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9 participants
37392 discussions

ds_mark_dst( ) in DISPATCHER
by alirsm 11 Feb '21

11 Feb '21

Hi, In my failover setting using Dispatcher module, I mark the GW if fails to response. OPTIONS are being sent every 30 sec: failure_route[DISPATCH_FAILOVER]{ if (t_is_canceled()) exit; if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { ds_mark_dst("ip"); if (ds_next_dst()) { t_set_fr(0,1000); t_on_failure("DISPATCH_FAILOVER"); route(RELAY); exit; } } } However, Kamailio still recognize it as active gw ( A.A.A.A is down, no response from OPTIONS being sent) $ sudo sercmd dispatcher.list { SET_NO: 1 SET: { SET_ID: 1 DEST: { URI: sip:A.A.A.A:5060 FLAGS: AP PRIORITY: 0 ATTRS: } DEST: { URI: sip:B.B.B.B:5060 FLAGS: AP PRIORITY: 0 ATTRS: } } } Any help would be appreciated. AL

1 0

R-R to MS Teams Proxy
by Juha Heinanen 11 Feb '21

11 Feb '21

When request is sent from Kamailio to MS Teams SIP Proxy, the top R-R URI needs to contain FQDN of Kamailio SIP proxy instead of its IP address. Document https://skalatan.de/de/blog/kamailio-sbc-teams suggest to replace record_route(); call with record_route_preset("SBC-DNS-DOMAIN:5061;transport=tls", "SBC-IP-ADDR:5060"); That works only in a very simple case where the request came in over UDP or TCP and SIP Proxy has only one listening address, i.e., SBC-IP-ADDR. One way to solve the problem might be a new r_r function that would take FQDN of the top R-R URI as argument or introduction of a pv from where the current record_route() function would take the FQDN if it has been set. Any comments or other solutions? -- Juha

6 19

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

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