- sr-users - kamailio.org

Dialog module generates error since upgrade to 4.2.2
by Jan Hazenberg 26 Jan '15

26 Jan '15

Hi All, I'm running into a issue with the dialog module since the upgrade from 4.2.1 to 4.2.2. I use the dialog module to add extra vars to the cdr's generated by the ACC module. I use the following config: # ----- dialog params ----- modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "dlg_flag", FLT_DLG) modparam("dialog", "rr_param", "did") modparam("dialog", "wait_ack", 1) # Create dialog if (method=="INVITE") { # Create Dialog dlg_manage(); # Add test dialog var $dlg_var(src_ua) = $hdr(User-Agent); } This seems to work fine on kamailio 4.2.1 but after the upgrade i see the following errors in the logs: Jan 22 13:16:57 sip /usr/sbin/kamailio[2773]: CRITICAL: dialog [dlg_hash.c:901]: log_next_state_dlg(): bogus event 6 in state 1 for dlg 0x7f2f0332ed40 [3693:11222] with clid '9192c5fc24627a14c2ec42f084a96587(a)192.168.149.126' and tags '4017642921' '' Jan 22 13:17:01 sip /usr/sbin/kamailio[2770]: CRITICAL: dialog [dlg_hash.c:901]: log_next_state_dlg(): bogus event 7 in state 1 for dlg 0x7f2f0332ed40 [3693:11222] with clid '9192c5fc24627a14c2ec42f084a96587(a)192.168.149.126' and tags '4017642921' '' If i'm correct "bogus event 6 in state 1" indicates that a ACK whas received while the dialog still was in unconfirmed state. Should i catch the 200 OK on the INVITE to update the dialog? Thanks, Jan Hazenberg

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ims_charging module
by Shane Harrison 26 Jan '15

26 Jan '15

Hi all, I am exploring the use of the ims_charging module having set up a freediameter server (all very new to me). I have loaded and configured the dependencies i.e. cdp, dialog_ng, tm, cdp_avp but the startup gets stuck on: Jan 26 03:57:05 host-172-16-0-82 /usr/sbin/kamailio[2931]: ERROR: ims_charging [mod.c:269]: mod_init(): Could not import ul_register_ulcb Jan 26 03:57:05 host-172-16-0-82 /usr/sbin/kamailio[2931]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module ims_charging (/usr/lib64/kamailio/modules/ims_charging.so) The corresponding code in mod.c is: /*Register for callback of URECORD being deleted - so we can send a SAR*/ if (ul.register_ulcb == NULL) { LM_ERR("Could not import ul_register_ulcb\n"); return -1; } But unfortunately this doesn't mean much to me - what is a URECORD, what is a SAR? Any words of wisdom around what a "ul.register_ulcb" is about would be appreciated. Kind regards Shane Shane Harrison Senior Software Engineer Imagination Technologies NZ Limited Level 2 1 Market Grove Lower Hutt, 5010 New Zealand PO Box 30-449 Lower Hutt, 5040 New Zealand Phone: +64 4 890-3681 ext 3361

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How to forward sip message "statefully" so that the dialogue is preserved
by aft 26 Jan '15

26 Jan '15

We want to implement the following topology : INVITE--->kamailio1-----forward--->kamailio2--->route[NATMANAGE]--->Next Sip endpoint Client <---- route[NATMANAGE]<--------kamailio1<------forward-----kamailio2<---- reply with SDP So the caller to callee will be processed by one kamailio and callee to caller will be processed by another kamailio. Both kamailio will be up with a public IP. How this can be achieved?

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lookup_branches and rtpengine_manage
by Yuriy Gorlichenko 26 Jan '15

26 Jan '15

Hello. I need parallel forking calls with the same username. (Call to all contacts with name for example User123), my endpoints may be WebSocket based and standart UDP endpoints. And I use rtpengine_manage for nmanaging calls wor webphones and standart softh/hard phones. I get all contacts manually and than at the branch route set rtpengine_manage settings for every call. It works fine but it works for one kamailio server. When I use 2 kamailio servers as load balansers Server that handle call get all endpoints from location but call to only one, that registred only at ths server for example I call user123 I have 3 contacts user123(a)1.2.3.4 - was registered at kamailio 1 user123(a)3.2.1.4 - was registered at kamailio 2 user123(a)4.3.2.1 - was registered at kamailio 1 So if call goes through kamailio 1 it call only to user123(a)1.2.3.4 and user123(a)4.3.2.1 I use this settings for usrloc at 2 kamailios to share all table between 2 servers modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "user_column", "username") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "cflags_column", "cflags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "path_column", "path") modparam("usrloc", "ruid_column", "ruid") modparam("usrloc", "instance_column", "instance") modparam("usrloc", "use_domain", 1) and this code for calling them [GET_CONTACTS] { 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)"); $du=$var(recieved); xlog("L_INFO","SQL query return recieved {$var(recieved)} for {$tU}. Destination is {$du}\n"); append_branch("sip:$tU@$(du{s.select,1,:})"); } else { xlog("L_INFO", "This is a classic UDP call to endpoint"); $var(recieved)=''; sql_pvquery("ca", "select received from location where contact='$dbr(ra=>[$var(i),0])'","$var(recieved)"); xlog("L_INFO", "SQL query return RECIEVED {$var(recieved)}"); if ($var(recieved)==0){ xlog("L_INFO", "Recieved string is EMPTY"); $du="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); } else { xlog("L_INFO", "Recieved string is {$var(recieved)}"); $du=$var(recieved); } $var(UDP_contact)="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); append_branch("sip:$tU@$(du{s.select,1,:})"); xlog("L_INFO","Classic Destination URI is {$dbr(ra=>[$var(i),0])} for {$tU}}. Destination is {$du}\n"); } $var(i) = $var(i) + 1; } } t_on_branch("1"); return; } } } branch_route[1]{ if($du=~"transport=ws"){ xlog("L_INFO","Websocket Branch is {$du} for {$tU}\n"); rtpengine_manage("internal extenal force trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); t_on_reply("REPLY_FROM_WS"); } else{ xlog("L_INFO","UDP Branch is {$du)} for {$tU}\n"); rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } When it try to branch endpoint without registration at server that handle call I get errors that tm module can not build Via header **via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found* *ERROR: <core> [msg_translator.c:1725]: build_req_buf_from_sip_req(): could not create Via header* *ERROR: <core> [forward.c:607]: forward_request(): ERROR: forward_request: building failed* * UDP calls get errors something like above (sorry than can not share error code, This situation not often). So I think I have this trouble because I use manually handling call and tried to substitute to lookup_branches function. but I have no Idea how to set rtpengine_manage paraments for each endpoint depending this is websocket or standart call. IF there is write problem for callings thhrough 2 kamailios as loadbalansers pleas let me know about how to set rtpengine_manage parametrs wor endpoints for every fork. If not- can you tell me how I can call to all endpoints endepending of registration server (kamailio 1 or 2). Thanks.

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Re: [SR-Users] remote mysql server
by Slava Bendersky 25 Jan '15

25 Jan '15

Hello Everyone, Thank you for help, fixed by adding to dialog module parameters. modparam("dialog", "db_url", DBURL) Slava. From: "Carsten Bock" <carsten(a)ng-voice.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, January 25, 2015 8:36:52 AM Subject: Re: [SR-Users] remote mysql server Hi Slava, is most modules a default database connection is defined. Probably this is the case here. As this error is coming from the dialog module: Did you configure some sort of "db_mode"? e.g.: modparam("dialog", "db_mode", 1) Thanks, Carsten 2015-01-25 9:27 GMT+02:00 Slava Bendersky <volga629(a)skillsearch.ca>: > Hello Everyone, > > I am trying start kamailio 4.2 with remote mysql server and on debug I see > DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening > connection: mysql://xxxx:xxxx@localhost/kamailio > but in cfg defined actual mysql server ip and not localhost > > 0(28885) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): > find_export_record: found <db_bind_api> in module db_mysql > [/usr/lib64/kamailio/modules/db_mysql.so] > 0(28885) DEBUG: <core> [db.c:213]: db_bind_mod(): using db bind api for > db_mysql > 0(28885) DEBUG: <core> [db.c:318]: db_do_init2(): connection 0x7faca59eced8 > not found in pool > 0(28885) DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): > opening connection: mysql://xxxx:xxxx@localhost/kamailio > 0(28885) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): > driver error: Can't connect to local MySQL server through socket > '/var/lib/mysql/mysql.sock' (2 "No such file or directory") > 0(28885) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection > to the pool 0(28885) ERROR: dialog [dlg_db_handler.c:147]: init_dlg_db(): > unable to connect to the database > > > Slava. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-0 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users oon why I see in log connection attempt to localhost ? From: "Carsten Bock" <carsten(a)ng-voice.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, January 25, 2015 8:36:52 AM Subject: Re: [SR-Users] remote mysql server Hi Slava, is most modules a default database connection is defined. Probably this is the case here. As this error is coming from the dialog module: Did you configure some sort of "db_mode"? e.g.: modparam("dialog", "db_mode", 1) Thanks, Carsten 2015-01-25 9:27 GMT+02:00 Slava Bendersky <volga629(a)skillsearch.ca>: > Hello Everyone, > > I am trying start kamailio 4.2 with remote mysql server and on debug I see > DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening > connection: mysql://xxxx:xxxx@localhost/kamailio > but in cfg defined actual mysql server ip and not localhost > > 0(28885) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): > find_export_record: found <db_bind_api> in module db_mysql > [/usr/lib64/kamailio/modules/db_mysql.so] > 0(28885) DEBUG: <core> [db.c:213]: db_bind_mod(): using db bind api for > db_mysql > 0(28885) DEBUG: <core> [db.c:318]: db_do_init2(): connection 0x7faca59eced8 > not found in pool > 0(28885) DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): > opening connection: mysql://xxxx:xxxx@localhost/kamailio > 0(28885) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): > driver error: Can't connect to local MySQL server through socket > '/var/lib/mysql/mysql.sock' (2 "No such file or directory") > 0(28885) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection > to the pool 0(28885) ERROR: dialog [dlg_db_handler.c:147]: init_dlg_db(): > unable to connect to the database > > > Slava. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-0 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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remote mysql
by Slava Bendersky 25 Jan '15

25 Jan '15

Hello Everyone, I am trying start kamailio 4.2 with remote mysql server and on debug I see DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@localhost/kamailio but in cfg defined actual mysql server ip and not localhost Any help thank you, Slava.

3 2

CDRs and MongoDB
by Mickael Marrache 25 Jan '15

25 Jan '15

Hi, Is it possible to write CDRs directly to MongoDB using the new db_mongodb module? modparam("acc", "db_url", DB_MONGO_URL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_extra", " . ") modparam("acc", "cdr_log_enable", 0) modparam("acc", "cdrs_table", "cdrs") In the acc module documentation, I found the following statement: Note that CDR generation does not involve any kind of database storage (yet). In order to persist the CDRs into a database you will have to set up an exterior process (i.e., a script living outside of Kamailio) and implement the storage task yourself. Thanks, Mickael

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Wrong dialog selected in on_reply/failure route in case of spirals.
by Julia Boudniatsky 25 Jan '15

25 Jan '15

After kamailio upgrade from v3.3 to v4.1.7 in production ~5% of calls with spirals dialog are failed with wrong dialog variables in on-reply/failure routes. The problem has been reproduced in the LAB (load test generated by SIPP). Please see the attached file with debug3 log. Scenario: Proxy Location Kamailio P1 Nbr translation Redirect | INVITE | | | dlg 0x7f8f8501aba0 | | | Call-ID:(5-8286@10.25.153.150) | | |------------------------------->| | | 100 Trying | | | dlg 0x7f8f8501aba0 | | | Call-ID:(5-8286@10.25.153.150) | | |<-------------------------------| | | | | | INVITE | | | dlg 0x7f8f8501aba0 | | | Call-ID:(5-8286@10.25.153.150) | | |<-------------------------------| | | 100 Trying | | | dlg 0x7f8f8501aba0 | | | Call-ID:(5-8286@10.25.153.150) | | |------------------------------->| | | | | | INVITE | | | dlg 0x7f8f8501aba0 | | | Call-ID:(5-8286@10.25.153.150) | |-------------------------------------------------------------->| | 100 Trying | | dlg 0x7f8f8504cda0 !!!!!!!! | | Call-ID:(5-8286@10.25.153.150)| |<--------------------------------------------------------------| | 301 Moved Permanently | | dlg 0x7f8f8504cda0 !!!!!!!! | | Call-ID:(5-8286@10.25.153.150)| |<--------------------------------------------------------------| | | Thank you, Julia.

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mistakenly detected NAT in IPv6 setup
by Sebastian Damm 23 Jan '15

23 Jan '15

Hi, it looks to me that Kamailio detects NAT without NAT being there, when used in IPv6 scenarios. I have a register, which comes in like this: U 2015/01/23 11:42:31.013284 2a01:1234:123:1234::2:16732 -> 2001:9876:98:9876::aa01:5060 REGISTER sip:domain SIP/2.0 Via: SIP/2.0/UDP [2a01:1234:123:1234::2]:16732;branch=z9hG4bK45528eba After it has run through Kamailio, the packet is sent out like this: U 2015/01/23 11:42:31.015126 1.2.3.4:5060 -> 9.8.7.6:5060 REGISTER sip:domain SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK8d21.7fa5fb060c2acc9583f05a93b93f9873.0 Via: SIP/2.0/UDP [2a01:1234:123:1234::2]:16732;received=2A01:1234:123:1234:0:0:0:2;branch=z9hG4bK45528eba As far as I know, the received part should only get appended, if the source IP of the packet differs from the IP advertised in the topmost Via header. But to me it looks like it always gets appended. In just IPv4 scenarios, the received parameter only gets appended if a difference is detected. And if it gets appended, shouldn't the received parameter IP be in square brackets, too? Later on, we have something like this in our code: if(method=="REGISTER") { if (nat_uac_test("23")) { append_hf("X-nathint: nat\r\n"); force_rport(); if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); }; } else { if (!add_path()) { sl_send_reply("503", "Internal Path Error"); }; }; } In our code, Kamailio always jumps into the add_path_received() part, so I guess, NAT detection triggers. But why? And then the Path header appended looks like this: Path: <sip:1.2.3.4;lr;received=sip:2A01:1234:123:1234:0:0:0:2:16732> The path module probably doesn't know anything about IPv6, too, so the IP in the received parameter is not enclosed in square brackets. When using the Path header later to send packets to the client, Kamailio complains about a parse error for the destination URI again. I tried looking into the path module, but I guess my C skills aren't sufficient to submit a patch. Best regards, Sebastian

2 2

Siremis Installation Web Page NOT Displayed.
by Mahmoud Ramadan Ali 23 Jan '15

23 Jan '15

Hi Dears, I'm trying to install Siremis 4.1 on Debian and whenever i open the URL in IE i get location.href='$url'"; exit; } ?> and in Firefox i see <?php if(is_file(dirname(__FILE__).'/install.lock')){ include 'bin/_forward.php'; }else{ $script_name = $_SERVER['SCRIPT_NAME']; $url = str_replace("index.php","install/",$script_name); echo "<script>location.href='$url'</script>"; exit; } ?> I've verified the permissions of - siremis/log - siremis/session - siremis/files - siremis/themes/default/template/cpl files without success also no logs gets written to the logs file...please help to fix this issue... Thanks in advance

1 0

RFC: best way to create dialogs
by Klaus Darilion 22 Jan '15

22 Jan '15

Hi! I use the dialog module to count and limit concurrent calls per user. It worked fine with 4.1.7 but fails with 4.2.2. My config basically looks like: route{ ... dlg_manage() ... authentication (stateless replies + exit) ... t_on_reply() t_on_branch() ... t_relay() exit; } initial_cbs_inscript uses the default value 1. Using 4.1.7, after stateless reply+exit, Kamailio executes dialog callbacks and deletes the dialog: [dlg_var.c:55]: dlg_cfg_cb(): new dialog with no trasaction after config execution [dlg_hash.c:872]: dlg_unref(): unref dlg 0xb233c3a0 with 1 -> 1 [dlg_hash.c:872]: dlg_unref(): unref dlg 0xb233c3a0 with 1 -> 0 [dlg_hash.c:872]: dlg_unref(): ref <=0 for dialog 0xb233c3a0 [dlg_hash.c:355]: destroy_dlg(): destroying dialog 0xb233c3a0 (ref 0) With 4.2.2, after stateless reply+exit, Kamailio just exits without calling any dialog callback (to destroy the dialog) I think I could work around the problem by creating the transaction later. Here are some ideas and I would be happy about your comments and best practices. - set initial_cbs_inscript to 0 - call dlg_manage() just before t_relay - do not use dlg_manage() but use the dialog flag and set the flag somewhere before t_relay Finally, do t_on_reply() and t_on_branch() already create the transaction or is it created with t_relay()? Thanks Klaus

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reINVITE with uac_req_send()
by Le Truong Giang 22 Jan '15

22 Jan '15

Hi ! I have the following simple scenario A <---> Kamailio <---> Asterisk <---> B now i'm trying to handle Timeout reply when B is unavailable. First i suspend transaction and wait for B available. i use uac_reg_send() to reINVITE B .The problem is when B available, it receive INVITE but A doesn't receive ACK therefore the call is incomplete I guess the reason is source IP in reINVITE is Kamailio's not A's IP, so ACK can't reach A. How can i set this source IP to A's IP ? Any idea is appreciated Thank you !

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RedHat 7 packages issue
by Alberto Panizzo 22 Jan '15

22 Jan '15

Hi all, Thanks for your hard working on Kamailio! I was just trying to install kamailio 4.2.X on a RedHat 7 machine but I got checksum mismatch: I took the yum configuration from: http://rpm.kamailio.org/ [kamailio] name=RPMs for Kamailio on RHEL 7 type=rpm-md baseurl=http://rpm.kamailio.org/stable/RHEL_7/ gpgcheck=1 gpgkey=http://rpm.kamailio.org/stable/RHEL_7/repodata/repomd.xml.key enabled=1 Than started yum update $ sudo yum update Loaded plugins: amazon-id, rhui-lb kamailio | 1.2 kB 00:00:00 kamailio/primary FAILED ] 0.0 B/s | 0 B --:--:-- ETA http://rpm.kamailio.org/stable/RHEL_7/repodata/primary.xml.gz: [Errno -1] Metadata file does not match checksum] 0.0 B/s | 0 B --:--:-- ETA Trying other mirror. kamailio/primary | 7.4 kB 00:00:00 http://rpm.kamailio.org/stable/RHEL_7/repodata/primary.xml.gz: [Errno -1] Metadata file does not match checksum Trying other mirror. Indeed http://rpm.kamailio.org/stable/RHEL_7/repodata/repomd.xml says: <data type="primary"> <checksum type="sha">4cc322a95d4a978e383903feaf8fbbe8ccc2ff9e</checksum> <open-checksum type="sha">9fd04e59185a8e844133b834a662cae3cc0837b7</open-checksum> <location href="repodata/primary.xml.gz"/> <timestamp>1413465311</timestamp> <size>7344</size> <open-size>70829</open-size> </data> while http://rpm.kamailio.org/stable/RHEL_7/repodata/primary.xml.gz I can download is: $ ls -l primary.xml.gz -rw-rw-r-- 1 alberto alberto 7562 gen 22 10:58 primary.xml.gz $ shasum primary.xml.gz e8e052970984feb0cdb16b98f05aefc516295aba primary.xml.gz Do I have to switch to other repository? Thank you, Best Regards -- Alberto Panizzo CTO - Co-Founder Amarula Solutions BV Cruquiuskade 47 Amsterdam 1018 AM NL T. +31(0)851119171 F. +31(0)204106211 www.amarulasolutions.com

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How to make presence module allocating an etag only for an initial Publish??
by Nuno Reis 21 Jan '15

21 Jan '15

Hello all. I've been testing presence module for a while and I do notice that presentity table grows endlessly over time. Each time a call is made a new record is added in presentity with a different etag. In documentation, namely developers guide we can read this: int etag_not_new; /* * 0 - the standard mechanism (allocating new etag for each Publish) * 1 - allocating an etag only for an initial Publish */ How can I tell the presence module in kamailio config to work using the second form of the above? Looking forward to hear from you. BR, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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Kamailio doesn't response to some INVITE packages
by Volkan Oransoy 21 Jan '15

21 Jan '15

Hi, My Kamailio box works very well for a year, in front of a Freeswitch box. Nowadays, some of my users live problems about receiving calls. But this problem occurs on a couple of specific users. %1 of users complain about it. When I capture the traffic, I see that, Kamailio receives an invite package from freeswitch but takes no action. Normally it should response with a 100 trying message. Here is the INVITE package. 1.2.3.6 is the Freeswitch box and 1.2.3.2 is the Kamailio box. I captured this package on Kamailio box. U 1.2.3.6:5060 -> 1.2.3.2:5060 INVITE sip:2000@test.example.net SIP/2.0. Via: SIP/2.0/UDP 1.2.3.6;rport;branch=z9hG4bKZU1Z76FvDZ26r. Route: <sip:1.2.3.2>;lr. Max-Forwards: 65. From: "908508850000" <sip:908508850000@test.example.net>;tag=KjcgDe0NyKN0B. To: <sip:2000@88.235.112.33:1028>. Call-ID: 72b344f4-1c09-1233-ad8d-0050568e9066. CSeq: 70585075 INVITE. Contact: <sip:mod_sofia@1.2.3.6:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit. 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. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 270. X-AUTH-IP: 159.253.40.138. X-BF-SRC: 2. X-FS-Support: update_display,send_info. Remote-Party-ID: "908508850000" <sip:908508850000@test.example.net >;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1421813895 1421813896 IN IP4 1.2.3.6. s=FreeSWITCH. c=IN IP4 1.2.3.6. t=0 0. m=audio 28384 RTP/AVP 3 8 0 101 13. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. I use standart location lookup and relay procedure in Kamailio default script. Is there anything wrong with this INVITE package and how can I trace this issue on Kamailio box? Thank you. /Volkan

3 3

FW: KAMAILIO IMS Questions
by Zaka Ul Isam 21 Jan '15

21 Jan '15

Hello Folks: Hope you can see this and share your ideas as to how to fix it? Thanks and regards, Zaka ________________________________________ From: Zaka Ul Isam Sent: Wednesday, January 21, 2015 12:20 PM To: Carsten Bock Cc: miconda(a)gmail.com Subject: RE: KAMAILIO IMS Questions hello All: First of all many thanks for your kind response. In fact my last few emails did not make it to the list (and nor bounced). It seems there is an error on the receiving mail server! However, I am regularly receiving list emails and can respond to threads but due to unknown reason, can not initiate a thread! Yes, it is an install from stable version source. Inorder to bouble check, we even compiled this module from source and copied to the kamailio modules folder replacing the actual module,but without luck! Please suggest the fix. Thanks and regards, Zaka ________________________________________ From: Carsten Bock [carsten(a)ng-voice.com] Sent: Wednesday, January 21, 2015 11:56 AM To: Zaka Ul Isam Cc: miconda(a)gmail.com Subject: Re: KAMAILIO IMS Questions Hi Zaka, please forward the request to the sr-users list, as neither me or Daniel provide direct support. How did you install Kamailio? Are you sure, that you have the SIP-Trace in (/usr/local/lib64/kamailio/modules/)? >From your log, i can see, that you did not install it from the Debian-Packages.... Thanks, Carsten 2015-01-21 12:17 GMT+02:00 Zaka Ul Isam <Zaka.Ul(a)albtelecom.al>: > Hi Gentlemen: > > Yet another stumbling block! > Now it is not setting module parameters or finding module <SIPTRACE> > The module is there, I have confirmed it. Below is the error log & relevant > code snippet: > > Error LOG > > kamailio -cf /usr/local/etc/kamailio/pcscf/kamailio.cfg > loading modules under config path: /usr/local/lib64/kamailio/modules/ > 0(22035) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module > matching <siptrace> found > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 366, column 42: Can't set > module parameter > 0(22035) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module > matching <siptrace> found > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 368, column 35: Can't set > module parameter > 0(22035) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module > matching <siptrace> found > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 369, column 44: Can't set > module parameter > 0(22035) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module > matching <siptrace> found > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 370, column 37: Can't set > module parameter > 0(22035) ERROR: <core> [modparam.c:166]: set_mod_param_regex(): No module > matching <siptrace> found > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 371, column 38: Can't set > module parameter > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 372, column 1: syntax error > 0(22035) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file > /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 372, column 1: > ERROR: bad config file (7 errors) > 0(22035) WARNING: <core> [ppcfg.c:219]: pp_ifdef_level_check(): different > number of preprocessor directives: N(#!IF[N]DEF) - N(#!ENDIF) = 1 > [root@IMS-dta1 pcscf]# locate modparam.c > bash: locate: command not found > > Source File > > } > > regfree(&preg); > pkg_free(reg); > if (!mod_found) { > LM_ERR("No module matching <%s> found\n", regex); > return -4; > } > return 0; > } > > > Please advise! > > Kindest regards, > > Zaka > > > > ________________________________________ > From: Zaka Ul Isam > Sent: Friday, January 09, 2015 4:07 PM > To: Carsten Bock > Subject: RE: KAMAILIO IMS Questions > > > Hi Carsten: > > Thanks a lot for your quick response and elaboration. > > As regards HSS, for testbed, we plan to use FOKUS HSS, Hope it will serve > the purpose. Please note that in addition to DIALOG_NG old DIALOG Module was > also required! > > > Thanks again & regards, > > Zaka > > ________________________________________ > From: Carsten Bock [carsten(a)ng-voice.com] > Sent: Friday, January 09, 2015 3:05 PM > To: Zaka Ul Isam > Subject: Re: KAMAILIO IMS Questions > > Hi Zaka, > > Thanks :-) > > You questions: > 1) you should be able to start the three components with "kamailio -f > /path/to/kamailio.cfg". > > You may need to modify the following line: > # ------------------ module loading ---------------------------------- > mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" > > This defines, where Kamailio will find it's modules. After the > source-code install, this is probably not correct :-) > > 2) The PCRF (and the pcscf.xml-Configuration) is optional (this > functionality has to be enabled by enabling "WITH_RX" in the > configuration, otherwise the pcscf.xml is ignored) > > 3) I assume, you also installed FHoSS from source instead from the > Package? In any case, you need to edit the hibernate.properties (e.g. > /opt/FHoSS/deploy/hibernate.properties). > > Regarding the errors of the Webinterface: It doesn't matter, if it's > 32 or 64 bit. FHoSS and it's Webinterface are for showcase and > scientific work only, so i assume, that's fine (as long as it works). > In production environments, we use e.g. a Nokia Siemens HSS (if > already present, e.g. for the LTE access) or our own drop in > replacement (with a totally different interface). > > Kind regards, > Carsten > > 2015-01-08 15:27 GMT+01:00 Zaka Ul Isam <Zaka.Ul(a)albtelecom.al>: >> Hi, Carsten: >> >> Thanks for the informative posts and responses on IMS related topics. I >> was >> trying to configure based on your last post at: >> >> http://www.kamailio.org/wiki/tutorials/ims/installation-howto >> >> With respect to above, I have few questions: >> >> 1: If the Kamailio based IMS, compiled from source (not package manager >> install) and all components in one machine. How do we start different >> components? >> >> kamailio -f /PATH/TO/RESPECTIVE COMPONENTS EXECUTABLE (kamailio.cfg with >> an >> include file P/I/S-CSCF.CFG) >> >> 2: In case of P-CSCF there are three config files as well (.xml for >> Diameter >> interface). The latter has reference to PCRF PEER >> Is it must? as there is no corresponding DNS entry in >> Kamailio-IMS.DNSZONE file. >> 3: As regards FOKUS HSS, in order to adapt the config to our scenario, do >> we >> also need to edit FHOSS/hibernate.properties file? Besides, we have errors >> with HSS WEB interface on 64 bit machines. Is 32 bit architecture the >> recommended one for HSS? >> >> Thanking you in anticipation, >> >> 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. > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Office +49 40 5247593-0 > Fax +49 40 5247593-99 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > ________________________________ > > 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. -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-0 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ ________________________________ 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.

1 0

Kamailio doesn't response to some INVITE packages
by Volkan Oransoy 21 Jan '15

21 Jan '15

My Kamailio box works very well for a year, in front of a Freeswitch box. Nowadays, some of my users live problems about receiving calls. But this problem occurs on a couple of specific users. %1 of users complain about it. When I capture the traffic, I see that, Kamailio receives an invite package from freeswitch but takes no action. Normally it should response with a 100 trying message. Here is the INVITE package. 1.2.3.6 is the Freeswitch box and 1.2.3.2 is the Kamailio box. I captured this package on Kamailio box. U 1.2.3.6:5060 -> 1.2.3.2:5060 INVITE sip:2000@test.example.net SIP/2.0. Via: SIP/2.0/UDP 1.2.3.6;rport;branch=z9hG4bKZU1Z76FvDZ26r. Route: <sip:1.2.3.2>;lr. Max-Forwards: 65. From: "908508850000" <sip:908508850000@test.example.net>;tag=KjcgDe0NyKN0B. To: <sip:2000@88.235.112.33:1028>. Call-ID: 72b344f4-1c09-1233-ad8d-0050568e9066. CSeq: 70585075 INVITE. Contact: <sip:mod_sofia@1.2.3.6:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit. 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. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 270. X-AUTH-IP: 159.253.40.138. X-BF-SRC: 2. X-FS-Support: update_display,send_info. Remote-Party-ID: "908508850000" <sip:908508850000@test.example.net >;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1421813895 1421813896 IN IP4 1.2.3.6. s=FreeSWITCH. c=IN IP4 1.2.3.6. t=0 0. m=audio 28384 RTP/AVP 3 8 0 101 13. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. I use standart location lookup and relay procedure in Kamailio default script. Is there anything wrong with this INVITE package and how can I trace this issue on Kamailio box? Thank you.

1 0

Removing unwanted characters from From-header
by Sebastian Thörn 21 Jan '15

21 Jan '15

Hi, Hello! I'm having trouble with matching massages for some Patton equipment, from the looks of it, Patton don't like </> in headers. The initial INVITE from Patton is sent without </> in To and From-headers. In the logs from Patton I can see that it matches the "100 trying" that Kamailio sends back to it, and this message has To and From-headers without </>, but the "180 Ringing" and "200 OK" that Kamailio sends is not matched, and these two has </>. I would like to remove < and > in my onreply_route, how can I do this? Or does anyone have experience on how Patton matches messages? Br Sebastian Thörn

1 0

need some help please
by mail＠sathees.co.uk 20 Jan '15

20 Jan '15

Hi need some help please, kamailio v-4.2.2 asterisk 13.1 ;#### sip.conf [general] context=incoming calls allowguest=yes bindport=5080 bindaddr=192.168.1.77 rtcachefriends=yes [kamailio] type=friend host=192.168.1.77 port=5060 context=kamailio-trunk canreinvite=no insecure=port,invite dtmfmode=rfc2833 nat=force_rport,comedia qualify=yes i have followed this tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb My DDI provider sending call to ip address via DDNS. DDI à DDNS à kamailio & asterisk in one server [192.168.1.77]. kamailio passing DDI to asterisk, but its sending to wrong context. If any of the sip users is registered, the calls were sending to registered user context. If no sip user is online then calls has been send to kamailio-trunk. How can I get all DDI to kamailio-trunk? How can I solve this problem? Many thanks Sathees

3 2

OMA/RCS Presence compliance
by Osbun, Alex 20 Jan '15

20 Jan '15

Hello, Are the Kamailio presence modules (presence, xcap, pua, rls, etc.) RCS 5.X and OMA Presence Simple spec compliant. I found the following bullet from this announcement - http://www.kamailio.org/w/2012/09/kamailio-at-11-years/ * SIMPLE Presence & IM extensions – embedded XCAP server, MSRP relay, RLS full OMA/RCS extensions implementation, IMS registrar records replication, a.s.o. Does that mean that only the RLS functionality is fully OMA/RCS compliant? Thanks and I appreciate any guidance. Regards, Alex

2 1

dialog module shared memory
by Grant Bagdasarian 20 Jan '15

20 Jan '15

Hello, I'm using the dialog module to control the maximum number of calls allowed to be passed through. Is it possible to share the dialogs stored in memory with multiple instances of kamailio? So if both kamailio instances process calls, and the maximum is set to a global of 10, both combined should never let more than 10 calls through. Regards, Grant

4 8

PCSCF Installation got issues
by Sravana R 20 Jan '15

20 Jan '15

Hi, Downloaded Kamailio-4.2.1.tar.gz file from Kamailio.org site and compiled it on RHEL 6.2 server. Configured as per setup instruction given in attached document. While start pcscf, getting the following errors: Jan 20 01:25:17 localhost pcscf: INFO: <core> [ppcfg.c:80]: pp_subst_add(): ### added subst expression: /PORT/4060/ Jan 20 01:25:17 localhost pcscf: INFO: <core> [ppcfg.c:80]: pp_subst_add(): ### added subst expression: /NETWORKNAME/open-ims.test/ Jan 20 01:25:17 localhost pcscf: INFO: <core> [ppcfg.c:80]: pp_subst_add(): ### added subst expression: /HOSTNAME/pcscf.open-ims.test/ Jan 20 01:25:17 localhost ./pcscf[4201]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Jan 20 01:25:17 localhost ./pcscf[4201]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available Jan 20 01:25:17 localhost ./pcscf[4201]: INFO: ims_usrloc_pcscf [hslot.c:62]: ul_init_locks(): locks array size 512 Jan 20 01:25:17 localhost ./pcscf[4201]: INFO: path [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Jan 20 01:25:17 localhost ./pcscf[4201]: INFO: path [path_mod.c:147]: mod_init(): outbound module not available Jan 20 01:25:17 localhost ./pcscf[4201]: ERROR: registrar [reg_mod.c:352]: mod_init(): can't bind usrloc Jan 20 01:25:17 localhost ./pcscf[4201]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module registrar (/usr/local/lib64/kamailio/modules/registrar.so) Please help us to resolve above issue. Thanks, Sravan. ============================================================================================================================ Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at http://www.techmahindra.com/Disclaimer.html externally http://tim.techmahindra.com/tim/disclaimer.html internally within TechMahindra. ============================================================================================================================

1 0

REGISTER request and save() from onreply_route
by Dmitry Sytchev 19 Jan '15

19 Jan '15

Hi all! Seems to be an old topic but still unclean is it possible or not, please correct me if I'm wrong. There are some old threads like http://lists.sip-router.org/pipermail/users/2007-May/011278.html I use Kamailio 4.2.1 trying to create intercepting registrar proxy, which handles NAT (rewrites contact), forwards REGISTER to backend and then stores REGISTER in its internal location on 200 OK from backend, so I call save() from reply route in noreply mode. The problem is, it stores IP source of reply as 'received' and modified contact which i sent out to backend server as 'contact', so it loses original information from initial REGISTER. Is there a way to make save() use original request fields like T_req($ct), or some way to call save() with full set of custom params? Of course, I can save() all REGISTERs in request_route and then delete on backend fail, but it seems to be waste of resources, especially when REGISTER traffic is high :( I'd be glad to hear any advice.

2 3

REGISTER request and save() from onreply_route
by Dmitry Sytchev 19 Jan '15

19 Jan '15

Hi all! Seems to be an old topic but still unclean is it possible or not, please correct me if I'm wrong. There are some old threads like http://lists.sip-router.org/pipermail/users/2007-May/011278.html I use Kamailio 4.2.1 trying to create intercepting registrar proxy, which handles NAT (rewrites contact), forwards REGISTER to backend and then stores REGISTER in its internal location on 200 OK from backend, so I call save() from reply route in noreply mode. The problem is, it stores IP source of reply as 'received' and modified contact which i sent out to backend server as 'contact', so it loses original information from initial REGISTER. Is there a way to make save() use original request fields like T_req($ct), or some way to call save() with full set of custom params? Of course, I can save() all REGISTERs in request_route and then delete on backend fail, but it seems to be waste of resources, especially when REGISTER traffic is high :( I'd be glad to hear any advice. -- Best regards, Dmitry Sytchev, IT Engineer

1 0

Diameter problem on Kamailio
by Kowalski Piotr 3 - Hurt 19 Jan '15

19 Jan '15

Dear sr-users! We are facing serious problem. We built openIMS based on Kamailio. Moreover, we have RCS users registered, but some users are automatically unregistered on the openIMS core: 1) We don't know what a piece of log on S-CSCF means: /usr/sbin/kamailio[17392]: ERROR: --- EXPLOIT --- We need to do an UNREG server SAR assignemnt 2) In log files there is "404 Not Found: Unknown destination" or "404 user offline". Probably there is some diameter problem. Has anyone of you already faced such problem? Many thanks for your answers. Piotr Kowalski Starszy Architekt Techniczny Tel. +48 22 699 50 93 Kom. +48 519 123 289 Orange Orange Polska Hurt Badania i Rozwój Sieci i Platform Usługowych Wydział Usług Konwergentnych Obrzeżna 7 /p. 309, 02-691 Warszawa Fax. +48 22 857 99 86 www.hurt-orange.pl<http://www.hurt-orange.pl/> - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Czy musisz drukować tę wiadomość? Pomyśl o środowisku. Korzystaj z e-faktury. Treść tej wiadomości jest własnością Orange Polska i zawiera informacje stanowiące tajemnicę przedsiębiorstwa Orange Polska. Jeżeli nie jesteście Państwo jej adresatem, bądź otrzymaliście ją przez pomyłkę, prosimy o powiadomienie o tym nadawcy oraz trwałe jej usunięcie. Orange Polska Spółka Akcyjna z siedzibą i adresem w Warszawie (02-326) przy Al. Jerozolimskich 160, wpisana do Rejestru Przedsiębiorców prowadzonego przez Sąd Rejonowy dla m.st. Warszawy XII Wydział Gospodarczy Krajowego Rejestru Sądowego pod numerem 0000010681; REGON 012100784, NIP 526-02-50-995; z pokrytym w całości kapitałem zakładowym wynoszącym 3.937.072.437 złotych.

2 1

Search Un-Register Expires header
by Emilio Soler Muñoz 19 Jan '15

19 Jan '15

Hi, I find a problem when I try to search the header "Expires" with value 0. At my kamailio.cfg the configuration is: if ( search_hf("Expires", ":0@", "a") ) { xlog("L_INFO", "Encontrado expires 0 ($rm)"); } This way find nothing. I test with 0 (no wildcards regular expression), but this make matches in any Register with header Expires content 0. Example: ... if ( search_hf("Expires", "0", "a") ) .... I saw some people control expires=0 at contact header, but not my case. Example: if (search("expires=0")) { xlog("L_INFO", "Encontrado expires 0 ($rm)"); } Regards, -- Emilio Soler emiliosoler(a)gmail.com

3 6

RTPengine
by Kalala Alexander 19 Jan '15

19 Jan '15

4 11

TCP connectivity issue
by Mickael Marrache 19 Jan '15

19 Jan '15

Hi, I use Kamailio as a stateless proxy with UDP + TCP. When TCP is used, I get the following TCP errors: WARNING: <core> [receive.c:265]: receive_msg(): error while trying onreply script ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect xxxxxxxxx failed (timeout) WARNING: <core> [receive.c:265]: receive_msg(): error while trying onreply script ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect xxxxxxxxxx failed (timeout) The TCP config in my script: tcp_connection_lifetime=3605 I get this issue when I receive replies to INVITE from the UAS. I can see the replies using ngrep but Kamailio doesn't seem to be able to process them correctly. I precise that the UAS is behind NAT and NAT configuration is correct (the INVITE does reach the UAS). I also precise that the UAS sends keepalive every 5 sec (I can see the ping-pong using ngrep). I set it to 5 sec to eliminate the possibility that the NAT closes the connection. Thanks, Mickael

1 0

call control for 4.2?
by Peter Villeneuve 18 Jan '15

18 Jan '15

Hi, I want to experiment with the following setup and was wondering which module(s) you all recommend. I have a working simple kamailio 4.2 setup where registered users can only call each other on the same domain (ie no outbound calling to PSTN or other domains). Now I want to explore the possibility of limiting call times for certain users but allowing unlimited call times for others (ie user 5000's calls are limited to 2 minutes while user 6000 has unlimited calling time). There are some tutorials available on the internet but they seem to be quite outdated. Given my simple scenario, should I be looking into using the callcontrol module and install the CDR tool from here http://callcontrol.ag-projects.com/ (I don't really need any complex billing scheme with various rates but rather a simple scheme were user x is either time limited or not). Before I dig in any further, I'd appreciate it of someone could tell me if this is the way to go or is there's a simpler solution. Are there any recent tutorials that I should be looking at? Cheers, Peter

3 4

UID-* modules
by Olle E. Johansson 18 Jan '15

18 Jan '15

hi! Frankly, I don't really know the difference between the uid* modules and the "regular" modules. I know the history, but would like to understand and maybe some text I can add to the README files of these modules to explain to other users what the difference is. /O

1 0

Load balancer - Path and Route header issue
by Mickael Marrache 18 Jan '15

18 Jan '15

Hi, I'm trying to use one Kamailio instance as load balancer for multiple Kamailio instances. The load balancer only performs NAT handling and dispatching of messages. The other instances handle registrations, routing, RTP proxy, authentication, etc. When a REGISTER request is received by the load balancer, a Path header is added to the request which is then forwarded to one of the registrars. The registration request is successfully handled and the Path header is saved in database. All subscribers register through the load balancer. When an INVITE is received from one subscriber by the load balancer, a Record-Route header is added by the load balancer which then forwards the INVITE to one of the other instances. The selected instance authenticates the subscriber, looks for the callee using the REGISTRAR module, adds a Record-Route header, adds a Route header according to the Path header associated with the callee, and forwards the INVITE to the load balancer. The load balancer performs loose routing (removing the Route header), adds another Record-Route header, and forwards the INVITE to the callee using the right transport address (indicated by the received parameter present in the Route header). The flow is right for the INVITE and its replies. However, when the caller sends the ACK, the route set is: load balancer, proxy, load balancer - without any information about how to reach the callee. Therefore, when the ACK reaches the "second" load balancer, the latter forwards it according to the RURI which is the Contact URI present in the 200 OK (which is not routable because of the NAT). I expected Kamailio to reuse the information that was present in the Path header to route the ACK correctly. What am I doing wrong? Thanks, Mickael

1 0

Kamailio with rtpproxy and asterisk in internal lan
by Willian Mazzardo - SYSSVOIP 17 Jan '15

17 Jan '15

Hi all. I want to create a Kamailio server who have dispatcher module, to loadbalance some asterisk servers who have my billing system. But, if I call from internet to my Public IP on Kamailio server, and then call goes to internal IP asterisk, who execute Echo App, I cant hear myself. If I call to my Public IP on Kamailio server, and call goes to EXTERNAL IP on Asterisk, everything works nice. I want this : Sip Client > Public IP Kamailio > Internal IP Kamailio > Internal IP Asterisk Server > Echo() I have rtpproxy running on same machine that Kamailio, and Im running with this command: rtpproxy -F -s udp:127.0.0.1:7722 -l PU.BL.IC.IP/172.16.1.1 -d DBUG:LOG_LOCAL0 My asterisk have 172.16.1.2 as internal IP. https://www.cloudshark.org/captures/15724d2d7e51 This is my tcpdump file captured. Thanks for any help. Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030

1 0

Using the iptel Switch for both home & office environments...
by H. KAIMARAH 15 Jan '15

15 Jan '15

Hello there, Kindly advise me on how i can download and install the iptel soft switch for both Windows and Linux platforms. I intend to install a hosted Sip VoiP service using your open source switch for both home and office environments. I have jst been checking out the documentation from the site but seams some documentation pages are failing to load (timing out). Kindly advise or share with me some of the download links for the application so i can make a successful installation. Sincere regards, -- ***/Happy New Year/*** *HENRY K. Kaimarah* /Cell:/ +256-754462010 /Now in Kampala/ - UGANDA.

2 2

REGISTER replication using t_replicate with ds_select_dst
by Spencer Thomason 15 Jan '15

15 Jan '15

Hello all, We are moving our registration and location services to dedicated instances geographically separate from our proxies and then using redirects to locate the registered user. The idea is to protect the proxies from a potential thundering herd of registrations if there was some network outage I’m using dispatcher to find a registrar that has a registration for the user, if a user is not locally registered a 503 is returned and the next server is contacted. Registrations are replicated to other servers when the client is not behind NAT. If the client is behind NAT, the registrar where the client initially registered performs keep-alives and NAT traversal and also acts as an outbound proxy instead of sending a redirect. I’m attempting to simplify the configuration by using a dispatcher group to classify incoming registrations and to select replication targets but the registrar that initially received the request is in the group as well. Is there an elegant way to chose all destinations except the local instance? I’m currently doing this with the hack below but any suggestions would be greatly appreciated. Also, if anyone has any comments or recommendations on the topology, please share! Thanks, Spencer if (isflagset(FLT_FROMREG)) return; subst('/^From:(.*)sip:.*@[a-zA-Z0-9.:]+(.*)$/From:\1sip:$au@DOMAIN\2/ig'); subst('/^To:(.*)sip:.*@[a-zA-Z0-9.:]+(.*)$/To:\1sip:$au@DOMAIN\2/ig’); $rd = "DOMAIN"; $var(local_uri) = "sip:" + $Ri + ":" + $Rp; ds_select_dst("10", "4"); if ($du != $var(local_uri)) { t_replicate($du); } while (ds_next_dst()) { if ($du != $var(local_uri)) { t_replicate($du); } }

1 0

Changing SDP with subst after rtpproxy_answer corrupts SDP
by Matthias van der Vlies 15 Jan '15

15 Jan '15

Dear all, I have an on-reply route that needs to change the SDP version for the reply coming in. The use case is that I have a mobile originated call and there is some Ericsson switch that doesn't like it when the SDP version is updated (in this case by asterisk) although nothing has changed to the actual SDP (183 session progress and then OK.) Funny thing is that Asterisk will actually drop a call if it receives a re-INVITE with same version... That's why they invented ignoresdpversion, but now it's the other way around.... :) Mobile phone -> Ericsson MSC -> ACME packet -> (18X.4X.XXX.XX) Kamailio (10.41.0.21) -> Asterisk The issue is that the asterisk sends a reply 200 OK, with an updated version because it already sent SDP for 183 session progress. This can be patched in asterisk, but in my scenario I can unfortunately not do that. Thus trying to fix this on Kamailio. I am able to 'fix' this currently by performing a subst on the sdp owner variable: onreply_route[WITHSDP] { if (has_body("application/sdp")) { if(ds_is_from_list()) { rtpproxy_answer("wrei"); *if(subst("/^o=someowner ([0-9]+) ([0-9]+) IN IP4 (.*)$/o=someowner \1 \1 IN IP4 \3/")) {* xlog("L_INFO", "Fixed Asterisk incorrect version number in SDP"); } # tried the answer here as well, but that corrupts it even more } else { rtpproxy_answer("wrie"); } } exit; } However this corrupts the SDP: v=0 o=tismi 652858233 652858233 IN IP4 10.41.0.21*18X.4X.XXX.XX* s=Some server c=IN IP4 18X.4X.XXX.XX t=0 0 m=audio 57644 RTP/AVP 8 0 18 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes When I do not substitute the SDP looks perfectly fine and the external address shows as the IN IP4. But of course the version is incremented: v=0 o=tismi 1606876535 *1606876536* IN IP4 *18X.4X.XXX.XX* s=Some server c=IN IP4 18X.4X.XXX.XX t=0 0 m=audio 55410 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=sdpmangled:yes The ericsson is now accepting this (although it's corrupt, I know.... probably the ACME doing something funky with it), but it causes issues with another unknown piece of equipment that fails on parsing the session owner. I hope there is something wrong with my subst, but I'm afraid I can not do this from the on_reply route because SDP is only updated once it finishes? I know it's dangerous to alter the session version like this, so I made sure the Asterisk will never send a re-INVITE. Now I need a way to not corrupt the o= Kind regards, Matthias van der Vlies

3 3

websocket convert and forward to asterisk
by jaflong jaflong 15 Jan '15

15 Jan '15

Hi Everyone, Does anyone have a example of the config where I can get the following to work I want Kamailio to process websocket converting wss to tcp and srtp to rtp and forward to asterisk as tcp and rtp incoming call on websocket (wss rtp/savpf) --> kamailio/rtpproxy (protocol - convert wss to tcp) and (sdp - convert rtp/savpf to rtp/avp) and (audio - convert srtp to rtp) --> asterisk (recieve protocol: tcp, sdp: rtp/avp and audio:rtp) This application would only have incoming call from websocket to forward on to asterisk.

1 0

Fosdem 2015
by Alexandr Dubovikov 15 Jan '15

15 Jan '15

Hello everybody, what is about Fosdem this year ? Who is going there ? Wbr, Alexandr

2 1

Kamailio with a third party operator
by Jyaim Jyaim 15 Jan '15

15 Jan '15

Hello all, I cannot find documentation on how to configure Kamailio to interwork with another operator's IMS Core. I think the s-cscf should be in charge of detecting a user is not known locally, in order to route requests toward other operator. Do you know where I should look to get infos on this topic ? Thank you!

1 0

Redirect Server Including Path/Recieved Information
by Asgaroth 15 Jan '15

15 Jan '15

Hi All, I am attempting to setup a standalone redirect server which will lookup contact info and redirect to appropriate outbound proxy. The problem I am having is that the registrar is storing the path and recieved information, however, when I perform a lookup and reply with 302, the recieved and path information is not included. Are there module parameters in the registrar module that will include these contact parameters (recieved) or create the appropriate headers (path/route) in the 302 response, or, is this something I need to do manually. Currently I have the following lookup code: route { t_check_trans(); if ( method == "INVITE" ) { xlog("route[MAIN] : $rm : ruri=$ru"); xlog("route[MAIN] : $rm : lookup=sip:$rU@registered.domain"); if ( lookup("location", "sip:$rU@registered.domain") ) { send_reply("302", "Moved Temporarily"); exit; } } } I had a look at the path_mode parameter but this looks like it only takes affect for REGISTER methods. The INVITE that comes in supports the PATH header but I dont see this being passed back. In fact, I'm not entirely sure if the path header is a supported header in the 302 response message. I had a quick google and I cant seem to easily find what headers are suported in the 302 message, and where I need to put the path/recieved information. I presume I can add the recieved info as a parameter to the contact header, but how would i specify the outbound proxy to use, would that be in a route or a path header in the 302 response? I can see that the $du pseudo variable is set with the appropriate outbound path and received information, now I just need to include this information in the 302 response. Any suggestions/comments to assist in how I get this info into the 302 message would be greatly appreciated. Thanks in advance.

4 12

Crash in usrloc, get_all_db_ucontacts
by Kristian F. Høgh 14 Jan '15

14 Jan '15

Hi, On an ancient kamailio 3.1, we occasionally experience crash in the function get_all_db_ucontacts in usrloc. A backtrace reveals the following: #0 0x00002b505f2fb4ff in get_all_db_ucontacts (buf=0xadd270, len=<value optimized out>, flags=64, part_idx=1, part_max=30) at dlist.c:176 176 if (VAL_NULL(ROW_VALUES(row)+4) || p1==0 || p1[0]==0){ The lines around line 176 (which look similar to trunk, except p and p1 are str today) 159 for(i = 0; i < RES_ROW_N(res); i++) { 160 row = RES_ROWS(res) + i; 161 162 /* received */ 163 p = (char*)VAL_STRING(ROW_VALUES(row)); 164 if ( VAL_NULL(ROW_VALUES(row)) || p==0 || p[0]==0 ) { 165 /* contact */ 166 p = (char*)VAL_STRING(ROW_VALUES(row)+1); 167 if (VAL_NULL(ROW_VALUES(row)+1) || p==0 || p[0]==0) { 168 LM_ERR("empty contact -> skipping\n"); 169 continue; 170 } 171 } 172 p_len = strlen(p); 173 174 /* path */ 175 p1 = (char*)VAL_STRING(ROW_VALUES(row)+4); 176 if (VAL_NULL(ROW_VALUES(row)+4) || p1==0 || p1[0]==0){ 177 p1 = NULL; 178 p1_len = 0; 179 } else { 180 p1_len = strlen(p1); 181 } The client from frame #2 (bt full below), send a REGISTER, which was replied "200 OK" at the same second as the crash. Is it a timing issue between REGISTER and nathelper ping? I have a trace and the core. Regards, Kristian Høgh Uni-tel bt full #0 0x00002b505f2fb4ff in get_all_db_ucontacts (buf=0xadd270, len=<value optimized out>, flags=64, part_idx=1, part_max=30) at dlist.c:176 sock = 0x0 dbflags = 11088 host = {s = 0x375ef8e "PROXY_IP:5060", len = 13} i = <value optimized out> row = 0xa13488 dom = 0x2b5062723278 p1 = 0x2 <Address 0x2 out of bounds> now_s = "'2015-01-13 12:58:51'\177\000\000)" port = 11088 p_len = 73 cp = <value optimized out> needed = 16 res = 0xa2ed38 p = 0xcfcdffaf <Address 0xcfcdffaf out of bounds> now_len = 21 proto = 1642847400 p1_len = 0 shortage = 0 query_buf = "select received, contact, socket, cflags, path from location where expires > '2015-01-13 12:58:51' and cflags & 64 = 64 and id % 30 = 1", '\000' <repeats 376 times> query_str = {s = 0x2b505f515520 "select received, contact, socket, cflags, path from location where expires > '2015-01-13 12:58:51' and cflags & 64 = 64 and id % 30 = 1", len = 135} #1 get_all_ucontacts (buf=0xadd270, len=<value optimized out>, flags=64, part_idx=1, part_max=30) at dlist.c:398 No locals. #2 0x00002b5061eb6b1a in nh_timer (ticks=<value optimized out>, timer_idx=<value optimized out>) at nathelper.c:1623 rval = <value optimized out> buf = 0xadd270 cp = <value optimized out> c = {s = 0x0, len = 0} opt = {s = 0x0, len = 0} path = {s = 0x0, len = 0} curi = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0xadd836 "CLIENT_IP:62252\250u\220", len = 14}, port = {s = 0xadd845 "62252\250u\220", len = 5}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = { s = 0x7fff110134b0 "", len = 566862356}, port_no = 50336, proto = 70, type = ERROR_URI_T, flags = 65536, transport = {s = 0x8cbfa0 "", len = 5441948}, ttl = {s = 0x7fff1101352f "", len = 4448297}, user_param = {s = 0x375ef8e "PROXY_IP:5060", len = 13}, maddr = {s = 0x9204b0 "sip3", len = 10677560}, method = {s = 0x0, len = 0}, lr = {s = 0xadd84a "\250u\220", len = 0}, r2 = {s = 0xa2ebc0 "\250\272\252", len = 285291296}, transport_val = {s = 0x40 <Address 0x40 out of bounds>, len = 335544320}, ttl_val = { s = 0x3221c302d0 "H\307\300\017", len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}} he = <value optimized out> send_sock = <value optimized out> flags = <value optimized out> proto = <value optimized out> dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {in6_u = {u6_addr8 = '\000' <repeats 15 times>, u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, id = 0, proto = 0 '\000', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}} iteration = 1 __FUNCTION__ = "nh_timer" #3 0x0000000000521c54 in fork_dummy_timer (child_id=<value optimized out>, desc=<value optimized out>, make_sock=<value optimized out>, f=0x2b5061eb6a70 <nh_timer>, param=0x0, interval=1) at timer_proc.c:71 pid = <value optimized out> #4 0x00002b5061eb42ee in child_init (rank=<value optimized out>) at nathelper.c:673 i = 58055190 #5 0x00000000004db1d0 in init_mod_child (m=0x917dc0, rank=0) at sr_module.c:829 No locals. #6 0x00000000004db1a9 in init_mod_child (m=0x918420, rank=0) at sr_module.c:826 No locals. #7 0x00000000004db1a9 in init_mod_child (m=0x918aa0, rank=0) at sr_module.c:826 No locals.

2 10

One way audio & video - Kamailio + rtpproxy ( NAT call )
by Chirag Ajmera 14 Jan '15

14 Jan '15

Kamailio - 4.2.2 ( SIP server ) Rtpproxy - Git Compiled ( miconda patched version ) Issue: Remote NAT Call Bria Rmt Iphone SIP Extn (3G) ----> Kamailio Server -----> Desktop Bria Client ( Wifi ) Audio and Video packets are sent from iPhone to desktop client .. but nothing otherway Also, being totally new to kamailio I can't figure out logging SIP messages ( INVITE etc ) .. Kindly if anyone can help Thank You Best, Chirag A.

1 0

Fwd: Need a support for Siremis and kamailio server
by Gaurav Rai 14 Jan '15

14 Jan '15

Hi I am using a siremis and *kamailio * hosted server with lylix.net and they give a link for getting a support about Siremis. I need a support to how to make OUT call on PSTN with registered subscriber on Softphone. Currently we have registered user on subscriber table with their location and can make a PEER to PEER audio call and text messaging also. To make OUT call on PSTN we had create a LCR gateway and LCR rule with Prefix but still can not make a OUT call getting a error 404 can u please help me out to make a PSTN call Waiting for your reply Thanks Gaurav Rai +919716238804

2 1

Making rtp streams appear unidirectional
by aft 14 Jan '15

14 Jan '15

Hi, Is it possible to make the rtp stream appear unidirectional? By that i mean, The rtp stream from client to proxy will go through one rtpproxy and proxy to client stream will go through another rtpproxy instance? If not, is it possible to mimic something like that by running rtpproxy in bridge mode where both IPs in bridge mode will be public IP? Thanks in advance cheers arif

2 4

Dispatcher 8 algorithm based on priority strange works
by Yuriy Gorlichenko 14 Jan '15

14 Jan '15

Daniel. I added 8 algorithm to our server and it works with 2 asterisk now but it works strange because: While works server with priority 1 - all ok. When this server goes down dispatcher choose next server with lowes priority. But when server with highest priority waking up dispatcher use server with lowes priority until this server not goes down. here id my config for dispatcher modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:kamailio1@10.0.1.12") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("tm", "reparse_on_dns_failover", 0) (!ds_select_dst("$var(setid)", "8")){ sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","{$rm} from [$fU@$si:$sp] But NO destinations available for $rd \n"); t_on_failure("DISPATCHER_ROLLOVER"); } if (!t_relay()) { sl_reply_error(); we use 4.3 master branch kamailio

2 3

Unexpected evaluation to false on a phonenumber in $avp
by Daniel Tryba 14 Jan '15

14 Jan '15

Not a really important question, just an observation that caused some headscratching on my first day after a nice vacation. I'm using an avp to make a redirect decision. Normally this avp contains Dutch telephone numbers (+31[0-9]{7,10}). With a simple: if($avp(dst_redirectnumber)) { #change $ru } this works flawless (normally). But a client decided, to forward to a Swiss number (+4141743xxxx) but with this number the if condition results in a false. A German number just works fine (+49) The avp is stuffed in an avp_db_query loop with $avp($avp(i:1))=$avp(i:2) as key/value pair from the table. I changed above condition to check if the string length of dst_redirectnumber is larger than 0. But I'm at a loss why the false occurs, first guess was an octal conversion (number only contains the digits 0-7) which results in a number "just" above 2^32. But changing a digit to 8 still fails. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

4 5

Dispatcher Failover algorithm
by Yuriy Gorlichenko 14 Jan '15

14 Jan '15

I as wrote before - we find dispatcher algorithm than can do mechanism something like this: Try call to fist server with max priority or weight. OIf this server unavailible then call second server with less weight and etc. Does anyone know what ling of algorithm we can use for this?

2 6

event on table location Duplicate entry for key 'ruid_idx'
by Yuriy Gorlichenko 14 Jan '15

14 Jan '15

Hello. We use 2 kamailio servers cluster and we have porblems with db. Database failed pecause of error: Could not execute Write_rows_v1 event on table production.location; Duplicate entry 'uloc-54aae947-86d-a67' for key 'ruid_idx', Error_code: 1062; handler error HA_ERR_FOUND_DUPP_KEY; the event's master log FIRST, end_log_pos 380, Internal MariaDB error code: 1062 But a location table no row 'ruid_idx' and no entry uloc-54aae947-86d-a67.

3 3

Kamailio with Asterisk 13 PjSip Realtime
by Chirag Desai 14 Jan '15

14 Jan '15

Hi, I saw Matt Jordan's recent Kamailio world talk and was interested in the idea he proposed of stripling out authentication and registration from asterisk and solely letting Kamailio handle it. In order to do this would I be correct in assuming I would have to use the asterisk database rather than the Kamailio database? I've compared the two and the table structures are very different. If I use the asterisk database I guess asterisk still needs to be responsible for handling authentication, registration and writing the contacts to the database. If I use the Kamailio database how would I dial an extension from asterisk, because as far as I can fell asterisk will have no idea who is registered or how to find them (contact details). Maybe I'm over thinking something simple, or maybe I'm not. Either way I would love your thoughts on how this could be done. Kind regards, C

1 0

pua_subscribe and force_send_socket trouble
by Mikko Lehto 13 Jan '15

13 Jan '15

Hi I am having strange behavior on local port assignment when proxying out locally generated request. Kamailio is 4.2.1 from git. I use MI pua_subscribe to create subscription to be handled by external presence server. In event_route[tm:local-request] I set $du to point back to proxy itself. When request hits initial request_route, I add some headers, change request URI and prepare failure_route for later uac_auth() processing. So far so good and TLS packets are flying to correct destination. Here is the trouble: when I observe outgoing TLS traffic with tcpdump, I can see that local source port is not following what I set with force_send_socket(). I tried to place force_send_socket() in request_route, branch_route, and tm:local-request, but it is always some random high port (>30000), never the intended one. I am trying to set it to same as my TLS listening socket. Is my usage somehow incorrect? What should I try next to make Kamailio use constant source port? Relevant config snippet below (IP address and domain part is obscured). --- mhomed=1 listen=udp:A.B.C.D:5041 listen=tls:A.B.C.D:5041 request_route { route(REQINIT); if ( blahblah... ) { $ru = "sip:" + $rU + "@domain.part.invalid;transport=tls"; $avp(uac_auth) = 0; route(PR_HDRS); route(PR_TRIGGERS); force_send_socket(A.B.C.D:5041); t_newtran(); route(RELAY); } } route[PR_HDRS] { remove_hf("User-Agent"); insert_hf("User-Agent: pua_subscribe\r\n","Call-ID"); } route[PR_TRIGGERS] { t_on_branch("PR_BRANCH"); t_on_failure("PR_FAILURE"); } branch_route[PR_BRANCH] { force_send_socket(A.B.C.D:5041); } failure_route[PR_FAILURE] { if ( $avp(uac_auth) == 0 && ($T_reply_code == 401 or $T_reply_code == 407) ) { uac_auth(); $avp(uac_auth) = 1; route(PR_TRIGGERS); route(RELAY); } } event_route [tm:local-request] { force_send_socket(A.B.C.D:5041); $du = "sip:A.B.C.D:5041"; } route[RELAY] { is from default config } --- I found these recent commits, bugs and threads somehow relating to force_send_socket and tm:local-request: http://sip-router.org/tracker/index.php?do=details&task_id=462 http://lists.sip-router.org/pipermail/sr-users/2014-August/084459.html dbd8ea9b1fa216e59d4c36e2eb4b671202824259 http://lists.sip-router.org/pipermail/sr-dev/2014-September/024984.html e404d123610b63ddd1c75d39667b373c40071eab http://lists.sip-router.org/pipermail/sr-dev/2014-September/024977.html -- Mikko Lehto

3 5

Kamailio 4.2.1 crashes on using exec_msg
by Tobias 13 Jan '15

13 Jan '15

Hi! We recently upgraded our Kamailio 4.1 to 4.2.1. With the newer version Kamailio crashes after just running a few minutes. After some debugging it looks as though the problem is in exec_msg (which is used in our config). After disabling this 4.2.1 seem to run just fine. Core file exists, here's the output: (gdb) backtrace #0 0x00000000005ebf0f in fm_extract_free (frag=0x7f053ea08d18, qm=0x7f053e88e010) at mem/f_malloc.c:206 #1 fm_malloc (qm=0x7f053e88e010, size=<optimized out>, file=file@entry=0x7f053cbedfd4 "exec: exec_hf.c", func=func@entry=0x7f053cbef378 "replace_env", line=line@entry=375) at mem/f_malloc.c:490 #2 0x00007f053cbe7953 in replace_env (list=0x7f053ea08868) at exec_hf.c:375 #3 0x00007f053cbe862e in set_env (msg=0x7f053e91d690) at exec_hf.c:547 #4 0x00007f053cbeb835 in w_exec_msg (msg=0x7f053e87c480, cmd=0x7f053ea5e168 "X֤>\005\177", foo=<optimized out>) at exec_mod.c:164 #5 0x00000000004274f7 in do_action (h=h@entry=0x7fff02e9cf90, a=a@entry=0x7f053ea5cfb8, msg=msg@entry=0x7f053e87c480) at action.c:1094 #6 0x00000000004261a9 in run_actions (h=h@entry=0x7fff02e9cf90, a=a@entry=0x7f053ea5cfb8, msg=msg@entry=0x7f053e87c480) at action.c:1583 #7 0x0000000000432980 in run_top_route (a=0x7f053ea5cfb8, msg=msg@entry=0x7f053e87c480, c=c@entry=0x0) at action.c:1669 #8 0x00007f053e610d2a in run_failure_handlers (t=t@entry=0x7f0534ecc770, rpl=0x7f053ea71040, code=486, extra_flags=extra_flags@entry=64) at t_reply.c:1051 #9 0x00007f053e612aa3 in t_should_relay_response (Trans=Trans@entry=0x7f0534ecc770, new_code=new_code@entry=486, branch=branch@entry=0, should_store=should_store@entry=0x7fff02e9d2a0, should_relay=should_relay@entry=0x7fff02e9d290, cancel_data=cancel_data@entry=0x7fff02e9d490, reply=reply@entry=0x7f053ea71040) at t_reply.c:1406 #10 0x00007f053e616126 in relay_reply (t=t@entry=0x7f0534ecc770, p_msg=p_msg@entry=0x7f053ea71040, branch=0, msg_status=msg_status@entry=486, cancel_data=cancel_data@entry=0x7fff02e9d490, do_put_on_wait=do_put_on_wait@entry=1) at t_reply.c:1809 #11 0x00007f053e61a9f3 in reply_received (p_msg=0x7f053ea71040) at t_reply.c:2493 #12 0x00000000004920a6 in do_forward_reply (msg=msg@entry=0x7f053ea71040, mode=mode@entry=0) at forward.c:783 #13 0x0000000000493637 in forward_reply (msg=msg@entry=0x7f053ea71040) at forward.c:885 #14 0x00000000004f5634 in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:275 #15 0x00000000005d929d in udp_rcv_loop () at udp_server.c:521 #16 0x00000000004a73f1 in main_loop () at main.c:1629 #17 0x0000000000425085 in main (argc=<optimized out>, argv=<optimized out>) at main.c:2561 Kind regards,/Tobias

2 3

TLS capture with sip_trace()
by Mikko Lehto 13 Jan '15

13 Jan '15

Hi I am getting incorrect source port to Homer web while tracking outgoing request from my proxy to remote SIP server. Juha Heinanen <jh(a)tutpro.com> wrote in another thread: > in case of tcp (and tls) the source port is always a random one. > only the destination port can be predetermined. Interface capture (tcpdump) shows exactly what Juha wrote i.e. random source port for TLS connection. I always get local TLS listener address as Homer source_port, clearly not true when comparing to output of tcpdump. I am calling sip_trace() from branch_route and onsend_route. Is it even possbile to use siptrace with TLS so that it records correct TCP (random) source port to Homer? I tried briefly printing some structure variables in siptrace/siptrace.c but it seems that real source port is not available in struct dest_info at that context. -- Mikko

2 1

siputils: E164 check is broken
by Seudin Kasumovic 12 Jan '15

12 Jan '15

Hi, The fix below is wrong, condition must be '<' instead of '<='. https://github.com/kamailio/kamailio/commit/fbcfa198a2c89868fa021f6a6377442… and backported in 4.2 https://github.com/kamailio/kamailio/commit/2d1032b3c8a1184357615b37da99dfd… Regards, -- Seudin Kasumovic

2 2

dispatcher html-rpc
by Yuriy Gorlichenko 12 Jan '15

12 Jan '15

Hello. How I must use this function for dynamic reload dispatcher without restarting me server?

3 4

Maxmind GeoIP2 implementation.
by Sergey Okhapkin 12 Jan '15

12 Jan '15

Is anybody working on implementation of geoip2 module for kamailio? The advantage of the new API is a seamless support of both ipv4 and ipv6 addresses. If there is no work in progress, I'm goild to take the duty to implement module geoip2 based on the existing geoip module. I don't think it worth to extend geoip module to support both maxmind database formats.

2 2

Problem with tag field with using a custom table for the permissions module.
by Jan Hazenberg 09 Jan '15

09 Jan '15

Hi, I'm trying to use a custom table for the permissions module and run into a issue with the tag field, all other fields seem to work fine. Here is my config: modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) modparam("permissions", "peer_tag_avp", "$avp(permissions_tag)") modparam("permissions", "trusted_table", "trunks") modparam("permissions", "source_col", "ip") modparam("permissions", "proto_col", "transport") modparam("permissions", "from_col", "from_pattern") modparam("permissions", "tag_col", "trunk_uuid") When i start kamailio i see the following error in the log: Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'trunk_uuid' in 'field list' Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: ERROR: permissions [address.c:112]: reload_address_table(): failed to query database Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: CRITICAL: permissions [address.c:363]: init_addresses(): reload of address table failed Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: ERROR: permissions [permissions.c:631]: mod_init(): failed to initialize the allow_address function Jan 6 21:49:57 sip /usr/sbin/kamailio[24327]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module permissions (/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so) Jan 6 21:49:58 sip kamailio: ERROR: <core> [daemonize.c:315]: daemonize(): Main process exited before writing to pipe If i change the tag_col to tag it works fine, both fields are varchar(64). And the field trunk_uuid exists in the trunks table. I'm running kamailio 4.2.1 Thanks, Jan Hazenberg

2 4

Camarillo 4.2 as outbound proxy with TLS support
by Maqbul Khan - Reform InfoTech 09 Jan '15

09 Jan '15

Hello, I just installed Kamailio for the first time 4.2 on Debian Wheezy. Everything went alright, I created two users, both can register and make calls, but with no audio. So the problems are: 1. No audio 2. I want to setup Kamailio as Outbound Proxy 3. I wish to have this proxy with TLS support. Please suggest solution Maq Khan

3 3

dialog timeout reinvite
by Kelvin Chua 09 Jan '15

09 Jan '15

Happy new year everyone! :) I have stumbled into a case where a switch would send a reinvite to my kamailio server every 15 minutes. I am setting dialog timeout using avp for prepaid purposes. now i think it resets the timer upon processing of the reinvite. what is the best way to work around this?

3 5

listening to specific interface with ip
by Jijo 09 Jan '15

09 Jan '15

HI All. I know that kamailio support listen on an interface with port (listen = eth0:5060). Does it support listen on an interface with ip, port, transport? For example listen=eth0: udp:10.10.10.10:5060 Thanks Jijo

2 1

Can you use the sr.modf() method / function to call append_branch?
by J Lee 08 Jan '15

08 Jan '15

I would like to know if there's a way to do parallel forking via lua code. So specifically, I'd like to be able to do something like this: seturi("sip:a@example.com"); append_branch("sip:b@example.com"); append_branch("sip:c@example.com"); append_branch("sip:d@example.com"); t_relay(); except via lua, because the list of branches is dynamic. I've tried the following: sr.modf("append_branch","sip:10.x.x.x") but the error message I'm getting is: /usr/sbin/kamailio[7437]: ERROR: app_lua [app_lua_sr.c:177]: lua_sr_modf(): function 'append_branch' is not available Just wondering if I "should" be able to do this. I've been reading http://www.kamailio.org/wiki/embeddedapi/devel/lua#srtm_package but haven't seen any references to append_branch. The other alternative I'm looking at is to just use lua to generate the list of destinations and pass back a hashtable of sorts to the cfg file. Then from there I will try to loop through the hashtable and do the append_branch command for each entry in the table. If you have other suggestions, I'd appreciate it. Thanks.

2 1

Kamailio v4.2.2 Released
by Daniel-Constantin Mierla 08 Jan '15

08 Jan '15

Hello, Kamailio SIP Server v4.2.2 stable release is out. This is a maintenance release of the latest stable branch, 4.2, that includes fixes since release of v4.2.1. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.2. For more details about version 4.2.2 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/01/kamailio-v4-2-2-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

Planning to release v4.2.2
by Daniel-Constantin Mierla 08 Jan '15

08 Jan '15

Hello, I am considering to release next patch update in 4.2 series next week, most probably on Thursday, January 8, 2015. Given the winter holidays, if there are issues not being taking care of, send a reminder to mailing list or, even better (less chances to be forgotten), open a case on Github issue tracker: - https://github.com/kamailio/kamailio/issues Wishing everyone a great 2015! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 7

dispatcher weight dont work
by Yuriy Gorlichenko 08 Jan '15

08 Jan '15

Hello I use dipatcher algorithm that works with weight. I added 2 Asterisks and try to call its with my kam, but this still works like 4 algorithm. Weight does not work. How I must configure dispatchr for working with weight? My configuration now is modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@10.0.1.1") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("tm", "reparse_on_dns_failover", 0)

2 3

Dialog update timeout
by Pars3c 07 Jan '15

07 Jan '15

Hi, I wish to update dialog timeout when i receive the 200ok in the on_reply route I try to make it with the dlg_set_timeout but it give an error "bogus dialog..." How can i make it? Thanks

4 12

3rd party REGISTER body
by Jyaim Jyaim 07 Jan '15

07 Jan '15

Hello, I'm wondering if I can configure Kamailio to get in 3rd party REGISTER request's body the initial REGISTER request received from user ? Thanks in advance and best regards.

4 6

Hiding topology header From
by Kalala Alexander 07 Jan '15

07 Jan '15

2 2

Queue implementation in kamailio
by Ansuman Manna - Asergis Telecom 07 Jan '15

07 Jan '15

Hi, I am developing a system using Kamailio 4.2.1 as a front-end to a group of Asterisk servers. All of the usual functionality (registations, inbound, outbound, internal calls) are fine. We need to add call queueing to the system. I looked at the mohqueue module and read that we can use mohq_process() to send inbound calls to a suitably defined queue, based on the incoming URI. But think is that how can I get sip extension status in the particular queue? Any help would be appreciated. Thanks. Regards, Ansuman Manna Software Developer Asergis Telecom Services Private Limited Office Tel: +91 11 6606 8400 Office Fax: +91 112554 5156 http://www.asergis.in <http://www.asergis.in>

2 4

Hiding topology header From
by Kalala Alexander 07 Jan '15

07 Jan '15

1 1

Unregister is failing
by Anil Kumar 07 Jan '15

07 Jan '15

Hi All, I am running ims servers(pcscf,scscf,icscf and hss) as part of kamailio proxy. And I am trying to register and unregister the end-points, with help osip lib. But I see the below ERROR message. ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio/pcscf/kamailio.cfg] l=908 a=25 n=t_check_status Please help me to fix the issue. Thanks and Regards, Anil

2 1

No audio when connect from public network but it works for lan users
by CK Lee 06 Jan '15

06 Jan '15

I am new to Kamailio after starting to play around for 3 weeks. Before Kamailio, I used asterisk and it works quite well in my network which has 2 public IP addresses and 10+ wired and wireless users. The asterisk listens to only one IP which my netgear router port forward the necessary sip and rdp ports to the asterisk. My interest is to use Kamailio as SIP proxy to sit in front of the asterisk. I follow the tutorial here - http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I use debian wheezy and kamailio is installed from the repository with version 4.2.1 Everything works for my lan subscribers. However, when I am outside the lan, I can register but do not the incoming packets at my sip client. What I observe is two strange behaviours 1. in asterisk, the kamailio peer has to use "sip.example.org or its public IP" in the host name field, otherwise it is not online 2. in looking at the log, I found the sip client peer is registered at my lan address i.e. 192.168.118.xx instead of a public IP when I directly login to asterisk. I add on rtproxy but not succeed with error as force_rtp_proxy_body: incorrect port 0 in reply from rtpproxy Below is my kamailio.cfg Help is appreciated. -------------------------------------- #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT #!define WITH_TLS #!define ADDR_IPV4 192.168.118.30 # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://user:password@localhost/asterisk" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="sip.example.org" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:192.168.118.30:5060 listen=tls:192.168.118.30:5061 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_XCAPSRV tcp_accept_no_cl=yes #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT #tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.118.30" desc "Asterisk IP Address" asterisk.bindport = "15066" desc "Asterisk Port" kamailio.bindip = "192.168.118.30" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/i386-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "outbound.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- modparam("ctl", "user", "kamailio") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$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) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "name") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") #modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rtpproxy_offer(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sipusers", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif

2 4

Kamailio Active / Active HA.
by Mahmoud Ramadan Ali 05 Jan '15

05 Jan '15

Hi Dears, I have successfully configured Kamailio HA using Heartbeat and Pacemaker so if one of the two servers should go down the other server will own the virtual IP address and take over. But i have two questions: 1.This model is considered to be Active / Passive redundancy so one server will process the SIP signaling until it goes down and i'm wondering is there is any way two achieve Active / Active redundancy and if so how the signaling will be handled on the servers so they can be aware of the transactions and dialogs traversing the servers ? does the signalling will be replicated or synchronized between the servers or what ?! 2.What about the servers DB and how they should be designed in a cluster mode ? i want to replicate all the DBs of the server to get consistent user registration using the subscriber table for example. Thanks in advance and Best regards.

7 14

Kamailio fails to generate core dump
by Alex Balashov 03 Jan '15

03 Jan '15

Hi, I recently had a simultaneous crash on three production instances of Kamailio. Unfortunately, I have not been able to get to the bottom of the issue because Kamailio refused to produce a core dump: Dec 30 15:37:29 xxxxxx /usr/local/sbin/kamailio[13743]: ALERT: <core> [main.c:784]: handle_sigs(): child process 13763 exited by a signal 11 Dec 30 15:37:29 xxxxxx /usr/local/sbin/kamailio[13743]: ALERT: <core> [main.c:787]: handle_sigs(): core was not generated I do not have disable_core_dump=yes (default is no), and checked ulimits for limits on core file size. Kamailio was running as root, and there was no limit. What are the other reasons why Kamailio may not generate a core dump in such a case? Thanks! -- Alex -- 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 5

Double attempt to record-route, RR param not added
by ron.kamailio＠mcleodnet.com 03 Jan '15

03 Jan '15

I have been away from Kamailio for a while and starting working on a project where it would be a good fit. I am having a problem when I try to add a Record-Route header with a tag. I see a message in the log indicating that a Double attempt to record-route was made, and the tag that I specified (;rtpproxy=yes), does not get included. I have checked-over the scripts defined in the kamailio.cfg, and I am very sure that record_route() does not get called more than once. Also, the log only shows record_route being execute a single time. Can anyone suggest how I can troubleshoot this problem further? 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=206 a=57 n=set_adv_addr 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=207 a=58 n=set_adv_port 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=208 a=25 n=record_route_preset 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=209 a=62 n=force_send_socket 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=210 a=2 n=exit 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=235 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=231 a=41 n=isflagset 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=232 a=25 n=add_rr_param 5(32087) DEBUG: rr [record.c:899]: add_rr_param(): rr_param_buf=<;rtpproxy=yes> 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=235 a=25 n=xlog 5(32087) ERROR: <script>: new ru: sip:19999009595@192.168.100.52:5080 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=236 a=24 n=record_route 5(32087) ERROR: rr [rr_mod.c:262]: w_record_route(): Double attempt to record-route 5(32087) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[//etc/kamailio/kamailio.cfg] l=237 a=5 n=route 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=247 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=244 a=25 n=is_method 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=246 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=245 a=25 n=t_is_set 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=245 a=25 n=t_on_branch 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=250 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=247 a=25 n=is_method 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=249 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=248 a=25 n=t_is_set 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=248 a=25 n=t_on_reply 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=254 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=250 a=25 n=is_method 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=252 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=251 a=25 n=t_is_set 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=251 a=25 n=t_on_failure 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=257 a=16 n=if 5(32087) ERROR: *** cfgtrace:request_route=[RELAY] c=[//etc/kamailio/kamailio.cfg] l=254 a=24 n=t_relay The complete log file, script files, and SIP trace can be found here: http://www.siplabs.net/tracebin/record-route/kamailio-log.txt http://www.siplabs.net/tracebin/record-route/kamailio.cfg http://www.siplabs.net/tracebin/record-route/bgcf-local.cfg http://www.siplabs.net/tracebin/record-route/sip-trace.txt Thanks and Happy New Year, Ron

2 3

presence - Error deleting subscription
by Jan Gaida 03 Jan '15

03 Jan '15

Hello, We are upgrading our servers to Kamailio 4.2 and we see a lot of error messages "Failed to delete subscription from memory". Scenario: This happens every time there is a Notify Timeout (408) and Kamailio tries to delete the subscription. Consequences: I don't know whether the delete fails because the entry did never exist, was already deleted or because the comparison (to_tag, callid or from_tag) is incorrect. It would be nice to at least have some details in the log message. I have not noticed any consequences to general functionality. I suspect the commit (74d73953327ad7702922b27ba37c71e3283659c3) to be responsible for this error because it introduced some extra checks on deletion. Does this happen to anybody else? What may be configured incorrectly on our system? Kind regards *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.gaida(a)grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario.

2 4

t_suspend()/t_continue() problem
by Dmytro Bogovych 02 Jan '15

02 Jan '15

Greetings. I'm trying to adopt kamailio to handle incoming calls / generate push notifications for softphone running on Windows Phone 8. Starting point was this publication http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin… I adopted the script from publication a bit - now notification URI is extracted from REGISTER messages + INVITE transactions are not resuming when unregistering happens. However it does not work for me yet. I see the suspicious output in kamailio log when softphone registers after incoming push notification. Please ignore ERROR: level name - it is just dirty logging call, not real problem. Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: ERROR: <script>: resuming trasaction [40951:63820139] dbogovych (40951:63820139) Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: tm [t_lookup.c:1635]: t_lookup_ident(): DEBUG: t_lookup_ident: transaction found Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:181]: lookup(): contact for [dbogovych] found by address Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 90211984 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91061218 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91229062 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 90750343 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 87612750 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: DEBUG: registrar [lookup.c:327]: lookup(): instance is 91841578 Dec 17 14:20:11 voipobjects /usr/local/sbin/kamailio[29652]: WARNING: tm [t_lookup.c:1537]: t_unref(): WARNING: script writer didn't release transaction Can this "script writer didn't release transaction" be the problem? Thank you :)

1 2

Happy New Year!
by Daniel-Constantin Mierla 01 Jan '15

01 Jan '15

A very dynamic 2014 for Kamailio has reached its end! Thank you everyone for contributing to it! Looking forward to 2015, a lot of new features in Kamailio and new contributors! I wish a healthy and prosperous year to all Kamailio friends, hoping to meet many of you at Kamailio World Conference and other events around the globe! Happy New Year! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 3

Making media packets associated with calls appear unidirectional using rtpproxy and kamailio
by aft 31 Dec '14

31 Dec '14

Hi, I'm trying use two instances of rtpproxy listening to two different public IPs, to handle incoming and outgoing legs. That means i want to implement this network topology: SipClient--------->media from client to proxy--->rtpproxy1 Sipclient<----------media from proxy to client<---rtpproxy2 Where rtpproxy1 listening to IP1 and rtpproxy2 listening to IP2 in kamailio cfg : modparam("rtpproxy", "rtpproxy_sock", " udp:127.0.0.1:22222 udp: 127.0.0.1:22223") if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } rtpproxy_offer("IP1"); } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } rtpproxy_answer("IP2"); } But its not working. Thanks in advance

1 0

kamailio 4.2.1 snmpstats module depends on libsnmp15
by René Bormann 31 Dec '14

31 Dec '14

hello guys, just wanted to add the snmpstats modules to our kamailio 4.2.1 installation. but we are running into a libssnmp15 dependency. searching google showed me that on 14.04 the lib is called libsnmp30 now. does anybody know how to fix this? thanks, rené ------------------- root@kamailio1:~# apt-get install kamailio-snmpstats-modules Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: kamailio-snmpstats-modules : Depends: libsnmp15 (>= 5.4.3~dfsg) but it is not installable E: Unable to correct problems, you have held broken packages. root@kamailio1:~#

2 1

Dialog_end from event route
by Dan-Cristian Bogos 30 Dec '14

30 Dec '14

Hey Guys I need assistance for the following case: I receive dialog_end request from external component in evapi:message-received block and would like to somehow kill the dialog corresponding to h_id and h_entry I get in the message. Is there any way to do that without involving mi_*? Any tip appreciated! Ta, DanB.

3 2

about kamailio
by ThatOther Dev 30 Dec '14

30 Dec '14

is it possible with ypur server project to call to country mobile networks just from a pc, without special equipment?

2 1

ul.dump failing when more than 26 users are registered
by João Vitor Arruda 30 Dec '14

30 Dec '14

Hello, I´m doing a test with the ul.dump output on both Kamailio 4.1.5 or 4.2.1 and it works great if the number of registered users is 26 or lower. Starting with 27 registered users the "kamcmd ul.dump" command returns the error: "error: 500 - Internal error creating aor struct" In Kamailio 4.2.1 the new (and nice) "kamcmd stats.get_statistics usrloc:" returns the correct number of registered_users without this "26 limit". Its possible to check why ul.dump is failing this way? Thanks,

2 3

'from' user != auth username
by Dima T. 30 Dec '14

30 Dec '14

Hello, I need migrate users from the old proprietary softswitch but they have predefined from/to header which is differ from auth username. I read about flags in auth_check but I need strict pairs from user/auth username. How this authentication scheme could be implemented with Kamailio? -- Dmitry

2 1

kamailio configuration problem
by javier falbo 30 Dec '14

30 Dec '14

Hi, I would like some help from a kamailio expert. I have installed on a Debian7 64 kbytes VPS this tutorial: http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour, Problem is that there is no voice in a zrtp connection, but i hear only the ring sound ok. Strange is that on my local wifi works all perfect including the voice, but with external ip no. Any help? I would like to hire a kamailio expert for fixing this and also adding some extras. Write me directly so we could get in contact. regards, javier

2 2

Re: [SR-Users] bug in allow_source_address?
by Henry Fernandes 29 Dec '14

29 Dec '14

Does anyone have feedback about this? -H From: Henry Fernandes <henry(a)usinternet.com> Date: Wednesday, December 17, 2014 at 3:39 PM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: bug in allow_source_address? I¹m using allow_source_address in my Kamailio config and have been for months/years. I¹ve had no issues with it until today. The issue is that allow_source_address("1²) returns ³1" (for $si=123.17.29.201 and $sp=5060) even though we don¹t have this IP address in the address¹ table for group 1. We have defined this IP address for group 2 but not group 1. select * from address order by mask; +----+-----+---------------+------+------+----------+ | id | grp | ip_addr | mask | port | tag | +----+-----+---------------+------+------+----------+ | 20 | 1 | 123.17.0.0 | 23 | 5061 | Testing | | 22 | 1 | 123.17.0.0 | 23 | 5062 | Testing | | 23 | 1 | 123.17.0.0 | 23 | 5063 | Testing | | 24 | 1 | 123.17.0.0 | 23 | 5064 | Testing | | 1 | 1 | 123.17.0.0 | 23 | 5060 | | | 4 | 1 | 20.5.176.0 | 24 | 5060 | | | 21 | 1 | 123.17.1.235 | 32 | 5060 | Override | | 46 | 2 | 123.17.29.201 | 32 | 5060 | customer | +----+-----+---------------+------+------+----------+ Can anyone explain this? I¹m wondering if there¹s a bug in how allow_source_address does its net mask calculations. -H

2 2

about destination blacklisting
by Juha Heinanen 29 Dec '14

29 Dec '14

i made a test where domain name in request uri hostpart has two udp srv records, one with higher priority than the other. when i turn off the higher priority uac, there is about 5 sec delay before the lower priority uac starts to ring. if i make another call within a small number of seconds from the previous one, my sip proxy remembers that the higher priority uac is dead and the lower priority one starts to ring immediately. but if i wait about five seconds or more, my sip proxy again first tries the higher priority uac and there is again the about 5 second delay before the lower priority one starts to ring. i have in config use_dst_blacklist=on dst_blacklist_expire=20 from the docs i get the impression that my sip proxy should remember 20 seconds that the higher priority uac is dead, but it only remembers it about 5 seconds. what could be the explanation? -- juha

1 0

RTPProxy metod UPDATE
by Kalala Alexander 29 Dec '14

29 Dec '14

1 0

What's happening? Follow us on more channels than e-mail!
by Olle E. Johansson 29 Dec '14

29 Dec '14

Friends, The Kamailio project has great mailing lists for discussions, but did you know that we also spread news in various other channels? If you use these channels, please follow us there and spread the word about Kamailio! Facebook https://www.facebook.com/Kamailio Twitter @kamailioproject Google plus has both a Kamailio page and a discussion group - Page https://plus.google.com/u/0/b/114984922071338017631/114984922071338017631/p… - Group https://plus.google.com/communities/104535904708990386927 In addition to the news from the web site there are a few additional tips and tricks coming on these channels. See you there! /O

1 0

SIPURI sending private IP address
by Cibin Paul 28 Dec '14

28 Dec '14

Hi, I have a kamailio 4.1 as a gateway and registrar behind asterisk servers. On asterisk, I save the IP address of the originating call parsing the SIPURI or SIPCHANNEL(recvip). In both cases, I am receiving the private ip of the user agent registered with kamailio. Do I need to change anything in kamailio to receive the actual IP address. Please suggest. Regards Cibin

3 5

Meddling with password during authentication
by Olli Heiskanen 27 Dec '14

27 Dec '14

Hello all, During authentication, is there any way to affect the password user is sending? I do suspect not as it is a clear security matter, but won't hurt to ask. I use auth_db module with calculate_ha1 parameter set to 1. For reasons in integrating Kamailio into my system architecture there is a need to store a password in some other format than for example md5('555:domain.com:password)') while not allowing any passwords to be stored as plaintext. For example: md5('555:domain.com:md5('password')') but this would require me to hash the password before authentication, in Kamailio script as I can't do it in the clients. Reason for this question is to have my users in a separate database, and these users could have 0-n sip peers assigned to them, and have users authenticate to my software and the sip peers using the same password. cheers, Olli

3 3

Authorization Issue kamailio 4.2
by Ahmed Salem 27 Dec '14

27 Dec '14

Hi, I'm using kamailio 4.2 some of my clients are getting 401 unauthorized after a couple of days of usage without any problems, their usernames and passwords are still in the database server unchanged. Any idea what's going on? Thanks Ahmed

3 2

Merry Christmas!
by Daniel-Constantin Mierla 25 Dec '14

25 Dec '14

Another year getting slowly to the end, an excellent opportunity to thank to all old and new friends of Kamailio project, from contributing code to using it, helping on mailing lists and other places online, or advertising it across the world! Merry Christmas and Happy Winter Holidays! Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 3

Media trouble with kamailio/rtpengine
by Marc Soda 24 Dec '14

24 Dec '14

I'm trying to use Kamailio and rtpengine as a webrtc gateway. I'm not getting audio back to my browser. From a packet capture I can see media from the browser to rtpengine, and then bi-directional RTP back and forth from my asterisk server, but rtpengine is not sending the media on to the browser, i.e.: browser ---------> kamailio/rtpengine <---------> asterisk This is the output from rtpengine: https://gist.github.com/marcantonio/bfe72644306b205cc7e1 Thanks.

2 4

Dispatcher weight nor working
by Yuriy Gorlichenko 24 Dec '14

24 Dec '14

Hello. We use kamailio 4.3 and dispatcher with 9 algorithm. We use db instanse for dispatcher and at attrs column for our backend servers set WEIGHT=40 and 60 for first and second server, but packets sended only at first server ignoring weight

1 0

Re: [SR-Users] Mediaproxy do not work well with Kamailio SIP server issue
by Alfred C.M. Lu 24 Dec '14

24 Dec '14

|---------+-----------------------------> | | dong-ming.li(a)mail.| | | foxconn.com | | | | | | 2014/12/24 上午 | | | 11:54 | |---------+-----------------------------> >--------------------------------------------------------------------------------------------------------------------------------------------------| | | | To: alfred.lu(a)foxconn.com | | cc: | | Subject: Mediaproxy do not work well with Kamailio SIP server issue | >--------------------------------------------------------------------------------------------------------------------------------------------------| Secret Level: |------------------------| | ( ) 1.Non-Confidential | | ( ) 2.Confidential | | ( ) 3.Secret | | ( ) 4.Top Secret | |------------------------| Priority Level: |--------------------| | ( ) 1.Normal | | ( ) 2.Urgent | | ( ) 3.Extra Urgent | |--------------------| ------------------------------------------------------------------------- Hi, sir, We try to setup Mediaproxy on CentOS 6.6 with kamailio-4.2.1 SIP server. We use Linphone android version as SIP client to test. attached kamailio.cfg.txt is our kamailio-4.2.1 SIP server config file. pls help to rename kamailio.cfg.txt to kamailio.cfg. kamailio-4.2.1 SIP server is run on public IP server,and Linphone SIP client is run behind NAT. The problem is that only callee can hear caller's rtp audio data. caller can not hear callee's audio data. And the call will auto disconnect after about 30s time out. Below's our modification regarding the kamailio.cfg file,pls help to check if this is OK or not ? #foxconn gallice add start #### MediaProxy module loadmodule "mediaproxy.so" modparam("mediaproxy", "disable", 0) modparam("mediaproxy", "ice_candidate", "high-priority") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock") modparam("mediaproxy", "mediaproxy_timeout", 500) modparam("mediaproxy", "signaling_ip_avp", "$avp(nat_ip)") modparam("mediaproxy", "media_relay_avp", "$avp(media_relay)") #foxconn gallice add end # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting use_media_proxy(); #foxconn gallice add } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { #foxconn gallice add #route(NATMANAGE); #foxconn gallice delete use_media_proxy(); #foxconn gallice add } #foxconn gallice add } For mediaproxy,we can see the media related processes are already run. 31212 ? SL 0:01 python ./media-dispatcher start 31218 ? SL 0:16 python ./media-relay start ***************************************************** dong-ming / 李東明 Foxconn Electronics Inc. HH Precision Ind. Co., Ltd. CNSBG NSDI TIPBU R&D Dept. LH Office 地址:深圳市龍華新區龍華東環二路二號富士康科技集團分機 : +86-0755-2812-9588 #26109 Mobile : 13510123670 Mail : dong-ming.li(a)mail.foxconn.com, ***************************************************** 溫馨提醒﹕入廠人員須遵守富士康出入管制,請勿攜帶手提電腦、各種攝像、U盤、MP3 等存儲設備本電子郵件及附件所載信息均為保密信息，受合同保護或依法不得洩漏。其內容僅供指定收件人按限定範圍或特殊目的使用。未經授權者收到此信息均無權閱讀、使用、複製、洩漏或散佈。若您因為誤傳而收到本郵件或者非本郵件之指定收件人，請即刻回覆郵件或致電Super Notes郵件客服熱線 560-104，並永久刪除此郵件及其附件和銷毀所有複印件。謝謝您的合作！ This e-mail message together with any attachments thereto (if any) is confidential, protected under an enforceable non-disclosure agreement, intended only for the use of the named recipient(s) above and may contain information that is privileged, belonging to professional work products or exempt from disclosure under applicable laws.Any unauthorized review, use, copying, disclosure, or distribution of any information contained in or attached to this transmission is STRICTLY PROHIBITED and may be against the laws. If you have received this message in error, or are not the named recipient(s), please immediately notify the sender by e-mail or telephone at Super Notes support hotline 560-104 and delete this e-mail message and any attached documentation from your computer. Receipt by anyone other than the intended recipient(s) is not a waiver of any attorney-client or work product privilege. Thank you! mail from ip-->192.168.2.2 mail from pc-->MICROSOF-700F48 Version: Super Notes 1.6.4.5B (See attached file: kamailio.cfg.txt)

1 0

TLS enable false.
by Thanh Truong 23 Dec '14

23 Dec '14

Hi all, I have tried several configure TLS in kamailio but no luck. Please give me some suggestion that I can make it work correctly. This is my configure in TLS module. modparam("tls", "tls_method", "SSLv23") modparam("tls", "private_key", "/usr/local/etc/kamailio/ca/privkey.pem") modparam("tls", "certificate", "/usr/local/etc/kamailio/ca/kamailio1_cert.pem") modparam("tls", "ca_list", "/usr/local/etc/kamailio/ca/calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) I am only getting issue with verify_certifiate = 1, it i let it to 0, my client can register correctly. When I set this flag, i got message from server as: Dec 18 10:26:31 17237 /usr/local/sbin/kamailio[12655]: ERROR: <core> [tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: tls [tls_server.c:1193]: tls_read_f(): TLS accept:error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: <core> [tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading I Cant add any pem file into client, i have used Blink phone but no luck. Thank all in advance. ThanhTruong.

3 3

[BOUNTY] Configs for Video calls between Linphone Android App and WebRTC clients (JsSIP, SIPML)
by Andrey Utkin 23 Dec '14

23 Dec '14

I hope the subject line says it all. I need to set up a backend to establish calls between browser clients and usual SIP clients, and the abovementioned applications are of the highest interest. (Of course, calls between same UAs must work, too.) If Linphone is not going to work due to its internal issues, then other established open source mobile apps for iOS and Android will be fine. Currently I have configured latest kamailio+rtpengine with configs from here https://github.com/caruizdiaz/kamailio-ws The calls pass this way currently: jssip -> android: no sound from phone to the browser, i see that jssip sends "sendonly" attribute for audio in INVITE's SDP. Audio from browser to phone, and both video streams appear immediately, everything is fine with them. android -> jssip: video from browser to the phone appears in 1-2 _minutes_ after the call is answered. All other media streams are fine. The above results are the same in such browsers [IP-] [ ] www-client/firefox-bin-31.3.0:0 [IP-] [ ~] www-client/google-chrome-unstable-41.0.2243.0_p1:0 With sipml currently i have no stable results, so it's hard to describe what happens. Please contact me if you have configs to make the needed things work, or if you have experience of such things working stable, and can configure it quickly. -- Andrey Utkin

2 4

websocket ws_handle_handshake() error
by jaflong jaflong 23 Dec '14

23 Dec '14

Hi list, I am getting this error "ERROR: websocket [ws_handshake.c:153]: ws_handle_handshake(): retrieving connection" Any idea why? This is the debug log 12(24113) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... 12(24113) DEBUG: xhttp [xhttp_mod.c:358]: xhttp_handler(): new fake msg created (537 bytes): <GET /ws HTTP/1.1 Via: SIP/2.0/TLS 10.1.1.1:45968 Host: 10.1.1.2:10443 User-Agent: Mozilla/5.0 (X11; Linux x86_64; rv:34.0) Gecko/20100101 Firefox/34.0 Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 Accept-Language: en-US,en;q=0.5 Accept-Encoding: gzip, deflate Sec-WebSocket-Version: 13 Origin: http://mydomain.co.uk Sec-WebSocket-Protocol: sip Sec-WebSocket-Key: GRbV+ovG+DKIBJtwb4pRiQ== Connection: keep-alive, Upgrade Pragma: no-cache Cache-Control: no-cache Upgrade: websocket > 12(24113) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: 12(24113) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <GET> 12(24113) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: </ws> 12(24113) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <HTTP/1.1> 12(24113) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 12(24113) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 12(24113) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via 12(24113) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) 12(24113) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header 12(24113) DEBUG: <core> [msg_translator.c:205]: check_via_address(): check_via_address(10.1.1.1, 10.1.1.1, 0) 12(24113) DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (24113 (12)), reusing fd 12(24113) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... 12(24113) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fc08162ad68 n=421 fd=7 12(24113) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf= �/�*�޹��!��S��zo�e�/D��V�Gs��s�� ^cd��S"Ŏ�K��<�c7��4lx�<Z��#�c(AJ�8V��d� ��4bF��Wu�(�w�a� f��n��s�KjQv�݈�q|Z��64{Wk��Z\ױ��M�,[�Arٵ��Q�_��u�d�Gza�f��I�fI��z��Š9=��Ħ3�ȀJPx�}��S?��eVe��\id��y�wP�r(kZs��Z�"��4`�N_MR��z�!��Eb}��+�\;�8�3��V��l��M�|�1�2��VVK�@%�a�B&�P��k 12(24113) INFO: <script>: Received HTTP request to /ws from [10.1.1.1:45968] with protocol tls 14(24115) DEBUG: <core> [tcp_main.c:3565]: handle_ser_child(): handle_ser_child: read response= 7fc08162ad68, -1, fd -1 from 12 (24113) 14(24115) DEBUG: tls [tls_server.c:604]: tls_h_close(): Closing SSL connection 0x7fc081626e20 12(24113) ERROR: websocket [ws_handshake.c:153]: ws_handle_handshake(): retrieving connection 12(24113) DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) 12(24113) DEBUG: <core> [msg_translator.c:205]: check_via_address(): check_via_address(10.1.1.1, 10.1.1.1, 0) 12(24113) ERROR: sl [../../forward.h:247]: msg_send(): tcp_send failed 12(24113) ERROR: websocket [ws_handshake.c:113]: ws_send_reply(): sending reply 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) 12(24113) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) 12(24113) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up 12(24113) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0xa31f40, 7, -1, 0x10) fd_no=2 called 12(24113) DEBUG: <core> [tcp_read.c:1434]: release_tcpconn(): releasing con 0x7fc08162ad68, state -2, fd=7, id=3 12(24113) DEBUG: <core> [tcp_read.c:1435]: release_tcpconn(): extra_data 0x7fc081626e20 14(24115) DEBUG: <core> [tcp_main.c:3331]: handle_tcp_child(): handle_tcp_child: reader response= 7fc08162ad68, -2 from 2 9(24110) DEBUG: websocket [ws_conn.c:459]: wsconn_get_list(): wsconn_get_list

1 0

Rpc dialog variable
by Pars3c 22 Dec '14

22 Dec '14

Hi How can i do to see the dialog variables like $dlg_var with RPC ? It's possible? Thanks

1 0

ERROR: dispatcher [dispatch.c:1402]: ds_load_remove(): cannot find load for
by Will Ferrer 22 Dec '14

22 Dec '14

Hi All We recently discovered that our dispatcher boxes have this error constantly at the end of our calls. ERROR: dispatcher [dispatch.c:1402]: ds_load_remove(): cannot find load for (<callid goes here>) We aren't seeing any issues arising from it that we can notice but wanted to see if any one knew what it meant and what it might cause. Thanks much in advance. All the best Will Ferrer Switchsoft Inc

2 7

Kamailio not start if redis server is down
by pars3c 22 Dec '14

22 Dec '14

Hi, i'm trying to use the ndb_redis connector , but i've a problem when redis server is down and i try to restart kamailio. Kamailio in this case not start and release this log: 2(19590) ERROR: ndb_redis [redis_client.c:112]: redisc_init(): error communicating with redis server [srv1] (xxxxxx/0): Connection refused 2(19590) ERROR: ndb_redis [ndb_redis_mod.c:126]: child_init(): failed to initialize redis connections 2(19590) ERROR: <core> [sr_module.c:927]: init_mod_child(): init_mod_child(): Error while initializing module ndb_redis (//lib/kamailio/modules/ndb_redis.so) 30(19622) ERROR: <core> [tcp_main.c:4962]: tcp_init_children(): ERROR: tcp_main: fork failed: Connection refused 31(19624) : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 11 31(19624) : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 31 0(19588) ALERT: <core> [main.c:775]: handle_sigs(): child process 19589 exited normally, status=255 0(19588) ALERT: <core> [main.c:775]: handle_sigs(): child process 19590 exited normally, status=255

2 2

rtpengine stats
by Marc Soda 22 Dec '14

22 Dec '14

Does anyone know how a can get stats from rtpengine? I see the $rtpstat pseudo variable in Kamailio, but from the documentation it looks like that will only give me stats on a particular call. I'm looking for overall stats like concurrent calls, bandwidth, etc... Thanks, Marc

3 2

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