- sr-users - kamailio.org

No ACK from MS Teams
by Phillman25 Kyriacou 11 Feb '21

11 Feb '21

Dear List Hope this email finds you all well. I have followed the below tutorial on how to integrate Kamailio with MS Teams https://skalatan.de/en/blog/kamailio-sbc-teams However i have been facing an issue with MS Teams with direct routing where MS Teams does not send back an ACK after a 200 OK. The current inbound call flow is the following: *MS Teams ==> Kamailio==>Asterisk* Once Asterisk answers with 200 OK we send this 200 OK back to MS Teams however, MS Teams just doesn't answer back with ACK and call drops. The below trace shows the INVITE coming from MS Teams and the 200 OK we send back. |||||||||||||||||||| ==================== tag: rcv pid: 26135 process: 23 time: 1613025911.983278 date: Thu Feb 11 08:45:11 2021 proto: tls ipv4 srcip: 52.114.132.46 srcport: 4352 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ INVITE sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone;transport=tls SIP/2.0 FROM: Phillip Kyriacou<sip:+357XXXXXXXX@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c3da9477e05d45fca31c24a155af3318 TO: <sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: 9cb9a2f3144e594c87bccda76791c28e MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK0cf2c45 RECORD-ROUTE: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> CONTACT: <sip:api-du-b-euwe.pstnhub.microsoft.com:443 ;x-i=6e5254d9-79c2-4657-b54b-68791aa8e81f;x-c=9cb9a2f3144e594c87bccda76791c28e/d/10/e21a 6ff7125e4cbd91e75ec687fd4c5d> CONTENT-LENGTH: 1097 MIN-SE: 300 SUPPORTED: timer USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.2.9.1 i.USEA.7 CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY SESSION-EXPIRES: 3600 v=0 o=- 170239 0 IN IP4 127.0.0.1 s=session c=IN IP4 52.113.40.94 b=CT:10000000 t=0 0 m=audio 51414 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118 c=IN IP4 52.113.40.94 a=rtcp:51415 a=ice-ufrag:BwFs a=ice-pwd:sFyvnfQ9iH101E80ZKjCxi9+ a=rtcp-mux a=candidate:1 1 UDP 2130706431 52.113.40.94 51414 typ srflx raddr 10.0.137.19 rport 51414 a=candidate:1 2 UDP 2130705918 52.113.40.94 51415 typ srflx raddr 10.0.137.19 rport 51415 a=candidate:2 1 tcp-act 2121006078 52.113.40.94 49152 typ srflx raddr 10.0.137.19 rport 49152 a=candidate:2 2 tcp-act 2121006078 52.113.40.94 49152 typ srflx raddr 10.0.137.19 rport 49152 a=label:main-audio a=mid:1 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Igc2LVM9FH5kQtkRpHw99H5SB7Rd7eKzJy4gdWJg|2^31 a=sendrecv a=rtpmap:104 SILK/16000 a=rtpmap:9 G722/8000 a=rtpmap:103 SILK/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=ptime:20 |||||||||||||||||||| ==================== tag: snd pid: 26086 process: 6 time: 1613025917.726664 date: Thu Feb 11 08:45:17 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.132.46 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 200 OK Via: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK0cf2c45 Record-Route: <sip:SBC-IP-ADDR:5060;ftag=c3da9477e05d45fca31c24a155af3318;lr=on> Record-Route: <sip:SBC-FQDN:5061;transport=tls;ftag=c3da9477e05d45fca31c24a155af3318;lr=on> Record-Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> From: Phillip Kyriacou<sip:+357XXXXXXX@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c3da9477e05d45fca31c24a155af3318 To: <sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone>;tag=as659924a4 Call-ID: 9cb9a2f3144e594c87bccda76791c28e CSeq: 1 INVITE Server: MediaGW V1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:357XXXXXXXX@ASTERISK_PUBLIC_IP:5060> Content-Type: application/sdp Require: timer Content-Length: 339 v=0 o=root 411266324 411266324 IN IP4 ASTERISK_PUBLIC_IP s=ASTERISK c=IN IP4 ASTERISK_PUBLIC_IP t=0 0 m=audio 14416 RTP/SAVP 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9PcaAGjbemJKTERlFTcVwmLRQoDSEQPxX0L1a6RF a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv |||||||||||||||||||| Am I doing anything wrong here that MS Teams wont respond to my 200 OK? The Kamailio SBC is showing active on MS Teams and Kamailio dispatcher shows all the MS Team hubs as AP. I also can't see any errors in the log file My TLS cfg has the following: ====================== *[server:default]method = TLSv1.2+verify_certificate = yesrequire_certificate = yesprivate_key = /etc/letsencrypt/live/sbc.intelligentsupport.eu/privkey.pem <http://sbc.intelligentsupport.eu/privkey.pem>certificate = /etc/letsencrypt/live/sbc.intelligentsupport.eu/fullchain.pem <http://sbc.intelligentsupport.eu/fullchain.pem>ca_list = /etc/ssl/certs/ca-certificates.crt* *[client:default]method = TLSv1.2+verify_certificate = yesrequire_certificate = yesprivate_key = /etc/letsencrypt/live/sbc.intelligentsupport.eu/privkey.pem <http://sbc.intelligentsupport.eu/privkey.pem>certificate = /etc/letsencrypt/live/sbc.intelligentsupport.eu/fullchain.pem <http://sbc.intelligentsupport.eu/fullchain.pem>ca_list = /etc/ssl/certs/ca-certificates.crt* Thanks very much in advance! Phillip

5 6

Time elapsed since previous message.
by David Villasmil 10 Feb '21

10 Feb '21

Hello all, Is it possible to know the elapsed time since the previously received message? On outgoing calls, I.e: when i get a 180, how long did the 100 arrived? Or the INVITE... Thanks David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 11

Kamailio Split Variable
by Mihai 10 Feb '21

10 Feb '21

Hi! Can I split the variable $fd ? the $fd has the value: asterisk-wind.voicevw.lan; and i want to take the the first part - asterisk-wind any hints ?

4 3

Health Check on Dispatcher Failover
by alirsm 10 Feb '21

10 Feb '21

Hi I have setup failover on Kamailio 5.1 using Dispatcher module as below. On a round robin algorithm, INVITEs are being sent to BOTH healthy and faulty GW. When an INVITE is being sent to the faulty GW, a re-transmission will be sent after "t_set_fr" time to the healthy GW and calls are completed successfully. I was wondering if I can setup the health check correctly so the faulty IP GW can be out of round robin till the health check pass it again, i.e. no INVITE is being sent to the faulty GW as long as the health check not pass it. Your help is greatly appreciated, AL # dispatcher.list 1 sip:X.X.X.X:5060 0 0 1 sip:Y.Y.Y.Y:5060 0 0 # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_threshold", 1) # Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "4")) { send_reply("503", "No destination"); exit; } t_on_failure("DISPATCH_FAILOVER"); t_set_fr(0,1000); route(RELAY); exit; } # Failure Route failure_route[DISPATCH_FAILOVER]{ if (t_is_canceled()) exit; if(!ds_next_dst()) { send_reply("503", "Out of gateways"); exit; } t_on_failure("DISPATCH_FAILOVER"); t_relay(); }

1 0

http_async_client expected result in $http_rr
by Anthony Joseph Messina 10 Feb '21

10 Feb '21

https://www.kamailio.org/docs/modules/stable/modules/http_async_client says $http_rr will be the HTTP reason phrase. I'm understanding this to mean: Code: Reason: 200 OK 404 Not Found In my current 5.4 build, $http_rr returns empty (not <null> or anything else) regardless of what the server sends in the headers. Am I misunderstanding this? I'm writing an API and was hoping to do send_reply("$http_rs", "$http_rr"); -- Anthony - https://messinet.com F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 2

authenticated INVITE badly managed when topos is activated
by frédéric Gaisnon 10 Feb '21

10 Feb '21

Hi, Here is some more information about my problem. I think that topos impacts challenge computing. Do you have the same behaviour I observed? Do you need more information? My tests were done with kamailio 5.4.3 on Centos7 Without topos activated (note that with topoh activated I have the same good behaviour): CPE - INVITE -> SBC CPE <- 407 ---- SBC CPE - INVITE ->SBC (with proxy-authorization header) -- INVITE --> PROXY (So in this case challenge is validated and INVITE is forwarded) With topos activated: CPE - INVITE -> SBC CPE <- 407 ---- SBC CPE - INVITE ->SBC (with proxy-authorization header) CPE <-407 -----SBC topos configuration: loadmodule "ndb_redis.so" loadmodule "topos.so" loadmodule "topos_redis.so" # ----- topos params ----- modparam("topos", "storage", "redis") modparam("topos", "dialog_expire", 15000) Code used: # IP authorization and user authentication route[AUTH] { xlog("L_DBG", "route[AUTH]\n"); #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif #!ifdef WITH_AUTH if ((is_method("REGISTER")) || ($avp(need_auth) == "1")) { ####need_auth is equal to 1 in this case # authenticate requests $var(key)=$fU + "@" + $fd; if($sht(auth_cache=>$var(key))!=$null) { if (!pv_auth_check("$fd", "$sht(auth_cache=>$var(key))", "0", "1")) { auth_challenge("$fd", “1”); #################### we always go here with INVITE with proxy-authorization header and the return code is always -5 (AUTH_NO_CREDENTIALS) exit; } } else { if (!auth_check("$fd", "subscriber", "1")) { if ($rc == -1) { append_to_reply("Retry-After: 10\r\n"); send_reply("503", "Authentication server error"); exit; } auth_challenge("$fd", "0"); exit; } $sht(auth_cache=>$var(key)) = $avp(password); } # user authenticated - remove auth header consume_credentials(); ######## without topos we go here with INVITE with proxy-authorization header } #!endif return; } Note that in this case (with topos) the return code of function pv_auth_check is always -5 (AUTH_NO_CREDENTIALS) CASE OK: Frame 3279: 545 bytes on wire (4360 bits), 545 bytes captured (4360 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.102, Dst: 192.168.1.11 Transmission Control Protocol, Src Port: 5060, Dst Port: 60796, Seq: 1, Ack: 953, Len: 477 Session Initiation Protocol (407) Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bK2df8e195D1847B94;rport=60796;received=192.168.1.11 From: "6200" <sip:6200@entreprise-108.fr>;tag=B583B663-FBFBFCAA To: <sip:0900000000@entreprise-108.fr ;user=phone>;tag=83518db21d5b2e9b777975024049f5a3.8f270000 CSeq: 1 INVITE Call-ID: 9378ee27e6b7aea384a881c938de8138 [Generated Call-ID: 9378ee27e6b7aea384a881c938de8138] Proxy-Authenticate: Digest realm="entreprise-108.fr", nonce="YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO" Authentication Scheme: Digest Realm: "entreprise-108.fr" Nonce Value: "YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO" Content-Length: 0 Frame 3285: 1259 bytes on wire (10072 bits), 1259 bytes captured (10072 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.102 Transmission Control Protocol, Src Port: 60796, Dst Port: 5060, Seq: 1578, Ack: 478, Len: 1191 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:0900000000@entreprise-108.fr;user=phone;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bK827c83577BAADACE From: "6200" <sip:6200@entreprise-108.fr>;tag=B583B663-FBFBFCAA SIP Display info: "6200" SIP from address: sip:6200@entreprise-108.fr SIP from tag: B583B663-FBFBFCAA To: <sip:0900000000@entreprise-108.fr;user=phone> SIP to address: sip:0900000000@entreprise-108.fr;user=phone CSeq: 2 INVITE Call-ID: 9378ee27e6b7aea384a881c938de8138 [Generated Call-ID: 9378ee27e6b7aea384a881c938de8138] Contact: <sip:6200@192.168.1.33;transport=tcp> Contact URI: sip:6200@192.168.1.33;transport=tcp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomVVX-VVX_500-UA/5.7.0.14430 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Supported: replaces,100rel Allow-Events: conference,talk,hold Proxy-Authorization: Digest username="6200", realm=" entreprise-108.fr", nonce="YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO", uri=" sip:0900000000@entreprise-108.fr;user=phone;transport=tcp", response="3e0013cc3dc3855602ce1939af7e6f40", algorithm=MD5 Authentication Scheme: Digest Username: "6200" Realm: "entreprise-108.fr" Nonce Value: "YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO" Authentication URI: "sip:0900000000@entreprise-108.fr ;user=phone;transport=tcp" Digest Authentication Response: "3e0013cc3dc3855602ce1939af7e6f40" Algorithm: MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 270 Message Body Bad case (with topos activated): Frame 9071: 545 bytes on wire (4360 bits), 545 bytes captured (4360 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.102, Dst: 192.168.1.11 Transmission Control Protocol, Src Port: 5060, Dst Port: 43608, Seq: 1, Ack: 953, Len: 477 Session Initiation Protocol (407) Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bK5c0a58f3707458FA;rport=43608;received=192.168.1.11 From: "6200" <sip:6200@entreprise-108.fr>;tag=59191351-FD3B2D60 To: <sip:0900000000@entreprise-108.fr ;user=phone>;tag=83518db21d5b2e9b777975024049f5a3.8f270000 CSeq: 1 INVITE Call-ID: 727c871081e29672abcb8bd05dde8138 [Generated Call-ID: 727c871081e29672abcb8bd05dde8138] Proxy-Authenticate: Digest realm="entreprise-108.fr", nonce="YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/" Authentication Scheme: Digest Realm: "entreprise-108.fr" Nonce Value: "YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/" Content-Length: 0 Frame 9078: 1259 bytes on wire (10072 bits), 1259 bytes captured (10072 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.102 Transmission Control Protocol, Src Port: 43608, Dst Port: 5060, Seq: 1578, Ack: 478, Len: 1191 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:0900000000@entreprise-108.fr;user=phone;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bKbca400a5DCDB8264 From: "6200" <sip:6200@entreprise-108.fr>;tag=59191351-FD3B2D60 SIP Display info: "6200" SIP from address: sip:6200@entreprise-108.fr SIP from tag: 59191351-FD3B2D60 To: <sip:0900000000@entreprise-108.fr;user=phone> SIP to address: sip:0900000000@entreprise-108.fr;user=phone CSeq: 2 INVITE Call-ID: 727c871081e29672abcb8bd05dde8138 [Generated Call-ID: 727c871081e29672abcb8bd05dde8138] Contact: <sip:6200@192.168.1.33;transport=tcp> Contact URI: sip:6200@192.168.1.33;transport=tcp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomVVX-VVX_500-UA/5.7.0.14430 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Supported: replaces,100rel Allow-Events: conference,talk,hold Proxy-Authorization: Digest username="6200", realm=" entreprise-108.fr", nonce="YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/", uri=" sip:0900000000@entreprise-108.fr;user=phone;transport=tcp", response="281d775e7166a96d5efe2e100df3df9a", algorithm=MD5 Authentication Scheme: Digest Username: "6200" Realm: "entreprise-108.fr" Nonce Value: "YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/" Authentication URI: "sip:0900000000@entreprise-108.fr ;user=phone;transport=tcp" Digest Authentication Response: "281d775e7166a96d5efe2e100df3df9a" Algorithm: MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 270 Message Body Regards, Frederic

1 0

CANCEL transaction
by David Villasmil 10 Feb '21

10 Feb '21

Hello, When receiving a reply, there's a logic which if true, i must cancel the outgoing invite (can 480 the origination). I tried using something like t_set_fr(10,10); to have kamailio cancel it immediately, but it's not happening. Is there a way to cancel an INVITE/transaction? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 7

master daily build with disabled compiler optimisation
by Sergey Safarov 10 Feb '21

10 Feb '21

I want suggested to use "-ggdb3 -O0" gcc options when compiled daily master RPM files. Through?

1 0

Permissions module with Mongodb
by H Yavari 09 Feb '21

09 Feb '21

Hi to all, It's mentioned in the Permissions module doc, we can set DB driver:https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#i… So we can use MongoDB as a DB backend? All the modules that use DB, support Mongo? Thanks. Regards,Hossein

4 13

Kamailio as MS-Teams SBC
by Davide Zanichelli 09 Feb '21

09 Feb '21

Hello everybody! i am trying a setup of kamailio like sbc for ms teams, I currently have the following scenario: TLS Connection OK Out Call ok (except BYE 481 on reply) my problem is in incoming calls microsoft only supports srtp and my pbx (freeswitch based) only works with rtp, now i was reading that you need to use rtpengine to manage srtp to rtp, unfortunately it seems that rtpengine has become very difficult to build especially with recent os, I was wondering if it is possible to have an alternative to rtpengine for managing the srtp (rtpproxy does not seem to work) has anyone managed to get a working setup? in case it is possible to send a configuration template? Thanks You !!! Davide

8 26

run_trans_callbacks_internal crash
by Juha Heinanen 09 Feb '21

09 Feb '21

We got the crash below with 5.4.2. Before the crash there was db_mysql_async_exec_task(): failed to execute query related error messages in syslog. -- Juha Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f601d4b2f32 in run_trans_callbacks_internal (cb_lst=0x7f5ffb53e318, type=512, trans=0x7f5ffb53e2a0, params=0x7ffd1c349720) at t_hooks.c:254 254 t_hooks.c: No such file or directory. (gdb) bt full #0 0x00007f601d4b2f32 in run_trans_callbacks_internal (cb_lst=0x7f5ffb53e318, type=512, trans=0x7f5ffb53e2a0, params=0x7ffd1c349720) at t_hooks.c:254 cbp = 0x72743b6d6f632e61 backup_from = 0x560e8abbe2f0 <def_list+16> backup_to = 0x560e8abbe2f8 <def_list+24> backup_dom_from = 0x560e8abbe300 <def_list+32> backup_dom_to = 0x560e8abbe308 <def_list+40> backup_uri_from = 0x560e8abbe2e0 <def_list> backup_uri_to = 0x560e8abbe2e8 <def_list+8> backup_xavps = 0x560e8abbeae0 <_xavp_list_head> backup_xavus = 0x560e8abbeae8 <_xavu_list_head> backup_xavis = 0x560e8abbeaf0 <_xavi_list_head> __func__ = "run_trans_callbacks_internal" #1 0x00007f601d4b35bb in run_trans_callbacks_with_buf (type=512, rbuf=0x7f5ffb53e370, req=0x7f5ffafe3b70, repl=0xffffffffffffffff, flags=0) at t_hooks.c:303 params = {req = 0x7f5ffafe3b70, rpl = 0xffffffffffffffff, param = 0x7f5ffa61e640, code = 487, flags = 0, branch = 0, t_rbuf = 0x7f5ffb53e370, dst = 0x7f5ffb53e3c0, send_buf = {s = 0x7f5ffa6d4d40 "F", len = 621}} trans = 0x7f5ffb53e2a0 #2 0x00007f601d458025 in relay_reply (t=0x7f5ffb53e2a0, p_msg=0xffffffffffffffff, branch=0, msg_status=408, cancel_data=0x7ffd1c349910, do_put_on_wait=0) at t_reply.c:2124 relay = 0 save_clone = 0 buf = 0x7f601e46a4e0 "SIP/2.0 487 Request Terminated\r\nVia: SIP/2.0/TCP 46.182.160.40;branch=z9hG4bK9ef2.1bbf20df35272cd85c43814302770601.2\r\nVia: SIP/2.0/UDP 95.216.75.211;rport=5060;branch=z9hG4bK9ef2.d6265991eb8da9b735273"... res_len = 621 relayed_code = 487 relayed_msg = 0xffffffffffffffff reply_bak = 0x20000000 bm = {to_tag_val = { s = 0x7f601e46a692 "f148edd50b878efc51a72f445b390303-e1110000\r\nCall-ID: c2e2b55c-e574-1239-3cbd-6cb3112351ce\r\nCSeq: 31882977 INVITE\r\nServer: OpenSIPg SIP Proxy (5.4.2-8 (x86_64/linux))\r\nContent-Length: 0\r\n\r\n", len = 41}} totag_retr = 0 reply_status = RPS_COMPLETED uas_rb = 0x7f5ffb53e370 to_tag = 0x7f601d525770 <tm_tag> reason = {s = 0x560e8aad2dbb "Request Terminated", len = 18} onsend_params = {req = 0x7ffd1c349940, rpl = 0x7f5ffaeeb7c0, param = 0x7ffd1c349880, code = 491640932, flags = 32608, branch = 0, t_rbuf = 0x59d, dst = 0x7f5ffb0ea590, send_buf = {s = 0x7ffd1c349940 "p\231\064\034\375\177", len = 491699179}} ip = {af = 1, len = 0, u = {addrl = {140050215200048, 140050209586032}, addr32 = {4216579376, 32607, 4210965360, 32607}, addr16 = {58672, 64339, 32607, 0, 15216, 64254, 32607, 0}, addr = "0\345S\373_\177\000\000p;\376\372_\177\000"}} __func__ = "relay_reply" #3 0x00007f601d49c7b3 in fake_reply (t=0x7f5ffb53e2a0, branch=0, code=408) at timer.c:295 cancel_data = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 517217456}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 517217456}}}} do_cancel_branch = 0 reply_status = 3845367323 #4 0x00007f601d49cc35 in final_response_handler (r_buf=0x7f5ffb53e540, t=0x7f5ffb53e2a0) at timer.c:462 silent = 0 branch_ret = -1 prev_branch = -1 now = 2146494577 #5 0x00007f601d49cd29 in retr_buf_handler (ticks=2146494577, tl=0x7f5ffb53e560, p=0xfa0) at timer.c:518 rbuf = 0x7f5ffb53e540 fr_remainder = 32608 retr_remainder = 473209280 retr_interval = 32765 new_retr_interval_ms = 547787427795 crt_retr_interval_ms = 140725076662752 t = 0x7f5ffb53e2a0 __func__ = "retr_buf_handler" #6 0x0000560e8a7375fa in slow_timer_main () at core/timer.c:1105 n = 12 ret = 0 tl = 0x7f5ffb53e560 i = 610 __func__ = "slow_timer_main"

2 2

Timeout for provisional response other that 100
by David Villasmil 09 Feb '21

09 Feb '21

Hello all, I’m trying to figure out if I can set. Timeout for specific provisional responses. I know I can set timeouts for INV, and restart the timer every time a provisional response is received. But this not what I actually need. What I need is to set a timeout from the 100 Trying to the next 180/183 (or a final response) and set back a 480 if not received. I’m not sure TM would allow me to do that, so maybe using a hash table to store transactions and calculate the time since the INV to the received 180/183. But there’s gotta be a simpler way. Many thanks! David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Best SIP reply for service activation
by Duarte Rocha 09 Feb '21

09 Feb '21

Hello, I'm currently developing a service on Kamailio where it's possible to configure redirections for a subscriber using a code. In order to implement that client should send an INVITE with a specific code on the Request URI and the redirection is configured. What should be the best way to answer this INVITE? A 200 OK will make the call answered and doesn't seem the best way. A negative reply (4/5/6xx) would work, but it seems strange to have the operation concluded with success and answering with a negative reply. I found "199 - Early Dialog Terminated" to be suitable, but the documentation suggests it should be used upstream. Thanks in advance for any suggestion. Cheers

2 1

Kamailio inbound proxy -> Asterisk
by Mihai 09 Feb '21

09 Feb '21

Hi! Don't know if this is a Kamailio Question, maybe someone had deal with it in the past. I use Kamailio with dispatcher to send the request (registration, invites, all) to my asterisk (inbound) and then the asterisk has the outbound trunks, everything works, but this is little thing in asterisk when entering the "sip show peers" it shows me that all the peers are registered with the kamailio IP, if I set the nat option to no, it will show me the correct IP but breaks other clients. Any idea what to do ?

1 0

How to drop 200 OK in onreply_route ?
by Denys Pozniak 09 Feb '21

09 Feb '21

Hello! In the configuration below Kamailio Proxy creates a transaction for the SIP PUBLISH to get info from the HTTP server in async mode. But before creating a transaction, a synthetic 200 OK is sent, so that I need somehow to drop the real 200 OK from the upstream Presence Server. If drop 200 OK in *reply_route*, tm module starts to retransmit. Those it is necessary that the 200 OK be consumed by the tm module, but does not go further. *if ( !is_method("PUBLISH") ) {* * sl_send_reply("200", "Ok"); t_newtran();* * $http_req(suspend) = 1; http_async_query("$var(url)", "CALLBACK");* *}* *route[CALLBACK] {* * <some header manipulation>* * t_on_reply("PUBLISH_REPLY"); route(RELAY); exit;* *}* *onreply_route[PUBLISH_REPLY] { if ( t_check_status("200") ) { drop; # Does not work!!! }}* Any advice is appreciated -- BR, Denys Pozniak

3 8

Kamailio config file for making Kamailio a Sip Proxy
by Derick Greenlee 08 Feb '21

08 Feb '21

Hello, I had some questions regarding the Kamailio config file. I am trying to make Kamailio just behave as a sip proxy between two sip trunks. Could you send me an example for this possibly?

2 1

authenticated INVITE badly managed when topos is activated
by frédéric Gaisnon 08 Feb '21

08 Feb '21

Hi, If topos is activated and kamailio responds with a 407 "proxy authentication required" to an initial INVITE, INVITE received after with credentials (MD5 response in Proxy-Authorization header) is badly managed. Credentials are never validated and Kamailio responds 407 again. If topoh is used instead topos all works fine and credentials are validated. call flow is: Phone sends INVITE to kamailio SBC kamailio respond 407 "proxy authentication required" because SBC wants to authenticate caller Phone resends INVITE with Proxy-Authorization header with all valid information kamailio responds 407 again instead forwarding INVITE. My code: if (!pv_auth_check("$fd", "$sht(auth_cache=>$var(key))", "0", "1")) { auth_challenge("$fd", “1”); exit; } # user authenticated - remove auth header consume_credentials(); So if topos is used, pv_auth_check always returns false even if the phone sets valid information. If topoh is activated, pv_auth_check always returns true (same phone and same kamailio SBC versioning) I made my tests with kamailio 5.4.3 on centos 7. Regards, Frédéric Gaisnon

1 0

Registered with sip user using user pass
by Gaurav Bmotra 08 Feb '21

08 Feb '21

Hi I m using kamailio 5.3 OS Ubuntu 18 I wan to Register on sip server and then send traffic to to it form kamailio. Like : Client ---(sip peer)-->kamailio------(sip peer)---->gateway Please help me Thank you Regards Gaurav Kumar

1 0

filter_body("application/sdp") on multipart/mixed; boundary with double Content-Length
by Karsten Horsmann 08 Feb '21

08 Feb '21

Hi List, I have some weird multipart/mixed;boundary issues. I use kamailio 5.3.6 with rtpengine 7.x.x (same with 8.x.x). If i try to remove a multipart/mixed and leave only the sdp with filter_body() that works except one thing. The sender used IMHO an non RFC compliant version of rfc5621 and wrote the Content-Length Header twice (one in the header, one in the sdp). After manipulating the SDP filter_body() lets Content-Length in the sdp (i guess). And rtpengine is not happy with the sdp after that. rtpengine [rtpengine.c:2588]: rtpp_function_call(): proxy replied with error: Failed to parse SDP Any ideas how to solve this? Kind regards Karsten Horsmann https://tools.ietf.org/html/rfc5621#section-3.1 route[REMOVE_MULTIPART] { if (has_body("multipart/mixed")) { if ($sel(cfg_get.remove.multipart) == 1) { if (filter_body("application/sdp")) { remove_hf("Content-Type"); append_hf("Content-Type: application/sdp\r\n"); msg_apply_changes(); xlog("L_WARN", "[$cfg(route)] Body part multipart/mixed with application/sdp found\n"); } else { xlog("L_ERROR", "[$cfg(route)] Body part multipart/mixed without application/sdp found\n"); } } else { xlog("L_INFO", "[$cfg(route)] Body part multipart/mixed found - manipulation disabled\n"); } } } ------------------- INVITE sip:+49XXXXXXXX@172.20.120.57:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.120.51:5061;branch=z9hG4bK-6589-1-0 From: sipp <sip:sipp@172.20.120.51:5061>;tag=6589SIPpTag001 To: sut <sip:+49XXXXXXXXX@172.20.120.57:5060> Call-ID: 1-6589(a)172.20.120.51 CSeq: 1 INVITE Contact: sip:sipp@172.20.120.51:5061 Max-Forwards: 70 Subject: Performance Test Content-Type: multipart/mixed;boundary=sonus-content-delim Content-Length: 1185 --sonus-content-delim Content-Type: application/sdp Content-Length: 305 v=0 o=Sonus_UAC 895311 862130 IN IP4 8.8.8.8 s=SIP Media Capabilities c=IN IP4 8.8.8.8 t=0 0 m=audio 26088 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:26089 a=ptime:20 --sonus-content-delim Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:cl="urn:ietf:params:xml:ns:pidf:geopriv10:civicLoc" mlns:btd="http://btd.orange-business.com" entity=" pres:geotarget@btip.orange-business.com"><tuple id="XXXX"><status><gp:geopriv><gp:location-info><cl:civicAddress><cl:country>DE</cl:countr <cl:A1>Some State</cl:A1><cl:A3>Somewhere</cl:A3><cl:A4>Somewhere</cl:A4><cl:A6>Somestreet</cl:A6><cl:YYY>1</cl:YYY><cl:PC>9999</cl:PC><cl:EMERGENCYZONE>XXXXXXX</cl:EMERG CYZONE></cl:civicAddress></gp:location-info><gp:usage-rules></gp:usage-rules></gp:geopriv></status></tuple></presence> --sonus-content-delim-- --------------------- INVITE sip:+49XXXXXXXX@foo SIP/2.0 Via: SIP/2.0/TCP 212.XX.XX.XX;branch=z9hG4bKc228.6b8fdf20d79998d1d355feb2dcf5ba63.0 From: sipp <sip:sipp@foo>;tag=6589SIPpTag001 To: sut <sip:+49XXXXXXXX@foo> Call-ID: 1-6589(a)172.20.120.51 CSeq: 1 INVITE Max-Forwards: 69 Content-Length: 347 Content-Type: application/sdp P-Asserted-Identity: <sip:+49XXXXXXXX@212.XX.XX.XX> Contact: <sip:btpsh-60211bbe-674b-1@212.XX.XX.XX;transport=tcp> Content-Length: 305 v=0 o=Sonus_UAC 895311 862130 IN IP4 8.8.8.8 s=SIP Media Capabilities c=IN IP4 8.8.8.8 t=0 0 m=audio 26088 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:26089 a=ptime:20 -- Mit freundlichen Grüßen *Karsten Horsmann*

1 0

Retransmission of 407
by Marat Gareev 08 Feb '21

08 Feb '21

Hello, UAC sends INVITE to UAS, authorization procedure is required, therefore UAS responds with an 407 error code. At this point, the client does not respond with ACK, but the server does not send retransmits. Is this the expected behavior? * UAC -- SIPp, UAS -- Kamailio. Can the Kamailio be configured to send 407 code retransmissions? Thanks!

3 4

TO Header corrupted in RE-INVITE
by alirsm 07 Feb '21

07 Feb '21

Hi, Call to a number forwarded to a FAX by using two Kamailio proxies. (INVITE) --> KamailioA --> GW1 --> KamailioB --> GW2 --> FAX FAX --> (RE-INVITE) --> GW2 --> KamailioB --> (TO Header being corrupted) KamailioB is corrupting TO header when receiving RE-INVITE and causing the call to fail. 1) Only one "uac_replace_from()" is being used in each Kamailio. 2) GW1 is adding +1 (using E164 format) to URI and FROM headers. I was wondering if any of the above might cause the TO header corruption. Any help would be appreciated. AL

2 1

Failover with Dispatcher Module
by alirsm 05 Feb '21

05 Feb '21

Hi I have setup dispatcher failover on Kamailio 5.1 as below. If destination X.X.X.X is down, traffic should be routed to Y.Y.Y.Y. The failover is working but every time after 30 sec Timeout. Is there a way to setup health check properly so the failed destination can be ignored while it is down? Your help is greatly appreciated, AL # dispatcher.list 1 sip:X.X.X.X:5060 0 0 1 sip:Y.Y.Y.Y:5060 0 0 # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_inactive_threshold", 4) modparam("dispatcher", "ds_ping_latency_stats", 1) modparam("dispatcher", "ds_probing_threshold", 1) # Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "4")) { send_reply("503", "No destination"); exit; } t_on_failure("DISPATCH_FAILOVER"); route(RELAY); exit; } # Failure Route failure_route[DISPATCH_FAILOVER]{ if (t_is_canceled()) exit; if(!ds_next_dst()) { send_reply("503", "Out of gateways"); exit; } t_on_failure("DISPATCHER_FAILOVER"); t_relay(); }

2 1

Make configuration as proxy, push, renew server
by James Chan 02 Feb '21

02 Feb '21

Hi We use a Cloud-based SIP-server that does not support the Push-Notifications Service. We try to use Kamailio as the front-end to work as a proxy and push-server. We need help configuring Kamailio as a proxy and push-server. In addition, we want to implement auto renew configuration as UAC using Kamailio. Because all clients need to renew registration our Cloud-SIP-server every 59 minutes. ( registration expires after 60 minutes) Summary: UAC -> Kamailio (proxy & push & auto renew) -> Cloud-SIP-Server -> UAC I have attached our config file and screens And if phone has multiple contacts, there is call bug, call will be drop since there are two incoming call #!KAMAILIO #!define WITH_DEBUG #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_ACCDB #james #!define WITH_UAC #!define WITH_TSILO # # Kamailio (OpenSER) SIP Server v5.3 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # 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 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** 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 block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** 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 ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** 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. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!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 #!ifdef WITH_TSILO #!define FLB_TSILO 8 #!endif ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif fork=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* 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.mydomain.com" /* 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:185.224.139.45:5060 /* 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 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### 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" pstn.gw_port = "" desc "PSTN GW Port" #!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 ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib64/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.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 "acc.so" loadmodule "counters.so" #!ifdef WITH_UAC loadmodule "uac.so" #!endif #!ifdef WITH_TSILO loadmodule "tsilo.so" loadmodule "htable.so" loadmodule "cfgutils.so" loadmodule "http_client.so" #!endif #!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 # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- 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 ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_UAC 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", 1) /* 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$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) modparam("debugger", "log_level_name", "exec") #!endif #!ifdef WITH_UAC modparam("uac", "reg_contact_addr", "pn.telesonique.ch:5060;transport=udp") modparam("uac", "reg_db_url", DBURL) modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 300) modparam("uac", "reg_db_table", "uacreg") #!endif #!ifdef WITH_TSILO modparam("htable", "htable", "vtp=>size=10;autoexpire=180") modparam("http_client", "httpcon", "pushapi=>https://www.myswissline.com") #!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); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # 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 } if (is_method("REGISTER")) { save("location", "0x06"); } # 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); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } #james set_forward_no_connect(); if (!t_relay()) { sl_reply_error(); } #!ifdef WITH_TSILO else if(isflagset(FLB_TSILO)) { resetflag(FLB_TSILO); if (ts_store()) { $sht(tsilo=>stored::$rU) = 1; xdbg("stored transaction [$T(id_index):$T(id_label)] $fU => $rU\n"); } route(PUSH_NOTIF); } #!endif exit; } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies #set_reply_no_connect(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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 route(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")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } save("location", "0x06"); switch($rc) { case -1: case -2: sl_reply_error(); break; #!ifdef WITH_TSILO case 1: $var(new_contact) = 1; #!endif } #!ifdef WITH_TSILO if ($var(new_contact)) route(WAKE_SUSPENDED); #!endif exit; } #!ifdef WITH_TSILO route[WAKE_SUSPENDED] { $var(suspended) = 0; lock("$tU"); $var(suspended) = $sht(tsilo=>suspended::$tU); $sht(tsilo=>suspended::$tU) = $null; $var(stored) = $sht(tsilo=>stored::$tU); unlock("$tU"); if ($var(suspended) == 0) { if ($var(stored)) { # no suspended INVITE (i.e. there was at least one # registered contact for target) and 'stored' set # (i.e. ongoing transaction) -> this newly just # registered contact needs alerting. ts_append("location", "$tu"); } return; } $var(id_index) = $(var(suspended){s.select,0,:}{s.int}); $var(id_label) = $(var(suspended){s.select,1,:}{s.int}); xdbg("resuming trasaction [$var(id_index):$var(id_label)] $tU .\n"); t_continue("$var(id_index)", "$var(id_label)", "RESUME"); } route[RESUME] { if(! lookup("location")) { xlog("failed to load just added destinations\n"); exit; } setflag(FLB_TSILO); route(RELAY); } #!endif # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($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 $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; #!ifdef WITH_TSILO send_reply("100", "Trying"); route(SUSPEND); route(PUSH_NOTIF); #!else route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } #!endif } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } #!ifdef WITH_TSILO setflag(FLB_TSILO); #!endif route(RELAY); exit; } #!ifdef WITH_TSILO route[SUSPEND] { if(! t_suspend()) { xlog("SUSPEND: failed to suspend transaction.\n"); # TODO: actually, the destination is not really checked: don't know if it's # own subscriber or invalid dialing in the first place. send_reply("503", "Temporarily unavailable"); exit; } $sht(tsilo=>suspended::$rU) = $_s($T(id_index):$T(id_label)); xdbg("suspended transaction [$T(id_index):$T(id_label)] $fU => $rU\n"); } route[PUSH_NOTIF] { xdbg("TODO: send push for $rU\n"); $var(res) = http_connect("pushapi", "/send_push.php?username=$tU&dev=1", "application/json", "", "$avp(gurka)"); xlog("L_INFO", "Push notification: $avp(gurka) Result code $var(res)\n"); if ($var(res)==200) { if ($var(gurka)=~"^.*rejected.*$") { sl_send_reply("503", "Not available right now"); } else { sl_send_reply("101", "Push Sent"); } } if ($var(res)==28) { xlog("L_INFO", "Push notification timed out. Do we care?"); } exit; } #!endif # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!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; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_TSILO 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 if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # 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; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal 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; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $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 reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context 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_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { 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")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } [image: call_issue.png][image: register.png]

1 0

failover on SIP servers
by AL RSM 02 Feb '21

02 Feb '21

Hi, I was wondering what is the recommended approach to implement a failover on several destination SIP servers: 1) using DNS SRV records 2) using dispatcher list 3) combination of both. Thanks, AL

3 2

Acc module unable to insert cdr in mysql server
by M Arqum CH 01 Feb '21

01 Feb '21

Hello Members, Kamailio acc module is not inserting cdr into MySQL DB, sometimes it works and sometimes didn't, the number of parameters/values are similar in failed as well success scenario. I also check with tcpdump for MySQL requests, unfortunately, Kamailio didn't issue insert query toward DB server, so it is clear to me that the MySQL server is nothing to do with failed queries. Please guide me on what I should do to find and fix the issue. Kamailio Installed version : kamailio -v version: kamailio 5.4.1 (x86_64/linux) 2ca223 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 2ca223 compiled on 12:51:24 Sep 21 2020 with gcc 6.3.0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:691]: tmcb_func(): acc callback called for t(0x7fe29b2a7870) event type 512, reply code 200 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:413]: should_acc_reply(): probing acc state - code: 200 flags: 0x6 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:446]: should_acc_reply(): acc is on Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 6 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 7 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 10 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 13 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 16 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 17 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 19 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 22 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 26 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 28 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 29 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 32 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 33 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: tm [t_hooks.c:256]: run_trans_callbacks_internal(): DBG: trans=0x7fe29b2a7870, callback type 512, id 0 entered Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 3 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 14 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 18 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 21 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 24 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 25 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: ERROR: acc [acc_cdr.c:267]: db_write_cdr(): failed to insert into database Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 3 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 14 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 18 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 21 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 24 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 25 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: ERROR: acc [acc_cdr.c:682]: cdr_on_end_confirmed(): failed to write cdr! Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: tm [t_reply.c:1758]: cleanup_uac_timers(): RETR/FR timers reset Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103682]: DEBUG: tm [timer.c:634]: wait_handler(): finished transaction: 0x7fe29727ba88 (p:0x7fe294d27288/n:0x7fe294d27288) -- Regards M Arqum

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