- sr-users - kamailio.org

diameter auth with round trip
by JB 06 Oct '15

06 Oct '15

Hello all, we are working on a SIP solution using Kamailio. We want to secure our base of user credentials even in case of attack on the SIP server, and for that reason we plan to use diameter authentication as described in RFC http://www.rfc-base.org/txt/rfc-4740.txt Paragraph 6.2 describes a mode where the HSS answer with code DIAMETER_MULTI_ROUND_AUTH ,and then validate user credential after a second round trip. This does NOT corresponds to what is done on Kamailio module ims_auth, where credentials (actually a hash of the credentials, but its enough to authenticate ) ) are pushed to kamailio, which does the computation of the expected answer (which corresponds to par 6.3 of the RFC 4740) Is there any kamailio module that would allow to use the method with DIAMETER_MULTI_ROUND_AUTH ? Thank you JB

2 1

Prepend values to PSTN gateway?
by Infinicalls Infinicalls 05 Oct '15

05 Oct '15

Hi, I am able to successfully connect to a PSTN gw from Kamailio. Now, he says I need to prepend a value (eg.123123) before the number(ENUM) and then only terminate the calls. Where do I add this? Can somebody help me? thanks Ganesh Kumar

3 3

Interoperability issues between browsers and mobile phones to place a call.
by Safdar Khan 05 Oct '15

05 Oct '15

Hi Daniel sir I have enabled web-sockets module into my kamailio sip server.I am trying to make call from web browsers and mobile phones. To make a call from web browser i am using sipml5 and as a mobile app i am using imsdroid. However i have done some search on net and i found nothing relevant. Any suggestions and help?.Waiting your response eagerly. Thank you

2 3

defaults missing from tcp ops module readme
by Juha Heinanen 05 Oct '15

05 Oct '15

after seeing some commits on tcpops module, i decided to check its readme and noticed that nothing is mentioned about the default values of parameters. it might be useful to tell that, e.g., if tcp_keepalive_enable is not set, then tcp_keepalive core parameter value tells what the status is. -- juha

2 2

problem with dispatcher load balancing
by Ding Ma 05 Oct '15

05 Oct '15

I'm trying to set up kamailio dispatcher to distribute calls to 2 asterisk servers. So far, the failover case seems ok, but I cannot get the dispatcher to distribute load. All calls are going to the last destination entry in the dispatcher table even if I have set the maxload attributes. I'm using algorithm 10 for load distribution. The number of calls sent to one asterisk is well above the maxload. Hope someone more experienced in dispatcher can review my config and give me some recommendations. Thanks in advance. Here is the content of dispatcher table in postgresql db. select * from dispatcher; id | setid | destination | flags | priority | attrs | description ----+-------+--------------------+-------+----------+---------------------------+------------- 1 | 1 | sip:10.0.1.31:5061 | 0 | 0 | duid=asterisk1;maxload=25 | Asterisk1 2 | 1 | sip:10.0.1.33:5061 | 0 | 0 | duid=asterisk2;maxload=25 | Asterisk2 (2 rows) Here are the dispatcher sections in kamailio.cfg. # ----- dispatcher params ----- #!ifdef WITH_DISPATCHER modparam("dispatcher", "db_url", DBASTURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "force_dst", 1). # If flag 2 is set, then failover support is enabled. modparam("dispatcher", "flags", 3) # the last address in destination set is used as a final option to send the request to modparam("dispatcher", "use_default", 1) # load balancing fail over modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") # PVs for hashing modparam("dispatcher", "hash_pvar", "$fU@$ci") # PVs to store results when calling ds_is_from_list modparam("dispatcher", "setid_pvname", "$var(setid)") modparam("dispatcher", "attrs_pvname", "$var(attrs)") # method to probe the gateways modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:dispatcher@localhost") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;code=488;class=3") modparam("dispatcher", "ds_probing_mode", 1) # size of hash table storing data for call load dispatching, power of two modparam("dispatcher", "ds_hash_size", 10) # expiration time in seconds to remove the load on a destination if no BYE was received modparam("dispatcher", "ds_hash_expire", 3600) # expiration time in seconds to remove the load on a destination if no 200 OK for INVITE was received # and state updated with ds_load_update modparam("dispatcher", "ds_hash_initexpire", 60) modparam("dispatcher", "ds_hash_check_interval", 30) #!endif route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing #!ifdef WITH_DISPATCHER if(is_method("BYE|CANCEL") && ds_is_from_list("1", "3")) ds_load_update(); #!endif ...... } } route[FROMASTERISK] { #!ifdef WITH_DISPATCHER if(ds_is_from_list("1", "3")) { xlog("L_DBG","$rm from $fU@$si:$sp: Call from Asterisk cluster\n"); return 1; } return -1; #!else if ($si==$sel(cfg_get.asterisk.bindip)) { return 1; } return -1; #!endif } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); #!ifdef WITH_DISPATCHER if(is_method("INVITE") && ds_is_from_list("1", "3")) { if(status=~"2[0-9][0-9]") { ds_load_update(); } else if(status=~"[3-7][0-9][0-9]") { ds_load_unset(); } } #!endif if(status=~"[12][0-9][0-9]") route(NATMANAGE); } #!ifdef WITH_DISPATCHER failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { # mark the destination Inactive and Probing ds_mark_dst("IP"); # select the new destination if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } else { # last available node failed to reply, no other destinations available send_reply("404", "No destination"); exit; } } } #!endif route[TOASTERISK] { #!ifdef WITH_DISPATCHER # ds_mark_dst("IP"); # Call load distribution if(!ds_select_dst("1", "10")) { sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","$rm from $fU@$si:$sp: No destinations available for $rd\n"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); #!else $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); #!endif route(RELAY); exit; } # Dispatcher detects a destination goes down event_route[dispatcher:dst-down] { xlog("L_ERR", "Destination down: $rm $ru ($du)\n"); } # Dispatcher detects a destination comes up event_route[dispatcher:dst-up] { xlog("L_ERR", "Destination up: $rm $ru\n"); }

2 10

Changed the bindport of Kamailio but still listening to older port
by amjad ali 04 Oct '15

04 Oct '15

Hi, I am scratching my head for one month to get this sorted and I have spent so many hours but couldn't figure out. Please help ! I have changed the bind port to 5099 but it is still listening to the previous port even I have rebooted the system. My Code is : kamailio.bindip = "192.168.1.105" desc "Kamailio IP Address"kamailio.bindport = "5099" desc "kamailio port" However, when I restart kamailio service it shows that it is still listening to 5060 (attached the screen shot) Regards,Amjad

2 1

Kamailio is not forwarding the call to asterisk
by amjad ali 04 Oct '15

04 Oct '15

Hi, I have kamailio and asterisk running on same machine. When I make an internal call, it routes the call to other extension. However, if I stop the asterisk service then the call still routes the same way to other extension. Is there a chance that kamailio does the call routing it self. Also, I cannot see any sip extension registered at asterisk when I run a command "Sip show peers" neither there is any activity in Asterisk's log. Regards,Amjad

2 1

Inconsistent string interpolation
by Sven Neuhaus 02 Oct '15

02 Oct '15

Hello, I'm occasionally running into issues with inconsistent string interpolation in the kamailio.cfg. Here's the latest example. Take this line: $var(x) = "sip:+$var(abc)-$var(def)-$si@$Ri:$Rp"; Here, the values of the variables will NOT be interpolated into the string. Instead, I have to use this: $var(x) = "sip:+" + $var(abc) + "-" + $var(def) + "-" + $si + "@" + $Ri + ":" + $Rp; However, when I have an instruction such as: append_hf("P-Asserted-Identity: $var(x)\r\n"); Then string interpolation takes place (i.e. the value of $var(x) is put into the string. This is very confusing and error prone. Is it something that can be fixed? The behaviour should be consistent. -Sven

2 6

Kamailio v4.3.3 Released
by Daniel-Constantin Mierla 02 Oct '15

02 Oct '15

Hello, Kamailio SIP Server v4.3.3 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.3. For more details about version 4.3.3 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/10/kamailio-v4-3-3-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

[ims_auth] compute authenticate field in diameter server, not kamailio
by JB 02 Oct '15

02 Oct '15

Hello all, we are working on a SIP solution using Kamailio. We want to secure our base of user credentials even in case of attack on the SIP server, and for that reason we plan to use diameter authentication as described in RFC http://www.rfc-base.org/txt/rfc-4740.txt Paragraph 6.2 describes a mode where the HSS answer with code DIAMETER_MULTI_ROUND_AUTH ,and then validate user credential after a second round trip. This does NOT corresponds to what is done on Kamailio ims_auth, where credentials (actually a hash of the credentials, but its enough to authenticate ) ) are pushed to kamailio, which does the computation of the expected answer (which corresponds to par 6.3 of the RFC 4740) Is there any kamailio module that would allow to use the method with DIAMETER_MULTI_ROUND_AUTH ? Thank you JB

1 0

reSIProcate v1.10.0 released
by Daniel Pocock 02 Oct '15

02 Oct '15

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 The v1.10.0 release has been tagged and made available for download: https://www.resiprocate.org/files/pub/reSIProcate/releases/ and a direct link to the ChangeLog: https://github.com/resiprocate/resiprocate/blob/resiprocate-1.10/ChangeLog Debian, Fedora and Ubuntu packages are in the process of being updated. It is about 20 months since 1.9.0 and there have been many improvements. Highlights of the v1.10.0 release: - - repro SIP proxy now includes: - full presence server - PostgreSQL support - Include directive in the config file - - enhancements to the stack and proxy include: - WebSocket authentication may now use URL parameters instead of cookies, client certs or DIGEST - better Syslog support - TLS improvements (PFS, private key passphrases) - better assertion handling (resip_assert) - WSAPoll for Windows, allowing thousands of TCP connections - removing transports at runtime - added netns (Linux network namespace) support The RTC Quickstart Guide provides an overview of how to get started using repro, Kamailio and other popular SIP, XMPP and TURN products http://rtcquickstart.org Contributors to v1.10.0 include Alan Hawrylyshen, Alexander G. Pronchenkov, Byron Campen, Daniel Pocock, Daniel Tacalau, Dan Petrie, Dario Bozzali, Fedor Brunner, Gregor Jasny, Kevin Green, Reiner Herrmann, Robert Sparks, Scott Godin, Thomas Thorne, Vasil Kolev and Wolfgang Rosenauer. Please also see the hall of fame in Github: https://github.com/resiprocate/resiprocate/graphs/contributors which emphasizes the enormous and sustained contribution Scott Godin has made to this project and this release in particular. We are also very grateful to all those who contributed feedback through the mailing list, testing the beta packages in Debian and all past contributors to previous releases of the reSIProcate project. reSIProcate point releases are now announced on the low-volume resiprocate-announce mailing list, please subscribe if you would like to know when 1.10.1 comes along: http://list.resiprocate.org/mailman/listinfo/resiprocate-announce SHA256 checksums for the files: resiprocate-1.10.0.tar.gz c089686e183b8a739064731fa9e5002ab4d0bd6908cd8f5e648e021d76363c98 resiprocate-1.10.0.zip df4ba3902770390e7a0f9f6b13c5959292feba04785f01b9d044ad31325fa1e0 resiprocate-contrib-1.10.0.tar.gz 2f40c568644cbf1dd8628da0535dadd8d17367860234163d003fe0fbd75ba9a0 resiprocate-contrib-1.10.0.zip ef1c50b7c70b533c7bec1bfb071ca19ff9368a0b1e8a951e430e273b8d413715 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1 iQIcBAEBCAAGBQJWDlraAAoJEOm1uwJp1aqDCxgP/jlJBE0fTSp1FzohA5IRmrmQ rVQcKOB6pI6rWTMEwv8WuBs6sA7z/z94b2jbLYZRo/iwXXb85TbQXqnIQkyw5GAs l+F73B+rPyNYN46CZvjYv/rrscW8rxDPoein/t7uEzfATm1xgAZrIykWhZx42WV2 QUEjZDAXRm2Q2duIBfnxyTmhyFurWgHJhtxLzx6zCVnml/hWt8xslzkH1WYtfXuI 0ltGqNuVlYlhHCi2/BpaxHVdsPELhdiTHpIu88UgBSHI5gKiUBp38dhQttwJr+bt VWF9Ri+KQPbn7d5JGc8mPMeBTlYmJUgFY4s//eIU2pYRcxZ2w1/iim2SHewMpift 51C9Lmp6vSBycpCNPAQ6yQqJdAjIlTlY9eKJ2td+bCUw66Zlx10h3tqY4ka95o0s TfFporu6dpaIu3TSo+qVW8fhXuYmRh86NHLLi3+LCbZ9i6j5E71qor5mkz9x/xLM 7kWbVJKNJFTdbTsVjAKZ7uh4YFfkn3NcYEu+5zwF5Fm/z6ohE36sme12Lo0khjyq kTDhefATpFvZLxu2l0RBmG+NktOj/H2MJvbPVGPyEtHaam1BJaRPv4UMh6bldoR4 vsrt8cL31HhnWGTZS7M7/yg39Ge+JrTAHQD8wpdVh6nBzFTqb2n2WIk2swPICrLA v8lx2bBfol6NYvB0GhAT =R1HL -----END PGP SIGNATURE-----

1 0

Siremis-4.2.0 - error with Alias and login page
by Infinicalls Infinicalls 02 Oct '15

02 Oct '15

Hi, I tried installing siremis-4.2.0 following the step given.The Alias was not working. So, I installed it in the /html folder and the installation was completed. But the login page looked strange. There was no logo displayed and the submit button was not there. Mine is apache 4.2 on Ubuntu LTS 14. Kindly let me know how to solve this problem. thanks Ganesh Kumar

2 2

how to use data received from diameter answer(CCA) in kamailio config script
by SureshTummala 02 Oct '15

02 Oct '15

Hi, I want retrieve the data present in the Credit-Control-Answer response from diameter in kamailio configuration script. Is there any provision in kamailio 4.1.x (or) later? Any suggestions are appreciated. Regards,Suresh Tummala

4 3

FW: Your iptel.org Registration
by Bijal Desai 02 Oct '15

02 Oct '15

When I call on sip:bijaldesai@iptel.org, it is not working. Can you please guide me -----Original Message----- From: registrar(a)iptel.org [mailto:registrar@iptel.org] Sent: Tuesday, September 29, 2015 10:00 PM To: Bijal Desai <Bijal(a)ssminfotech.com> Subject: Your iptel.org Registration Thank you for registering with iptel.org. We are reserving the following SIP address for you: sip:bijaldesai@iptel.org To finalize your registration please check the following URL within 24 hours: http://serweb.iptel.org/user/reg/confirmation.php?nr=473f1e2fc74bfb3e9d28eb… (If you confirm later you will have to re-register.) Here are the details of your account: --------------------------------------------------------------------- Username: bijaldesai(a)iptel.org Password: Angel1987@ Email: bijal(a)ssminfotech.com Full Name: Bijal Desai Sip address: sip:bijaldesai@iptel.org (You can always retrieve these via the "Forgot Password" link on the signup page) -- This message is automatically generated by SerWeb.

2 1

Kamailio-Asterisk real-time integration with Dynamic IP
by Rudy Eschauzier 02 Oct '15

02 Oct '15

All, The Kamailio-Asterisk integration that is described here asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb [Asipto - SIP and VoIP Knowledge Base Site] is limited to Asterisk servers with a fixed IP address. The reason is that Kamailio identifies and authorizes traffic from Asterisk based on the source IP address. Attached is a patch to kamailio.cfg that adds support for dynamic Asterisk hosts. It introduces the option to define a fully-qualified domain name instead of the IP address to identify the Asterisk server. The host name can be dynamically updated to always point at the (changing) IP address of the Asterisk server. Defining WITH_ASTERISK_FQDN enables the new option. When WITH_ASTERISK_FQDN is set, Kamailio will perform a DNS search on the host name in the asterisk.bindhost variable. It will authorize incoming traffic if it originates from one of the IP addresses the DNS lookup returns (and matches the asterisk.bindport variable as before). Furthermore, the patch introduce the option to automatically set the IP address of the server that Kamailio is running on. This does away with the need to properly define kamailio.bindip on every server that the configuration is running on. This option is enabled by defining WITH_AUTOIP. With the suggested changes I have been able to successfully run an Asterisk server in my home, having a dynamic public IP address and connecting to the Kamailio proxy on a VPS server with a static IP addres. Please let me know if you have any comments or suggestions. Kind regards,Rudy Eschauzier. | | | | | | | | | | | asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb [...Table of Contents Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database Architecture Registration Call Initiation Requirements MySQL Insta... | | | | View on kb.asipto.com | Preview by Yahoo | | | | |

2 1

cnxcc module
by Максим Панчоха 02 Oct '15

02 Oct '15

Hello! I'm trying to set max call duration, but after 5 seconds call is continues. Kamailio version: 4.3.2 Here's part of my config file: #!define CC_FLAG 29 ... modparam("cnxcc", "credit_check_period", 1) modparam("cnxcc", "redis", "addr=127.0.0.1;port=6379;db=1") modparam("dialog", "hash_size", 2048) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) ... if ((is_method("INVITE")) && (is_request()) && (!has_totag())) { route(CNXCC); } route[CNXCC] { setflag(DLG_FLAG); if (!is_method("INVITE")) return; $var(client) = $fU; $var(max_time) = 5; if (!cnxcc_set_max_time("$var(client)", "$var(max_time)")) { xlog("Error setting up credit control"); return; } } event_route[dialog:failed] { xlog("L_INFO", "********END CALL*********$DLG_lifetime*******************"); } event_route[dialog:end] { xlog("L_INFO", "******END CALL************$DLG_lifetime****************"); } event_route[dialog:start] { xlog("L_INFO", "***********************DIALOG STARTED****************************"); } event_route[cnxcc:call-shutdown] { $var(duration) = $TS - $dlg(start_ts); xlog("L_INFO", *****END CALL***BALANCE IS EMPTY************$var(duration)*************"); } Tell me please, where I made a mistake. Thank you for a response.

2 1

ERROR: mediaproxy [mediaproxy.c:1978] #347
by Ngoc 02 Oct '15

02 Oct '15

Hello supporter, I took a look kamailio and get the error: Sep 29 11:28:37 lvps46-163-74-198 /usr/local/sbin/kamailio[1738]: ERROR: mediaproxy [mediaproxy.c:1978]: EngageMediaProxy(): engage_media_proxy requires the dialog module to be loaded and configured. How to resolved this problem, please help me. Regards, --------------------------- Dinh Hong Ngoc Technical Lead Edge-works Software Co.,ltd Hall 5, Quang Trung Software City, Tan Chanh Hiep Ward, Dist. 12, Ho Chi Minh City, Vietnam. Email: ngoc.dinh(a)edge-works.net Phone: +84 8 54371075 | Mobile: +84 909 911 406 Site: http://www.edge-works.net

2 1

receive_fd(): EOF
by jery john 02 Oct '15

02 Oct '15

11(12454) : <core> [mem/q_malloc.c:149]: qm_debug_frag(): BUG: qm_*: fragm. 0x7f9e8fcb2448 (address 0x7f9e8fcb2478) end overwritten(c0c0c000, abcdefed)! 13(12456) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 20 0(12443) ALERT: <core> [main.c:784]: handle_sigs(): child process 12454 exited by a signal 6 0(12443) ALERT: <core> [main.c:787]: handle_sigs(): core was generated 0(12443) INFO: <core> [main.c:799]: handle_sigs(): terminating due to SIGCHLD 13(12456) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 12(12455) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 9(12452) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 10(12453) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 8(12451) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 4(12447) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 3(12446) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 7(12450) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 6(12449) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 5(12448) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 2(12445) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 1(12444) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 0(12443) INFO: <core> [sctp_core.c:48]: sctp_core_destroy(): SCTP API not initialized using kamailio 4.2 with modified tcp_read.c and tcp_main.c files for buffer manipulation both files attached gdb output Reading symbols from /usr/local/sbin/kamailio...done. warning: exec file is newer than core file. [New LWP 15502] warning: Error reading shared library list entry at 0x79e8c789480056e4 warning: Error reading shared library list entry at 0x482a50b70fd8458b Core was generated by `/usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -P /var/run/ka'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x000000000061f73d in _shm_resize (p=0x61f729 <_shm_resize+541>, s=32767, file=0x3f <error: Cannot access memory at address 0x3f>, func=0x0, line=-1433800800) at mem/shm_mem.c:111 111 DBG("WARNING:vqm_resize: resize(0) called\n");

2 1

blf problem
by huseyin kalyoncu 01 Oct '15

01 Oct '15

Hi I am trying to implement BLF functionality as described in http://kb.asipto.com/kamailio:presence:k43-blf. I am using the same conf with the one in the tutorial except that i am using postgres database instead of mysql. The problem is BLF is not working with this config (presence server and proxy server are in the same kamailio with same IP). But when I separate presence server from proxy server (2 different kamilio boxes with different IPs) BLF working perfect. i can see PUBLISH & NOTIFY messages in the tcpdump output with the working configuration. But i can not see any PUBLISH messages in the tcpdump output with the original config (one with presence and proxy servers are same). My question in the normal working environment (when BLF working perfectly); should i see PUBLISH and NOTIFY messages in the tcpdump output when the presence and proxy servers are the same? thanks Huseyin

2 9

SIP and conntrackd experience?
by Grant Bagdasarian 01 Oct '15

01 Oct '15

Hello, Does anyone have experience with SIP and conntrackd? Our media and signaling is separated. Does this work for VPN connections where SIP is sent over the VPN but the media over public internet? Thanks, Grant

2 5

modifying $fu / $fU (R-URI From)
by Sven Neuhaus 01 Oct '15

01 Oct '15

Hello, I'm having some troubles with modifying $fU in my kamailio.cfg. The following line appears to have no effect: $fU = $(fU{s.substr,2,0}); I would expect it to remove the first two leading characters from $fU. Has anyone else seen this? I have the same issue with modifying $fu. We're not using the uac module. And we're using the latest stable version of Kamailio. Thanks, -Sven

4 4

out of memory Error
by Yasin CANER 01 Oct '15

01 Oct '15

3 15

Crash kamailio 4.3.1
by Thibault Gueslin 30 Sep '15

30 Sep '15

After upgrading to kamailio 4.3.1 and modifying the config, I get this crash very often: ... CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 21 ALERT: <core> [main.c:728]: handle_sigs(): child process 9341 exited by a signal 11 ALERT: <core> [main.c:731]: handle_sigs(): core was not generated INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLD INFO: <core> [main.c:794]: sig_usr(): signal 15 received INFO: <core> [main.c:794]: sig_usr(): signal 15 received INFO: <core> [main.c:794]: sig_usr(): signal 15 received ... Is it known ? or may be related to my config ? Thank you Thibault

3 23

db_text and caching/reloading
by Asgaroth 30 Sep '15

30 Sep '15

Hi, I am busy testing the db_text module for some modules who's configuration does not change that often. If I set the db_mode for db_text to 0 (caching), it appears that if I modify the file directly (in this case the address dbtext file) and then issue a "kamctl address reload" the updated contents are *not* reloaded. However, if I set db_mode to 1, then the contents of the db_text file automatically reload when I change it on the file system. Is it possible to run db_text in caching mode but to also force a reload if I manually update the underlying database file? Additionally, in kamctlrc, I had to add the following variable: DBTEXT_PATH="/etc/kamailio/db_text" Whereas the original example for db_text path in kamctlrc is as follows: # database path used by dbtext, db_berkeley or sqlite #DB_PATH="/etc/kamailio/db_text" Is this just a typo in the kamctlrc distribution or is the variable meant to be DB_PATH or DBTEXT_PATH? Thanks

1 3

ACK and BYE are forwarded statelessly (failed to parse Route HF)
by Miroslav Dunaev 30 Sep '15

30 Sep '15

Hello everyone, I have a setup where kamailio acts as UAC to a SIP provider. Kamailio is also configured to pass all incoming calls to an Asterisk. Here is a diagram: Phone<-->Asterisk(10.10.11.232)<--->(PRIV:10.10.11.238)Kamailio(PUB:44.44.44.148)<--->(194.213.29.92)SIP-provider. When a call comes in from the SIP-provider, it gets all the way to the phone. But when the SIP-provider sends BYE or ACK (during hungup event or 200OK/SDP asterisk response.). I get this errors in my kamailio log: siputils [checks.c:100]: has_totag(): totag found <core> [parser/parse_rr.c:49]: do_parse_rr_body(): parse_rr_body(): No body for record-route rr [loose.c:106]: find_first_route(): failed to parse Route HF rr [loose.c:929]: loose_route(): There is no Route HF tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=10101, isACK=1 tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed ..... .... .... siputils [checks.c:100]: has_totag(): totag found tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=2 , global msg id=1 , T on entrance=0xffffffffffffffff tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=10101, isACK=1 tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found tm [t_funcs.c:279]: t_relay_to(): SER: forwarding ACK statelessly here is the link to: kamailio.cfg, ngrep -i any capture, debug_level6 log: http://labmir.com/kamailio/ Looking at the SIP ACK that comes from SIP provider I can see that they use strict routing. (i.e. TO field is set to kamailio's public IP that is next hop to them). Maybe I am wrong to judge that its not loose routing just because of the TO field. Anyways, because of (failed to parse Route HF) error, ACKs and BYEs are forwarded by kamailio statelessly, as a result kamailio uses wrong interface to forward them (SIP_dailog.txt line:233) and therefore asterisk never sees ACKs or BYEs. How can I make kamailio recognize ACKs and BYEs belonging to a dialog and pass them statefully through the right interface? It is worth to mention that I use exactly the same config with a different SIP-provider (which is peered to our PUB IP not registered through UAC like in this case) and it works flawlessly. Thank you, Mir

2 1

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