- sr-users - kamailio.org

AliasDB module & Forking
by Marino Mileti 05 Nov '14

05 Nov '14

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

2 3

DTMF Tones question
by Anthony Martin 05 Nov '14

05 Nov '14

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

3 2

DBURL
by shmick＠riseup.net 05 Nov '14

05 Nov '14

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

2 3

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

05 Nov '14

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

2 4

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

04 Nov '14

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

2 1

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

04 Nov '14

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

2 1

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

03 Nov '14

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

2 3

Problems using htable iterator
by Glenn Larsen 03 Nov '14

03 Nov '14

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

2 1

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

03 Nov '14

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

2 4

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

03 Nov '14

2 1

question
by Javier Ricke 03 Nov '14

03 Nov '14

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

2 1

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

03 Nov '14

2 1

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

03 Nov '14

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

3 3

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

03 Nov '14

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

2 1

using time stamp
by bob sacamano 03 Nov '14

03 Nov '14

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

2 3

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

02 Nov '14

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

2 1

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

01 Nov '14

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

2 1

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

01 Nov '14

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

2 4

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

31 Oct '14

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

2 4

pipelimit rate question
by Alex Balashov 31 Oct '14

31 Oct '14

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

2 4

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

31 Oct '14

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

2 6

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

31 Oct '14

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

2 4

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

31 Oct '14

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

4 6

cSeq increasing
by Yuriy Gorlichenko 30 Oct '14

30 Oct '14

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

4 8

UAC INVITE to Provider
by Yuriy Gorlichenko 30 Oct '14

30 Oct '14

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

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