.68:7722 ESTABLISHED 3905/./kamailio udp 0 0 10.0.2.68:59563 10.0.2.68:7722 ESTABLISHED 3909/./kamailio udp 0 0 10.0.2.68:51884 10.0.2.68:7722 ESTABLISHED 3921/./kamailio udp 0 0 10.0.2.68:46902 10.0.2.68:7722 ESTABLISHED 3930/./kamailio udp 0 0 10.0.2.68:33981 10.0.2.68:7722 ESTABLISHED 3931/./kamailio udp 0 0 10.0.2.68:60221 10.0.2.68:7722 ESTABLISHED 3898/./kamailio udp 0 0 10.0.2.68:35651 10.0.2.68:7722 ESTABLISHED 3900/./kamailio udp 0 0 10.0.2.68:33105 10.0.2.68:7722 ESTABLISHED 3927/./kamailio udp 0 0 10.0.2.68:43349 10.0.2.68:7722 ESTABLISHED 3887/./kamailio udp 0 0 10.0.2.68:39259 10.0.2.68:7722 ESTABLISHED 3899/./kamailio udp 0 0 10.0.2.68:35170 10.0.2.68:7722 ESTABLISHED 3907/./kamailio udp 0 0 10.0.2.68:38760 10.0.2.68:7722 ESTABLISHED 3908/./kamailio In the rtpproxy I see that it is sent data and it replies too, I even got a wireshark capture showing that as well: Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3931_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3927_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3926_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3928_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" But somehow kamailio does not see it: [root@ip-10-0-2-68 sbin]# ./kamctl fifo nh_show_rtpp which: no greadlink in (/sbin:/bin:/usr/sbin:/usr/bin:/) which: no gdb in (/sbin:/bin:/usr/sbin:/usr/bin:/) udp:10.0.2.68:7722:: set=0 index:: 0 disabled:: 1 weight:: 1 recheck_ticks:: 53253287 I even when to the kamailio code and saw something that may have something to do with it but havent tried it yet:, in the implementation of the function "send_rtpp_command" I see that the socket created is using TCP. But I tried configuring kamailio/rtpproxy to use tcp instead but I also got the same results, kamailio seems to fail to acknoledgwe the replies from rtpproxy. I also have SELINUX disabled and the iptables are not running. I dont know what else to check, if somebody can shed some light into this I'd appreciate it. jp2014

1 1

(no subject)
by Yuriy Gorlichenko 29 Oct '14

29 Oct '14

Hello. I use kamailio with last rtpengine and I have 5-7 Seconds voice delay. This happened only for from webphone. But it is not client issue as i see. Wireshark at client side shows that RTP starts as soon I pick up call. So rtp leaves rtpengine and goes to the destination with delay... I use WSS and think that problem at handshake. There are some statisticcs after call finished (as example). You may see that one of streams created after 5 seconds delay. Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] --- Tag 'rsik48leli', created 1:12 ago, in dialogue with 'as7b4cb593' Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34178 <> 8.2.10.25:52463, 340 p, 58032 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34179 <> 8.2.10.25:52468 (RTCP), 10 p, 960 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] --- Tag 'as7b4cb593', created 1:12 ago, in dialogue with 'rsik48leli' Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34194 <> 10.0.1.6:16376, 201 p, 36582 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34195 <> 10.0.1.6:16377 (RTCP), 1 p, 78 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] Final packet stats: Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] --- Tag 'as3af30098', created 1:17 ago, in dialogue with 'bqinihbhsf' Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------ Media #1, port 34142 <> 10.0.1.6:17258, 200 p, 35200 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------ Media #1, port 34143 <> 10.0.1.6:17259 (RTCP), 1 p, 78 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] --- Tag 'bqinihbhsf', created 1:12 ago, in dialogue with 'as3af30098' Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------ Media #1, port 34162 <> 8.2.10.25:52453, 216 p, 36768 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------ Media #1, port 34163 <> 8.2.10.25:52453 (RTCP),

2 1

Ubuntu upstart problem | Kamailio ready before mysqlDB
by Moritz Graf 29 Oct '14

29 Oct '14

Hi There, since 4.2 we had the problem that kamailio did not start after a reboot. A "/etc/init.d/kamailio restart" fixxed it. The problem is quite obvious: kamailio-loglile => Kamailio can't connect to mysqlDB 13:33:44 Oct 29 13:33:44 kamailio-test /usr/sbin/kamailio[1063]: DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@localhost/routing Oct 29 13:33:44 kamailio-test /usr/sbin/kamailio[1063]: ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Oct 29 13:33:44 kamailio-test /usr/sbin/kamailio[1063]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool mysql-logfile => ready for connections 13:33:45 141029 13:33:45 [Note] /usr/sbin/mysqld: ready for connections. Version: '5.5.40-0ubuntu0.12.04.1' socket: '/var/run/mysqld/mysqld.sock' port: 3306 (Ubuntu) So simply mysql is not ready for connnections when kamailio is ready. I "fixxed" this with a "sleep 5;" in the initfile. But this is not what I call a clean solution. Anyone else had the same problem and fixxed it a little cleaner? Maybe converted the init- to a upstart-script? regards, moritz -- Moritz Graf, M.Sc. Betrieb NGN-Plattform

1 0

rtpproxy_manage fails to rewrite SDP
by Igor Potjevlesch 29 Oct '14

29 Oct '14

Hello, I'm working with rtpproxy module for managing calls to a particular route. The thing is that for some calls, the "Connection information" is not rewritten properly whereas another calls, less than 1 second before or after, are properly rewritten. When "cr_route" defines that the call must be route to this carrier, I go to a route in order to make some change: - Keep only PCMA, PCMU, G729a and telephone-event - Depends on the R-URI: o If the uri contains IP addresses or domain name of the carrier: rtpproxy_manage("iefrcoz20","<CARRIER_IP_ADDRESS"); o else : rtpproxy_manage("eifrcoz20"); Kamailio listens on <CARRIER_IP_ADDRESS> and also on an internal one. A dedicated instance of RTPProxy runs with the following option: "-l <INTERNAL_IP_ADDRESS>/<CARRIER_IP_ADDRESS>". The idea is to tell to my customers to send their RTP to the internal IP address, and to tell to the carrier to send the RTP to the IP address that they have attributed to me. For hundred and hundred calls, this mechanism is working fine. But sometimes, no, the SDP remains as the original one. I don't know how to diagnose this. Could it be a saturation of RTPProxy? Regards, Igor.

2 5

content-length mismatch for kamailio 4.2.x
by andrew 29 Oct '14

29 Oct '14

hi all, kamailio 4.2.x has been released, And I used two endpoint to test kamailio server. I install kamailio 4.2.x on centos 6.5 via 'yum'. One endpoint failed to call another, I captured packets via wireshark, and found that invite method is recognized as error sip signal because of content-length' mismatch. I examined the invite packets, and calculated content-length in invite signal. There is no mistake for the value of content-length. I guess it may be a bug in kamailio 4.2.x. But I don't know what causes kamailio to report the mismatch response. who can give me help? I have uploaded the captured packets as an attachment. Below are two snapshots from the captured packets. B.R. andrew

2 1

an issue on terminating dialog for kamailio 4.2.x
by qw 29 Oct '14

29 Oct '14

hi, I install kamailio 4.2.x on centos 6.5 via 'yum'. One endpoint called another, where one dialog is created, and caller and callee can talk with each other. But when the caller hangup the phone(i.e. send bye to the callee), the callee can't receive the bye message, and the dialog can't be terminated normally. I have captured the packets via wireshark, and uploaded kamailio.cfg and the captured files as attachment. In the captured file, PRACK and BYE message can be sent to kamailio server successfully, but kamailio server doesn't forward the two messages to the callee. Why PRACK and BYE message can't be sent to the callee? Looking forward to your help! B.R. Andrew

2 1

Error while executing S-CSCF (Kamailio 4.2.0)
by Kamal Palei 28 Oct '14

28 Oct '14

Hi All I have followed link http://nil.uniza.sk/book/export/html/617 to install P-CSCF/I-CSCF/S-CSCF. Though I have some doubts in configuration , still I have put the configuration to best of my knowledge. When I execute SCSCF, I get below errors. I see some database related errors. As per the followed link, I find some DB configurations and I did accordingly. I do not see any DB configurations for SCSCF, still I see two below lines # Connection URL for the database: #!define DB_URL "con1=>mysql://scscf:secret@localhost/scscf" #!define DB_URL2 "con2=>mysql://scscf:secret@localhost/scscf" in file scscf.cfg Is this really required, or can I comment these two lines. *root@Kamal-E5430:/etc/kamailio/scscf# root@Kamal-E5430:/etc/kamailio/scscf# root@Kamal-E5430:/etc/kamailio/scscf# ./scscf -f kamailio.cfg -Eloading modules under config path: /usr/local/lib64/kamailio/modules/Listening on udp: 192.168.1.114 [192.168.1.114]:4080 tcp: 192.168.1.114 [192.168.1.114]:4080Aliases: *: scscf.group4-imslab11.in:* 0(25982) INFO: <core> [tcp_main.c:4745]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(25984) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(25984) INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available 0(25984) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Access denied for user 'scscf'@'localhost' (using password: YES) 0(25984) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool 0(25984) WARNING: db_cluster [dbcl_data.c:456]: dbcl_init_connections(): cannot connect to database - connection [con2] 0(25984) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Access denied for user 'scscf'@'localhost' (using password: YES) 0(25984) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool 0(25984) WARNING: db_cluster [dbcl_data.c:473]: dbcl_init_connections(): cannot connect to database - connection [con2] 0(25984) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Access denied for user 'scscf'@'localhost' (using password: YES) 0(25984) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool 0(25984) WARNING: db_cluster [dbcl_data.c:456]: dbcl_init_connections(): cannot connect to database - connection [con1] 0(25984) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Access denied for user 'scscf'@'localhost' (using password: YES) 0(25984) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool 0(25984) WARNING: db_cluster [dbcl_data.c:473]: dbcl_init_connections(): cannot connect to database - connection [con1] 0(25984) ERROR: <core> [db.c:401]: db_table_version(): error in db_query 0(25984) ERROR: <core> [db.c:440]: db_check_table_version(): querying version for table presentity 0(25984) ERROR: presence [presence.c:322]: mod_init(): error during table version check 0(25984) ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so)ERROR: error while initializing modules 0(25984) INFO: cdp [mod.c:257]: cdp_exit(): CDiameterPeer child stopping ... 0(25982) ERROR: <core> [daemonize.c:315]: daemonize(): Main process exited before writing to piperoot@Kamal-E5430:/etc/kamailio/scscf# root@Kamal-E5430:/etc/kamailio/scscf# * Best Regards Kamal

1 1

SCSCF Configuration
by Kamal Palei 28 Oct '14

28 Oct '14

Hi All I am trying to setup P-CSCF/I-CSCF/S-CSCF using Kamailio 4.2.0. I see configurations as below both in icscf.cfg and scscf.cfg files. *#!define HOSTNAME "icscf-1.imscore.org <http://icscf-1.imscore.org>"* *#!define NETWORKNAME "ims.imscore.org <http://ims.imscore.org>"* First line is *#!define HOSTNAME "icscf-1.imscore.org <http://icscf-1.imscore.org>"*Assuming *icscf-1* is the host name , *imscore.org <http://imscore.org> *is the domain name. Coming to second line *#!define NETWORKNAME "ims.imscore.org <http://ims.imscore.org>"* So it looks to me, NETWORKNAME is something like *ims.domainname* *Is this correct?* *OR* *I should put only domain name in NETWORKNAME* Please clarify. Best Regards kamal

1 0

Kamailio 4.2.0 - Error in default configuration
by Kamal Palei 28 Oct '14

28 Oct '14

Hi All I am in process of setting up of P-CSCF using Kamailio 4.2.0. I wanted to disable WITH_ANTIFLOOD (as I am not sure what shall be the content of dispatcher.list file for me, as it is juts a local setup). When I disable WITH_ANTIFLOOD, and try to execute I get below errors. Wherever *ds_is_from_list* is referred in Kamailio.cfg, I am getting error in those places. Can somebody pls suggest a work around when we disable macro WITH_ANTIFLOOD *root@Kamal-E5430:/etc/kamailio/pcscf# kamailio -f kamailio.cfg -Eloading modules under config path: /usr/local/lib64/kamailio/modules/ 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_is_from_list (params 0) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 486, column 23: unknown command, missing loadmodule? 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_is_from_list (params 0) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 658, column 45: unknown command, missing loadmodule? 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_is_from_list (params 0) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 672, column 23: unknown command, missing loadmodule? 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_is_from_list (params 0) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 706, column 37: unknown command, missing loadmodule? 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_select_dst (params 2) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 846, column 29: unknown command, missing loadmodule? 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_select_dst (params 2) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 993, column 29: unknown command, missing loadmodule?ERROR: bad config file (6 errors)* Thanks Kamal

3 7

Htable iterators, am I doing it wrong?
by davy van de moere 28 Oct '14

28 Oct '14

I was giving the htable iterator a go , but I can't get it to work. My relevant config would is: modparam("htable", "htable", "testhtable=>size=4;") in the request route I have this: sht_iterator_start("i1", "testhtable"); while(sht_iterator_next("i1")) { xlog("L_INFO"," $shtitkey(i1) is: $shtitval(i1)\n"); } sht_iterator_end("i1"); I added some values to my testhtable: $ kamcmd htable.dump testhtable { entry: 0 size: 2 slot: { item: { name: x[0] value: 1 } item: { name: x[1] value: 1 } } } But when I have something route over that part I get: Oct 27 15:33:36 kamailio[18828]: INFO: <script>: <null> is: <null> Oct 27 15:33:36 kamailio[18828]: INFO: <script>: <null> is: <null> So it is correct to find 2 items in the htable, but for both the key and value are not found. Tried it on both 4.2 and master. Thx!

2 2

Selecting rtpproxy instance on 200 reply
by Grant Bagdasarian 27 Oct '14

27 Oct '14

Hello, This was originally posted for opensips, but I figured someone here might know the answer as well since kamailio and opensips are "alike". I have the following RTPproxy configuration: modparam("nathelper", "rtpproxy_sock", "1 == udp:127.0.0.1:12221") modparam("nathelper", "rtpproxy_sock", "2 == udp:127.0.0.1:12222") In the main route I decide which instance to select for each INVITE containing SDP: if($ru=~"1.1.1.10") { set_rtp_proxy_set("1"); xlog("L_INFO", "USE RTP PROXY 01 \r\n"); } else if($ru=~"2.2.2.10") { set_rtp_proxy_set("2"); xlog("L_INFO", "USE RTP PROXY 02 \r\n"); } else { t_reply("404", "Not Found"); exit; } rtpproxy_offer(); I’m getting the following error for a 200 OK: ERROR:nathelper:select_rtpp_node: script error -no valid set selected ERROR:nathelper:force_rtp_proxy_body: no available proxies Do I need to do the same decision for 200 OK messages, and call rtpproxy_answer()? Regards, Grant

1 0

3.3.2 dumps core on unresolvable names
by Måns Nilsson 27 Oct '14

27 Oct '14

Hi, I've now twice experienced a core dump in my Kamailio install, where it tries to process an INVITE that is made to a domain name that does not exist. Log: Oct 27 14:49:58 landsort /usr/local/kamailio/sbin/kamailio[40058]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "ulndsort.sr.se" Oct 27 14:49:58 landsort /usr/local/kamailio/sbin/kamailio[40058]: ERROR: tm [ut.h:327]: failed to resolve "ulndsort.sr.se" Oct 27 14:49:58 landsort /usr/local/kamailio/sbin/kamailio[40058]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Oct 27 14:49:58 landsort /usr/local/kamailio/sbin/kamailio[40058]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) Oct 27 14:49:58 landsort /usr/local/kamailio/sbin/kamailio[40058]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction Oct 27 14:50:02 landsort /usr/local/kamailio/sbin/kamailio[40114]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 14 Oct 27 14:50:02 landsort /usr/local/kamailio/sbin/kamailio[40051]: ALERT: <core> [main.c:785]: child process 40058 exited by a signal 11 Oct 27 14:50:02 landsort /usr/local/kamailio/sbin/kamailio[40051]: ALERT: <core> [main.c:788]: core was generated Backtrack from coredump: (gdb) bt #0 t_unref (p_msg=0x8010af140) at atomic_x86.h:231 #1 0x0000000000509581 in exec_post_script_cb (msg=0x8010af140, type=Variable "type" is not available. ) at script_cb.c:195 #2 0x00000000004297fa in receive_msg ( buf=0x891ae0 "INVITE sip:vaxo-10@ulndsort.sr.se SIP/2.0\r\nVia: SIP/2.0/UDP 134.25.127.234:5060;rport;branch=z9hG4bKPjNOjNgRxGeWilvm07L79hdOmgzVAvYB78\r\nMax-Forwards: 69\r\nFrom: \"Teknisk Utveckling MTU 24\" <sip:mtu-24@"..., len=Variable "len" is not available. ) at receive.c:225 #3 0x00000000004b5579 in udp_rcv_loop () at udp_server.c:544 #4 0x0000000000455654 in main_loop () at main.c:1633 #5 0x0000000000459c33 in main (argc=Variable "argc" is not available. ) at main.c:2546 (gdb) Does this relate to any known issue? Should I just upgrade? FreeBSD 9, by the way. -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 You should all JUMP UP AND DOWN for TWO HOURS while I decide on a NEW CAREER!!

1 0

Kamailio for ARM with rtpproxy-ng
by Marino Mileti 27 Oct '14

27 Oct '14

Hi guys, I've cross compiled Kamailio (4.1.6) for ARM (Wandboard-IMX6). Everything seems works fine, but when I try to enable rtpproxy-ng module and Alignment trap error occurred just during rtpp_test (exchanging ping-ponge message betweek rtpproxy and rtpproxy-ng module of Kamailio) rtpengine[2258]: Got valid command from 127.0.0.1:50470: ping - { "command": "ping" } rtpengine[2258]: Returning to SIP proxy: d6:result4:ponge user.err kernel: [66832.254887] Alignment trap: not handling instruction edc30b08 at [<40b3a130>] user.alert kernel: [66832.262030] Unhandled fault: alignment exception (0x801) at 0x4033a782 local0.err kamailio: ERROR: <core> [daemonize.c:307]: daemonize(): Main process exited before writing to pipe In x86 architecture same packages works correctly without any problem. Does anybody encountered this kind of error? Marino ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-for-ARM-with-rtpproxy-ng-t… Sent from the Users mailing list archive at Nabble.com.

2 5

Kamailio 4.2.0 install issue
by Kamal Palei 27 Oct '14

27 Oct '14

Dear All I compiled Kamailio4.2.0 with below command make group_include="standard standard-dep stable experimental" all to compile all modules. When I install, I get below error touch /usr/local/lib64/kamailio/modules/counters.so install -m 755 counters.so /usr/local/lib64/kamailio/modules LD (gcc) [U kamcmd] kamcmd */usr/bin/ld: error: cannot find -lncurses* collect2: ld returned 1 exit status make[2]: *** [kamcmd] Error 1 ERROR: make --no-print-directory -C ../../utils/sercmd install-if-newer failed make[1]: *** [install-utils] Error 1 make: *** [install-modules] Error 1 am I missing any package here.., please let me know. Thanks Kamal

2 2

Error-While-Running-icscf.cfg
by Anil Kumar 27 Oct '14

27 Oct '14

Hi All, Please can anyone help me to solve this error, occurred while running icscf module . 0(4053) : <core> [cfg.y:3406]: yyerror_at(): parse error in config file /etc/kamailio-icscf/icscf.cfg, from line 52, column 1 to line 53, column 0: syntax error ERROR: bad config file (1 errors) Thanks Anil

2 1

How to setup Kamailio as P-CSCF/S-CSCF/I-CSCF
by Kamal Palei 27 Oct '14

27 Oct '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. Thanks kamal

2 5

Setting up web sockets and ephemeral auth
by Nolan Darilek 27 Oct '14

27 Oct '14

Having a hard time piecing together lots of pieces out of context. Here's what I want: 1. User logs in to my web app via its own authentication mechanisms. 2. Via an authenticated REST endpoint, my web app connects to the ephemeral auth service and returns a temporary username/password to the client. 3. The client connects to a web socket SIP interface, authenticating with the username/password it was given. 4. I'd also like to support authentication via standard SIP client using the user's default username/password, but that's a bit down the road still. Here is my kamailio-local.cfg. I'm using the web socket example copied from the docs, with a little cleanup and a few optional checks removed for now. loadmodule "db_mongodb.so" #!define DBURL "mongodb://mongo/perceptron" #!define WITH_AUTH #!define WITH_NAT loadmodule "xhttp.so" #modparam("xhttp", "url_match", "^/sip/") tcp_accept_no_cl=yes loadmodule "msrp.so" # Only required if using MSRP over WebSockets loadmodule "websocket.so" loadmodule "auth_ephemeral.so" modparam("auth_ephemeral", "secret", "Kamailio rulez!!11") event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); # xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm=~ "GET" ) { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { # xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { exit; } } xhttp_reply("404", "Not found", "", ""); } When I attempt to validate this, I get: 0(1) ERROR: <core> [pvapi.c:790]: pv_parse_spec2(): error searching pvar "hdr" 0(1) ERROR: <core> [pvapi.c:994]: pv_parse_spec2(): wrong char [U/85] in [$hdr( Upgrade)] at [5 (5)] 0(1) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /usr/local /etc/kamailio/kamailio-local.cfg, line 28, column 7-19: Can't get from cache: $h dr(Upgrade) ERROR: bad config file (1 errors) I don't know what that means. I've copied the config directly from the website. I tried quoting "Upgrade" and such in the $hdr lines, but that gives me: 0(1) ERROR: <core> [pvapi.c:790]: pv_parse_spec2(): error searching pvar "hdr" 0(1) ERROR: <core> [pvapi.c:994]: pv_parse_spec2(): wrong char ["/34] in [$hdr( "Upgrade")] at [5 (5)] 0(1) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /usr/local /etc/kamailio/kamailio-local.cfg, line 28, column 7-21: Can't get from cache: $h dr("Upgrade") ERROR: bad config file (1 errors) What am I missing here? I also had to comment out the xlog lines because those gave me errors. Do I need another module loaded to use xlog? Finally, does anyone have a working, complete example of SIP over web sockets with ephemeral auth integrated? I'm not sure if I should be checking the ephemeral auth credentials in the web sockets code, the SIP routes or somewhere else. I'm also not sure how I should set up authentication such that users can log in with both their standard credentials or those retrieved via the ephemeral auth module. Thanks.

2 1

Integrating Kamailio with existing IMS
by Muhammad Shahzad 27 Oct '14

27 Oct '14

Hi, I am trying add WebRTC support to existing IMS using kamailio. The idea is to let kamailio handle all webrtc calls alone with diameter backend to IMS for AAA. So, i thought of using auth_diameter and aac module for authentication and accounting. However auth_diameter docs say that the module is obsolete and will work no more. Also i am not sure about how to use diameter support in aac module. So, is there any any tutorial or how-to for this? I am aware of ims_* modules but i think they are meant to be used in case we want kamailio to act as IMS core (P-CSCF, S-CSCF etc.). Thank you.

1 0

best way to cope with peaks in traffic
by davy van de moere 27 Oct '14

27 Oct '14

Gents, assuming we don't use async methods (I know, it's against fashion), eventually you'll end up in having some bottlenecks in your config... Now thanks to a lot of children you can easily work around this, a maximum of 32 children gives you already quite something, and if you use the dispatcher, you can ramp that up to quite some capacity. ;) Now, there are moments where still limits are passed, so what would be the best way to keep your system healthy? I was thinking in the direction of using a hashtable value or $var to keep track of the children in use and e.g. if 31 children are used, refuse to process the 32th request (assuming 32 children are setup). Would this be a valid strategy? Or am I overthinking? Grtz, Davy

2 3

lcr module ping
by Debojit Kakoti 27 Oct '14

27 Oct '14

Hi, My current setup is I have one kamailio server and one asterisk server.In the asterisk server I have two trunk where there are two different PRI provider is used for outbound call. I am using uac_replace_from function to route call to different trunk of the same server. However I want to use lcr module and I can check if particular gateway is up and down using lcr module and route the call. But problem is In my current scenario the gateway is up but one PRI is down for one provider and other one is up.So how to check if one sip trunk is working and other one is not working? Can you suggest me a solution? -- Debojit Kakoti Mobile : +91 9650654870 Email : debojkakoti88(a)gmail.com Skype : deboj1988

2 1

rtpengine not working with append_branch
by Yuriy Gorlichenko 27 Oct '14

27 Oct '14

Hello all. I use rtpengine and rtpproxy-ng module at kamailio for proxying RTP and modifying SDP between endpoints. I use two types of clients - such as WSS based and UDP based clients. I have a trouble with append_branch and rtpengine handling for this packets. I try to implement this logic of my script Logic of my script: --checking location table for rows with needed account --get info from contact at loop for every step --check technology (sip or ws) --append_branch with existing destination for this account --rewrite packet with rtpengine to needed technology after loop forward packets via t_relay At listing bellow rtpproxy_manage have no flag "b" that implement needed functionality. It not included at the script because I tried include it but rtpengine says that ca nnot handle this request. As i understand this flag worked with rtpproxy and rtpproxy-ng apps only. Not with rtpengine. At my test I have 2 endpoints with WS and UDP phones (at fist step ir WS and 2 step it UDP). All calls going form asterisk via UDP. When I do these steps (at my script) I see packet at TCP dump and saw that sended only one packet to UDP but body of packet is WS. Then I saw log of kamailio. I see that at second step packet changed body to WS body (so strange because other steps before and after goes for UDP (as at logic of script)) If I set "b" flag I can call only UDP endpoint, but at log I see as I say that rtpengine can not do thomething with SIP packets and send with t_relay uncghanged UDP packet only. sql_query("ca", "select contact from location where username='$tU'", "ra"); xlog("rows: $dbr(ra=>rows) cols: $dbr(ra=>cols)\n"); if($dbr(ra=>rows)>0){ $var(i)=0; while($var(i)<$dbr(ra=>rows)){ xlog("L_INFO","SQL query return contact {$dbr(ra=>[$var(i),0])} for {$tU} at step {$var(i)}\n"); if ($dbr(ra=>[$var(i),0])=~"transport=ws"){ xlog("L_INFO", "This is a Websocket call to endpoint"); sql_pvquery("ca", "select received from location where contact='$dbr(ra=>[$var(i),0])'","$var(recieved)"); xlog("L_INFO","SQL query return recieved {$var(recieved)} for {$tU}\n"); $du=$var(recieved); xlog("L_INFO", "Request going FROM ASTERISK to WS. Destination is {$du}\n"); xlog("L_INFO","Websocket Destination URI is {$var(recieved)} for {$tU}\n"); rtpproxy_manage("froc+SP"); t_on_reply("REPLY_FROM_WS"); append_branch("sip:$tU@$du"); $var(i) = $var(i) + 1; } else { xlog("L_INFO", "This is a classic UDP call to endpoint"); $du="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); xlog("L_INFO","Classic Destination URI is {$dbr(ra=>[$var(i),0])} for {$tU}\n"); rtpproxy_manage("co"); t_on_reply("MANAGE_CLASSIC_REPLY"); append_branch("sip:$tU@$du"); $var(i) = $var(i) + 1; } #append_branch("sip:$tU@$du"); } } return 1; So it customised schema of standart example seturi("sip:a@example.com <https://e.mail.ru/compose/?mailto=mailto%3asip%253Aa@example.com>"); append_branch("sip:b@example.com <https://e.mail.ru/compose/?mailto=mailto%3asip%253Ab@example.com>"); append_branch("sip:c@example.com <https://e.mail.ru/compose/?mailto=mailto%3asip%253Ac@example.com>"); append_branch("sip:d@example.com <https://e.mail.ru/compose/?mailto=mailto%3asip%253Ad@example.com>"); t_relay(); I need fro advice how to fix this may be with another applications or with another steps. Thanks.

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