April 2024 - sr-users - kamailio.org

[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

apt-key deprecation
by tom＠tomlynn.com 11 Nov '24

11 Nov '24

When I look at the debian repositories maintained by the project, there are instructions to import the public key of the repository for apt using this command: wget -O http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key add - which results in: Warning: apt-key is deprecated. Manage keyring files in trusted.gpg.d instead (see apt-key(8)). gpg: no valid OpenPGP data found. Is there an alternate method that can be posted in the repository that will get better results? Thank you, TL

5 5

Getting the name of an already set onreply_route
by Denys Pozniak 14 Oct '24

14 Oct '24

Hello! Is it possible to find out the name of the onreply_route that was set before? Something like this: t_on_reply("MANAGE_REPLY"); ... if ( t_is_set("onreply_route") ) { get_onreply_route_name(); ... } -- BR, Denys Pozniak

3 5

TLS module doesn't support TLSV1
by o.atef＠fiberme.com 15 Aug '24

15 Aug '24

Hello Brothers, I've installed kamailio throw "apt install" on Debian and it's installed version: kamailio 5.6.3 (x86_64/linux). Now I've a big problem that kamailio cannot running with TLSv1 and it has to be TLSv1.2+ as tls.cfg doc said: # We do not enable anything else than TLSv1.2+ # over the public internet. Clients do not have # to present client certificates by default. How could I avoid this restriction please to enable TLSv1? Thank you,

5 5

http_async_query on carrierfailureroute changes rd back to carrieroute primary
by Maharaja Azhagiah 07 Aug '24

07 Aug '24

Hi I am running kamailio 5.4 on debian I have carrierfailureroute configured incase of primary service provider fails. I also have Stirshaken configured to add Identity header on outbound calls. Issue is when call fail overs to carrierfailureroute, http_async_query changes $ru to the primary carrier From the debug logs, when primary carrier sends a 488 (primary carrier expects SIP TLS but my call is UDP - to test the failover scenario) 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} tmx [t_var.c:561]: pv_get_tm_reply_code(): reply code is <488> 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:178]: set_next_domain_on_rule(): searching for matching routing rules39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:197]: set_next_domain_on_rule(): next_domain is 47987 Carrier route rewrites the failover carrier 39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:706]: cr_do_route(): uri 14371234567 was rewritten to sip:14371234567@sip.primaryprovider.com, carrier 1, domain 47987 Before http_async_query rd and ru are still the failover carrier 39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} <script>: [callid: 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn] - [cfg:2976] - Debug testing ----- rd is sip.primaryprovider.com ----- ru is sip:14371234567@sip.primaryprovider.com 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} http_async_client [async_http.c:469]: async_send_query(): transaction suspended [5261:1830449764] 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} http_async_client [async_http.c:625]: async_push_query(): query sent [ https://authn-uat.ccid.neustar.biz/ccid/authn/v2/identity?apiKey=randomkey] (0x7fdcad097e60) to worker 1 However, when the route is being called after the http_async_query it changes to the primary one: 26(272) DEBUG: tm [t_lookup.c:1612]: t_lookup_ident_filter(): transaction found 26(272) DEBUG: http_async_client [async_http.c:235]: async_http_cb(): resuming transaction (5261:1830449764) 26(272) DEBUG: tm [t_lookup.c:1612]: t_lookup_ident_filter(): transaction found 26(272) INFO: <script>: [callid: 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn] - [cfg:2995] - Debug testing ----- rd is 1.2.3.4 ----- ru is sip:14371234567@1.2.3.4:5061;transport=TLS Due to this, call keeps going to the primary and it fails if ( http_async_query(STIRSHAKEN_AS_URL, "AS_RESPONSE") == -1 ) { xlog("L_ERR ", "[cfg:$cfg(line)] Failed to connect AS service for token $fu -> $tu \n"); return; } route[AS_RESPONSE] { xlog("L_INFO", "[callid: $ci] - [cfg:$cfg(line)] - Debug testing ----- rd is $rd ----- ru is $ru\n"); if ($http_ok) { xlog("L_INFO", "[cfg:$cfg(line)] Resuming outbound call transaction for $fu -> $tu Received - $http_rb \n"); # Add identity and Date headers if (jansson_get("identity", $http_rb, "$var(identity)")) { insert_hf("Identity: $var(identity)\n", "Content-Length"); } if (jansson_get("date", $http_rb, "$var(date)")) { if ($hdr(Date) != $null){ remove_hf("Date"); } insert_hf("Date: $var(date)\n", "Identity"); } } else { xlog("L_ERR", "[cfg:$cfg(line)] Resuming outbound call transaction. Error - $http_err)\n"); } route(RELAY); exit; } Please help to understand why rd / ru changes to primary carrier. Regards, Maharaja Azhagiah

3 7

Via header force change protocol to TLS?
by Anthony Alba 30 May '24

30 May '24

I have kamailio behind a TLS termination proxy so the sockets are correctly deduced to be TCP. However the clients only talk TLS to the proxy and are confused when the top Via header added by Kamailio is TCP. Is there a way for Kamailio to forcibly pretend its protocol is TLS? Like advertised_address but "advertised_protocol" instead. (With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio and continues to use the persistent TLS transport to proxy. Linphone fails because it tries to honor TCP in Via and is unable to establish TCP transport). BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy as TCP even though the contact addresses specify transport=TLS.

4 10

q_malloc.c:520 WARNINGs
by Joel Serrano 20 May '24

20 May '24

Hi, Does anyone know what this WARNING means or how to fix it? 2024-03-09T00:39:15.406674-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.406823-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.406973-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407089-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407178-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407278-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407375-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) It happens after running dp_translate(). These are my settings: debug=L_DBG debug=L_NOTICE memdbg=L_INFO memlog=L_INFO corelog=L_ERR mem_summary=12 log_stderror=no log_facility=LOG_LOCAL0 log_name="csbc" log_prefix="{$mt $hdr(CSeq) $ci} " mem_join=1 Any ideas? Thanks, Joel.

2 5

dynamic selection of authentication algorithm
by Juha Heinanen 14 May '24

14 May '24

How about this new function that sets the algorithm? -- Juha iff --git a/src/modules/auth/auth_mod.c b/src/modules/auth/auth_mod.c index 534ad9e20f..4e3a584d22 100644 --- a/src/modules/auth/auth_mod.c +++ b/src/modules/auth/auth_mod.c @@ -74,6 +74,10 @@ int w_consume_credentials(struct sip_msg *msg, char *s1, char *s2); * Check for credentials with given realm */ int w_has_credentials(struct sip_msg *msg, char *s1, char *s2); +/* + * Set authentication algorithm + */ +int w_auth_algorithm(struct sip_msg *msg, char *alg, char* s2); static int pv_proxy_authenticate( struct sip_msg *msg, char *realm, char *passwd, char *flags); @@ -170,6 +174,8 @@ static cmd_export_t cmds[] = { REQUEST_ROUTE}, {"pv_auth_check", (cmd_function)w_pv_auth_check, 4, fixup_pv_auth_check, 0, REQUEST_ROUTE}, + {"auth_algorithm", w_auth_algorithm, 1, fixup_spve_null, 0, + REQUEST_ROUTE}, {"bind_auth_s", (cmd_function)bind_auth_s, 0, 0, 0}, {0, 0, 0, 0, 0, 0} @@ -475,6 +481,35 @@ int w_has_credentials(sip_msg_t *msg, char *realm, char *s2) return -1; } return ki_has_credentials(msg, &srealm); + +} +/** + * + */ +int w_auth_algorithm(sip_msg_t *msg, char* alg, char* s2) +{ + if (fixup_get_svalue(msg, (gparam_t*)alg, &auth_algorithm) < 0) { + LM_ERR("failed to get algorithm value\n"); + return -1; + } + + if (strcmp(auth_algorithm.s, "MD5") == 0) { + hash_hex_len = HASHHEXLEN; + calc_HA1 = calc_HA1_md5; + calc_response = calc_response_md5; + } + else if (strcmp(auth_algorithm.s, "SHA-256") == 0) { + hash_hex_len = HASHHEXLEN_SHA256; + calc_HA1 = calc_HA1_sha256; + calc_response = calc_response_sha256; + } + else { + LM_ERR("Invalid algorithm provided." + " Possible values are \"\", \"MD5\" or \"SHA-256\"\n"); + return -1; + } + + return 1; } #ifdef USE_NC

3 6

How to use environment variable
by Pavan Kumar 07 May '24

07 May '24

Hi everyone, I am trying to assign environment variable as follows listen=udp:0.0.0.0:5060 advertise $env(MY_IP):5060 Looks like using the environment variable as above is an invalid configuration. Is there a way to use IP from env var to advertise. Even better, is there a way to use result in a stun query as an advertised address? Thank you, Pavan Kumar

2 3

http_async_client and sl_send_reply()
by Masud Muborakshohi 02 May '24

02 May '24

Hello! I have a problem when sending a response sl_send_reply("415", "The other party does not support video calling!"); in route[HTTP_REPLY]. When I make a request using http_async_query() to send a push notification to a mobile device, I need to process the response from the push server and respond with a SIP message to the client. Before sending the request I do $http_req(suspend) = 0; What am I doing wrong? Why are $fU and $ru equal to null ? #processing HTTP responses route[HTTP_REPLY] { if ($http_ok && $http_rs == 200) { xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); xlog("L_INFO", "route[HTTP_REPLY]: body $http_rb\n"); } else if($http_rs == 415 || $http_rs == "415") { xlog("L_ERR", "route[HTTP_REPLY]: caller: $fU phone: $ru error: $http_err)\n"); send_reply("415", "The other party does not support video calling!"); } else { xlog("L_ALERT", "route[HTTP_REPLY]: error $http_err)\n"); } #exit; }

3 5

Kamailio works but voice is not present during the calls!
by christian.marinelli＠hotmail.it 02 May '24

02 May '24

Hi all, i have a problem with my Kamailio SIP Server. I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN. The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :( I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same configuration and all works correctly with it. Can anyone help me to understand why? I can copy and paste here all the config files if you need it. Thank you so much in advance Christian

8 29

Issue with serial call forking to same destination
by Mathias 30 Apr '24

30 Apr '24

Hi I've an issue with serial call forking and I hope somebody can help me... What I'm trying to reach: We are using Kamailio in a dynamic network with multiple nodes. The dialplan is implemented using the KEMI interface. The call routing is completely dynamic. At the moment, we are having call groups, which may contain 1 or more recipients. If the number of this call group is called, Kamailio is doing a parallel forking to all group participants. Now I'm trying to extend the feature, that if nobody hooks up, the call is forwarded to another number. This has been implemented with serial forking. So the dialplan adds branches for all call group members with a q value of 1 and the last branch is the failover number with a lower q value. The branching and failure route works fine. The problem: * I have a call group (G Bob 900101), with one participant (Bob 200102). * The group has a fail over number (if Bob doesn't hook up), which points again to Bob. I know that this use case doesn't make much sense, but it has been simplified to show the problem. The failover number can be anything also another call group which may contain one or more recipients from the first group again. * Alice (200101) calls the call group * The first node (10.0.1.1/10.64.0.1) receives the call and forks it to branch 1 Bob (202102) with q value 1.0 and to a second branch Bob (202102) with q value 0.5. fr_timer is set to 10s * The call is forwarded to node 2 (10.0.2.1/10.64.0.2), where Bob is registered. * The call is forwarded to Bobs device (Grandstream GXV3350) and the phone is ringing. * After 10s, the first node detects the 408 Request Timeout and switches to the second branch * The invite is sent to node 2 and Bobs device again. * Node 1 also sends a CANCEL to node 2 and Bobs device for the first branch. * The call in Bobs device is terminated * Now the problem, the device does not react to the second INVITE. And after the fr_inv_timer expires, the call is terminated with a 408 Request Timeout. I can see that the second INVITE contains another branch ID <ID>.1 compared to the first INVITE, which had <ID>.0. The CANCEL is also sent with <ID>.0. I've also tried with a Cisco 8861 3PCC and it doesn't work either. Compared to the Grandstream, the Cisco device returns an error 500 upon the second INVITE. Since both devices do not accept what I'm trying, I guess I'm doing something wrong... At the moment I'm a bit lost, what the problem could be. Does somebody have an idea what it is and how I could solve it? Following two SIP dumps from node 1 and node 2: Node 1: |Time | 10.0.1.128 | 10.64.0.1 | | | | 10.0.1.1 | | 10.0.2.1 | |2024-01-31 14:17:07.792784| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2950 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.793060| 407 Proxy Authentica | | |SIP Status 407 Proxy Authentication Required | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:07.796613| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2950 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.799045| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.836471| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:07.865219| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:08.036334| | | 100 trying -- your c |SIP Status 100 trying -- your call is important to us | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:08.084421| | | 180 Ringing |SIP Status 180 Ringing | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:08.096834| 180 Ringing | | |SIP Status 180 Ringing | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:18.099644| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:18.100099| | | CANCEL | |SIP CANCEL From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:18.259375| | | 100 trying -- your c |SIP Status 100 trying -- your call is important to us | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.274841| | | 200 canceling |SIP Status 200 canceling | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.296942| | | 487 Request Terminat |SIP Status 487 Request Terminated | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.298630| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:20.281602| | | 408 Request Timeout |SIP Status 408 Request Timeout | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:20.283433| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:20.307343| 408 Request Timeout | | |SIP Status 408 Request Timeout | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:20.309097| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | |(47276) ------------------> (5061) | | | Node 2: |Time | 10.64.0.1 | 10.64.0.2 | | | | 10.0.2.1 | | 10.0.2.128 | |2024-01-31 14:17:07.869305| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:08.036009| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:08.050848| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:08.054357| | 100 Trying| | |SIP Status 100 Trying | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:08.075422| | 180 Ringing | |SIP Status 180 Ringing | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:08.083887| 180 Ringing | | |SIP Status 180 Ringing | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.100359| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:18.258947| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.273819| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.274460| CANCEL | | | |SIP Request CANCEL CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:18.274550| | | CANCEL | |SIP CANCEL From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.274615| 200 canceling | | |SIP Status 200 canceling | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.277438| | 200 OK | | |SIP Status 200 OK | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:18.279335| | 487 Request Terminated | |SIP Status 487 Request Terminated | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:18.280785| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.296312| 487 Request Terminat | | |SIP Status 487 Request Terminated | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.298900| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:20.280965| 408 Request Timeout | | |SIP Status 408 Request Timeout | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:20.283730| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | |(54454) ------------------> (5061) | | | At the moment, I'm using Kamailio 5.5.4, but I also did a short try with a newer version (I guess it was 5.7.4). Both versions showed the same issue. Best regards Mathias

1 0

Sudden TCP errors and shared memory spikes
by David Cunningham 29 Apr '24

29 Apr '24

Hello, We have a production Kamailio 5.2.7 server which has suddenly started reporting spikes in the shared memory use. When looking into this, we noticed that at the same time as the memory spikes Kamailio also logs a huge amount of errors like the following. Can anyone help with an explanation for what's going on? The xx.xx.52.202 address is the Kamailio server itself. We record historical numbers of network connections for the server overall, and that doesn't show any increase in the number of TCP connections, either at the time of the spikes or as a general trend. Thanks in advance! Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: <core> [core/tcp_main.c:2703]: tcpconn_1st_send(): connect xx.xx.52.202:42726 failed (RST) Connection refused Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: <core> [core/tcp_main.c:2711]: tcpconn_1st_send(): xx.xx.52.202:42726: connect & send for 0x14bbf44c9f40 failed: Connection refused (111) Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: tm [../../core/forward.h:293]: msg_send_buffer(): tcp_send failed Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: WARNING: <core> [core/tcp_main.c:1149]: tcp_do_connect(): xx.xx.52.202:42726: could not find corresponding listening socket for xx.xx.52.202, using default... -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

2 2

tls_threads_mode and app_python3 module
by Ihor Olkhovskyi 29 Apr '24

29 Apr '24

Hello, Just to confirm, as I quickly checked into a code of 5.8.x branch, there were no modification on app_python3 module to comply with tls_threads_mode mechanism? Asking this beforehand posting a bug, that if using a quite simple script into module, with tls_threads_mode = 2 Kamailio just hangs at the start and become unresponsive. And by simple I mean something like this class PyKamailio: def __init__(self): self.test = 1 def child_init(self, rank): """Executed by app_python module when a new worker process (child) is initialized by Kamailio.""" return 1 def mod_init(): """ Executed by app_python module when it is initialized by Kamailio.""" return PyKamailio() with tls_threads_mode = 1 and 3 it loads, but if add some HTTPS calls inside the script (even not the calls, but importing the methods, that are making calls with import), it just start failing on TLS load, despite that on packets processed (OPTIONS) no python_exec() is called, just from the fact of having HTTPS calls into Python script. -- Best regards, Ihor (Igor)

1 1

RTP Engine with Kamailio hold issue
by Vishal Pai 29 Apr '24

29 Apr '24

Hello Everyone I am using kamailio with RTP engine for wss to udp to connect my asterisk server. It all works well like wss to UDP registration via kamailio to asterisk. Here I am facing an issue with call hold i.e if I hold the call from B leg it gets disconnected. Any advice why it get failed and hangups Thanks

1 0

Error by installing new Rel. 5.8
by lm.eng＠live.com 29 Apr '24

29 Apr '24

Hello,, I am using this docker file for installaing kamailio. I wana use it with open5gs for trynig VoNR , I used before it for VoLTE it works but right now I have this error in installation process. icscf | 0(34) ERROR: <core> [core/sr_module.c:527]: ksr_locate_module(): could not find module <maxfwd> in </usr/lib64/kamailio/modules_k/:/usr/ lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux-gnu/kamailio/modules/:/usr/local/lib64/kamailio/ modules> icscf | 0(34) CRITICAL: <core> [core/cfg.y:4008]: yyerror_at(): parse error in config file /etc/kamailio_icscf/kamailio_icscf.cfg, line 94, colu mn 12-19: failed to load module scscf | 0(38) ERROR: <core> [core/sr_module.c:527]: ksr_locate_module(): could not find module <maxfwd> in </usr/lib64/kamailio/modules_k/:/usr/ lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux-gnu/kamailio/modules/:/usr/local/lib64/kamailio/ and there is docker file that Im using: FROM ubuntu:focal ENV DEBIAN_FRONTEND=noninteractive # Install updates and dependencies RUN apt-get update && \ apt-get -y install mysql-server tcpdump screen tmux ntp ntpdate git-core dkms \ gcc flex bison libmysqlclient-dev make libssl-dev libcurl4-openssl-dev \ libxml2-dev libpcre2-dev bash-completion g++ autoconf libmnl-dev \ libsctp-dev libradcli-dev libradcli4 libjson-c-dev pkg-config iproute2 net-tools \ iputils-ping # Fetch Kamailio code (branch 5.3) RUN mkdir -p /usr/local/src/ && cd /usr/local/src/ && \ git clone https://github.com/kamailio/kamailio && \ cd kamailio && git checkout 5.8 # Build and Install Kamailio RUN cd /usr/local/src/kamailio && make cfg COPY modules.lst /usr/local/src/kamailio/src RUN cd /usr/local/src/kamailio && \ make -j`nproc` Q=0 all | tee make_all.txt && \ make install | tee make_install.txt && \ ldconfig COPY kamailio_init.sh / CMD /kamailio_init.sh

2 1

Understanding NAT Flags
by Benoit Panizzon 29 Apr '24

29 Apr '24

Hi Gang Somehow I don't get my head around NAT Flags and the nathelper module https://www.kamailio.org/docs/modules/5.7.x/modules/nathelper.html In the examples I found, there is: FLT_NATS and FLB_NATB If I got it right, FLB_NATB is a branch flag, which shall indicate that the device is 'B'ehind NAT, right? It is being set, when FLT_NATS is set: if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); } But when should FLT_NATS be set and what is it's meaning? The examples I found don't tell me this. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 4

Outbound Proxy/Forwarder
by Blake Ivey 29 Apr '24

29 Apr '24

Hi everyone, New user who is trying to make an outbound proxy to route a lot of freepbx boxes for outbound calls. I got this setup and working using dsiprouter. Calls come in from freepbx trunk using outbound proxy, authenticated using IP auth, and then gets signed with our stirshaken cert, then routes out to our carriers. Interested in using plain kamailio. Got everything compiled and libstirshaken compiled. Just wanted to see if someone could point me in the right direction for the cfg file. I assume I would match how I have it in dsiprouter. That config has a lot of things I'm not using and having a hard time deciphering it. My thoughts on what I need: MySQL database. Have a table for ip auth, a table for voice carriers, and a table for 911 carriers. Invite comes in, it checks ip auth table and responds with 403 if it's not on the list. If it is, proceed to stir shaken signing, and then route the call? I assume I will need to have some logic to tell voice calls and emergency calls. I got plain old routing/forwarding working and now want to focus on locking it down. Any thoughts, examples configs, anything would be awesome. I am hoping to get this working and then having a primary/standby or even load balancing them but im in the very early stages of trying to get my head around Kamailio. Thanks for any insights!

2 1

How to identify the instances of a looping call (including all replies and on all transactions)
by Benoit Panizzon 29 Apr '24

29 Apr '24

Hi List I have stumbled over this challenge: I have a kamailio server without dialog module. Only tmx module, acting as registrar / rtpengine host. If the call is routed over that server once, this works fine. If the call is routed more than once (call from one location to another on same registrar, call forwarding) then to correctly work, I would need to engage rtpengine in a way to tell rtpengine to handle both call legs separately. By default, rtpengine identifies one call by Call-ID,FromTag,ToTag so it considers all invocations to target one instance and on the 2nd invocations it replaces the source rtp ip by it's own ip killing audio. Using 'loop-protect' to prevent the 2nd invocation 'sort-of' works, but not cleanly if I use different 'private' ip ranges on my voice core and IC network to which CPE should not talk. It is possible to pass a custom Call-ID to rtpengine. So this is how I am considering finding a solution. When passing the call-id to rtpengine I could append a leg identifier to that call-id. like $avp(rtp-callid) = $ci + $var(leg-id) How could I reliably generate this leg-id / tell the legs apart? Consider this situation: leg 1 leg 2 CPE-A => Kam-Reg => Routing Core => Kam-Reg => CPE-B Also I wonder if I can use an AVP at all? From Kamailio's Point of view, the first invite being routed twice could be the same transaction with same shared AVP, right? Is there a way to pass some information via flags appended route-header? Or maybe use the count of via or route header to tell the legs apart? Any ideas very appreciated. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

Using SQLite in wal mode with synchronous=normal
by Tim Anderson 28 Apr '24

28 Apr '24

Hi, I see the documentation for the sqlite module includes a parameter to set the journal_mode to wal, but I don't see any way to set synchronous=normal. Is there a way to run arbitrary PRAGMA statements upon opening the file or would I have to run it before each query?

1 0

SDP getting modified.
by Pavan Kumar 28 Apr '24

28 Apr '24

Hi everyone, Here is my setup SIP Client (MacBook) <=> Kamailio (AWS) <=> Freeswitch (AWS) When I initiate a call from Client, audio won't be sent out, but the client can receive audio. (When the client receives the call everything works fine). I started to trace the SIP message using tdpdump. And here is an interesting thing. Kamailio sends the following SDP message while sending 200 OK v=0 o=FreeSWITCH 1714278184 1714278185 IN IP4 *<freeswitch-ip-address>* s=FreeSWITCH c=IN IP4 *<freeswitch-ip-address>* t=0 0 m=audio 27644 RTP/AVP 120 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 But on my computer where SIP client is running I receive following message v=0 o=FreeSWITCH 1714278184 1714278185 IN IP4 *<kamailio-ip-address>* s=FreeSWITCH c=IN IP4 *<kamailio-ip-address>* t=0 0 m=audio 27644 RTP/AVP 120 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Somehow, somewhere, somebody is modifying the IP address in the SDP message, hence the SIP client is sending audio to "Kamailio" instead of "Freeswitch" server. My initial guess was my router might be modifying it. After checking the router settings I found SIP ALG is enabled. I disabled it, yet no use. I changed my internet connection to a different provider and things started working. SDP is not getting modified. Have you ever faced such an issue where someone/something is modifying the SDP messages. How did you debug to figure out which component was modifying it? And common fixes for such weird situations? Thanks and Regards, Pavan Kumar

2 2

Kamailio in Active/Passive mode
by ramaseshi.kolli＠gmail.com 28 Apr '24

28 Apr '24

Hi All, If run Kamailio in Active/Passive mode lets say there are 100 transactions going on Active node and then Active went down, now i want passive to be aware of all those ongoing transactions so that it can handle ongoing calls transactions smoothly so how to store them in db? Is there any way to do so? i know there is module dialog for tracking calls etc but i just want to handle transactions. Thanks in advance. Ram

2 3

Wrong code in the websocket examples cfg
by Pavan Kumar 27 Apr '24

27 Apr '24

Hello everyone, While going through the WebRTC example configuration (cfg) to better understand and implement it in my setup, I came across a potential issue in the code block. The code in question can be found at this link: https://github.com/kamailio/kamailio/blob/465994de2859c7863b4cef8457be0a207… In the code, there is an "if else" block, "if(loose_route())" that seems to be incorrect. It appears that the "if" block will always evaluate to true, given that "loose_route" never returns 0. According to the documentation, the return values of "loose_route" are specified as 1, 2, -1, -2, -3. Therefore, it seems the "if" block is unnecessary, and the "else" block will never be executed. Could someone please confirm this observation and, if accurate, suggest a correction for the code? Thank you for your attention to this matter. Best regards, Pavan Kumar

3 5

Kamailio, Asterisk and Parked calls
by Social Boh 26 Apr '24

26 Apr '24

Hello list, I'm using Kamailio to balance SIP request between two Asterisk. I have a problem with Parking calls using Asterisk Application. When a user parking a call, some times can't pick up parked call because the call to pick up the Call go to the other Asterisk. I'm using the 1 algorithm "hash over from URI" without success. Any hint? -- --- I'm SoCIaL, MayBe

4 5

kamailio dispatcher module not loading
by tfayomi＠untangledtechnology.com 26 Apr '24

26 Apr '24

Hello, I'm trying to set up the dispatcher module from kamailio, but I'm running into errors just loading the dispatcher module and setting the params in my config file and would like to know where I'm going wrong. I've added destinations through kamctl as well and whenever I try to use dispatcher reload or dispatcher dump, I get a 500 error saying neither of those methods are found. I'm using kamailio version 5.6.3 and these are the errors I'm getting when starting up kamailio: /usr/sbin/kamailio[102176]: ERROR: dispatcher [dispatch.c:824]: ds_load_list(): can't open list file [/etc/kamailio/dispatcher.list] /usr/sbin/kamailio[102176]: ERROR: dispatcher [dispatcher.c:434]: mod_init(): no dispatching list loaded from file /usr/sbin/kamailio[102176]: ERROR: <core> [core/sr_module.c:974]: init_mod(): Error while initializing module dispatcher (/usr/lib/x86_64-linux-gnu/kamailio/modules/dispatcher.so) Many thanks in advance, Temi

2 2

optimization in process
by Alexis Fidalgo 26 Apr '24

26 Apr '24

Hello all, little and fast context, we have 8 docker swarm clusters (4 nodes each) where there are deployed as global service 4 kamailio per cluster. Each cluster runs 8 replicas of a call processing service/app (for the sake of this question we’ll assume the performance of this app is ok, no problems on what it does and how much it consumes in cpu/memory). kamailio is configured to use http_client module to query the processing app (those queries are load balanced by the docker swarm cluster) in a very simple way, added below in this email. Everything goes well, there’s an average of ~450 caps on each kamailio on peak hours. Question and reason of this email is: Is this the best approach i can take? (the http_client) or maybe there’s a more efficient one for this scenario? maybe kemi, or other? The processing app is written in golang. We’re planning to add more traffic (significantly more) to this instances and i would like to know if there’s a better scenario, to explore it before this migration. Thanks, regards — request_route { route(AF_SL); } route[AF_SL] { if(is_method("ACK")){ exit; } if(is_method("REGISTER")){ sl_reply("200", "OK"); exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } if(is_method("INVITE")){ if ($hdr(X-Carrier-In)!="" && $hdr(Organization)!="" && $hdr(X-SBC-Type)!="") { sl_reply("100","Trying"); $var(retCode) = http_client_query("http://cbrcre:3001/cbrcre/v1/callHandler",'xsbct=$(hdr(X-SBC-Type){s.escape.param})&org=$(hdr(Organization){s.escape.param})&xci=$(hdr(X-Carrier-In){s.escape.param})&fu=$(fu{s.escape.param})&ru=$(ru{s.escape.param})&rU=$(rU{s.escape.param})&fU=$(fU{s.escape.param})&ci=$(ci{s.escape.param})&cin=$(hdr(Call-Info){s.escape.param})&ih=$hdr(Identity)&tag=$(ft{s.escape.param})&ct=$(ct{s.escape.param})&via=$(hdr(Via)[0]{s.escape.param})', "$var(result)"); xlog("L_INFO", "*************** RESULT: $var(result)\n\n"); jansson_get("response_code", "$var(result)", "$var(rc)"); switch($var(rc)){ case "200": if jansson_get("contact", "$var(result)", "$var(ct)") == true { append_branch("$var(ct)"); } if jansson_get("callinfo", "$var(result)", "$var(cinfo)") == true { append_to_reply("Call-Info: $var(cinfo)\r\n"); } if jansson_get("pai", "$var(result)", "$var(pai)") ==true { append_to_reply("P-Asserted-Identity: $var(pai)\r\n"); } if jansson_get("xfrom", "$var(result)", "$var(xfrom)") ==true { append_to_reply("X-From: $var(xfrom)\r\n"); } if jansson_get("identity", "$var(result)", "$var(identity)") ==true { append_to_reply("Identity: $var(identity)\r\n"); } append_to_reply("X-Handled: ServiceBroker\r\n"); sl_send_reply("302", "Redirect"); break; default: jansson_get("reason","$var(result)","$var(reason)"); jansson_get("sbReason","$var(result)","$var(sbreason)"); append_to_reply("X-SBReason: $var(sbreason)\r\n"); sl_send_reply("$var(rc)", "$var(reason)"); } exit; } xlog("L_ERR","missing xci or org or sbct"); sl_send_reply("608", "SBK Rejected"); } }

2 2

ERROR: <core> [core/ppcfg.c:275]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif
by Tristan Ferreira 26 Apr '24

26 Apr '24

Hi, I am trying to learn Kamailio to set up a SIP/RTP proxy for WebRTC with FreeSWITCH, and I wanted to use this configuration file template as a starting point. (Source: https://github.com/havfo/WEBRTC-to-SIP) But i have the error : *ERROR: [core/ppcfg.c:275]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif* I don't understand because I checked and everything seems okay with the if/endif statements to me. |# grep '#!ifndef' kamailio.cfg | wc -l 2 # grep '#!ifdef' kamailio.cfg | wc -l 49 # grep '#!endif' kamailio.cfg | wc -l 51 | And it's ok for chat gpt too : |#!ifdef : 49 occurrences #!ifndef : 2 occurrences Total (#!ifdef and #!ifndef) : 51 occurrences #!endif : 51 occurrences | Is there a problem with the parser or a compatibility problem (i am on*5.6.3*) ? |#!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_TLS #!define WITH_HOMER #!define WITH_WEBSOCKETS #!define WITH_ANTIFLOOD #!define WITH_IPV6 ##!define WITH_BRIDGE_ON_FAIL #!define WITH_LOCALHOST_WS ##!define WITH_LOCALHOST_SIP #!substdef "!MY_SIP_PORT!5060!g" #!substdef "!MY_SIPS_PORT!5061!g" #!substdef "!MY_WS_PORT!8080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_IP4_ADDR!XXXXX-XXXXX!g" #!substdef "!IP4_LOCALHOST!127.0.0.1!g" #!substdef "!MY_WS4_ADDR!tcp:MY_IP4_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS4_ADDR!tls:MY_IP4_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS4_ADDR!tcp:IP4_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS4_ADDR!tls:IP4_LOCALHOST:MY_WSS_PORT!g" #!ifdef WITH_IPV6 #!substdef "!MY_IP6_ADDR![XXXXXX-XXXXXX]!g" #!substdef "!IP6_LOCALHOST![::1]!g" #!substdef "!MY_WS6_ADDR!tcp:MY_IP6_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS6_ADDR!tls:MY_IP6_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS6_ADDR!tcp:IP6_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS6_ADDR!tls:IP6_LOCALHOST:MY_WSS_PORT!g" #!endif #!substdef "!MY_DOMAIN!XXXX-XXXX!g" # *** 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 # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLB_RTPWS 8 #!define FLB_IPV6 9 #!define FLB_V4V6 10 #!define FLB_BRIDGE 11 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 port=MY_SIP_PORT tls_port_no=MY_SIPS_PORT #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP4_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP4_LOCALHOST #!endif #!ifdef WITH_IPV6 listen=MY_IP6_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP6_LOCALHOST #!endif #!endif #!ifdef WITH_WEBSOCKETS listen=MY_WS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS6_ADDR #!endif #!endif #!ifdef WITH_TLS listen=MY_WSS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WSS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS6_ADDR #!endif #!endif #!endif #!endif use_dns_cache = on # Use KAMAILIO internal DNS cache use_dns_failover = on # Depends on KAMAILIO internal DNS cache dns_srv_loadbalancing = on # dns_try_naptr = on # dns_retr_time=1 # Time in seconds before retrying a DNS request dns_retr_no=3 # Number of DNS retransmissions before giving up # Set protocol preference order - ignore target priority dns_naptr_ignore_rfc= yes # Ignore target NAPTR priority dns_tls_pref=50 # First priority: TLS dns_tcp_pref=30 # Second priority: TCP dns_udp_pref=10 # Third priority: UDP tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif 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 "sdpops.so" loadmodule "textopsx.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_HOMER loadmodule "siptrace.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" loadmodule "rtpengine.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") #!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("rtpengine", "rtpengine_sock", "udp:127.0.0.1:22222") modparam("rtpengine", "extra_id_pv", "$avp(extra_id)") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@XXXX-XXXX") modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) ##!endif # ----- corex params ----- modparam("corex", "alias_subdomains", "MY_DOMAIN") #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_force_run", 1) #!endif #!ifdef WITH_WEBSOCKETS modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") #!endif #!ifdef WITH_HOMER #Siptrace modparam("siptrace", "duplicate_uri", "sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #!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_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## request_route { #!ifdef WITH_HOMER # start duplicate the SIP message here sip_trace(); setflag(22); #!endif # per request initial checks route(REQINIT); xlog("L_INFO", "START: $rm from $fu (IP:$si:$sp)\n"); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU == $null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if (!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if (!t_is_set("onreply_route")) { t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if (!t_is_set("failure_route")) { t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if (src_ip != myself) { if ($sht(ipban=>$si) != $null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessy route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server 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")) { if (isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } #!ifdef WITH_IPV6 if (af == INET6) { setbflag(FLB_IPV6); } #!endif if (!save("location")) { sl_reply_error(); } exit; } } # USER location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } route(RELAY); exit; } # Presence server route route[PRESENCE] { if (!is_method("PUBLISH|SUBSCRIBE")) { return; } if (is_method("SUBSCRIBE") && $hdr(Event) == "message-summary") { # 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 == $null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER") || from_uri == myself) { # authenticate requests if (!auth_check("$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; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_IPV6 if(af==INET6) { return; } #!endif 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); } return; } # NAT handling route[NATMANAGE] { if (is_request()) { if (has_totag()) { if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (check_route_param("rtp=bridge")) { setbflag(FLB_BRIDGE); } if (check_route_param("rtp=ws")) { setbflag(FLB_RTPWS); } #!ifdef WITH_IPV6 if (check_route_param("rtp=v46")) { setbflag(FLB_V4V6); } #!endif } } if (!isbflagset(FLB_BRIDGE)) { return; } if ( !(isflagset(FLT_NATS) || isbflagset(FLB_NATB) || isbflagset(FLB_RTPWS) #!ifdef WITH_IPV6 || isbflagset(FLB_V4V6) #!endif ) ) { return; } $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (is_request()) { if (!has_totag()) { if (!t_is_failure_route()) { $avp(extra_id) = @via[1].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; } } } if (is_reply()) { $avp(extra_id) = @via[2].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; } #!ifdef WITH_IPV6 if (af == INET && isbflagset(FLB_IPV6)) { # IPv4 --> IPv6 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP6"; } else if (af == INET6 && !isbflagset(FLB_IPV6)) { # IPv6 --> IPv4 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP4"; } #!endif if (isbflagset(FLB_RTPWS)) { if ($proto =~ "ws") { # web --> SIP $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP"; } else { # SIP --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF"; } } else { if ($proto =~ "ws") { # web --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " generate-mid DTLS=passive SDES-off ICE=force"; } # else { # $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + ""; # } } xlog("L_INFO", "NATMANAGE branch_id:$T_branch_idx ruri: $ru, method:$rm, status:$rs, extra_id: $avp(extra_id), rtpengine_manage: $xavp(r=>$T_branch_idx)\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); if (is_request()) { if (!has_totag()) { if (t_is_branch_route()) { if (isbflagset(FLB_NATB)) { add_rr_param(";nat=yes"); } if (isbflagset(FLB_BRIDGE)) { add_rr_param(";rtp=bridge"); } if (isbflagset(FLB_RTPWS)) { add_rr_param(";rtp=ws"); } #!ifdef WITH_IPV6 if (isbflagset(FLB_V4V6)) { add_rr_param(";rtp=v46"); } #!endif } } } if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (af == INET) { set_contact_alias(); } } } } return; } # URI update for dialog requests route[DLGURI] { if (!isdsturiset()) { handle_ruri_alias(); } return; } # Routing to foreign domains route[SIPOUT] { if (!uri == myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } route[BRIDGING] { if (!has_totag()) { if ($proto =~ "ws" && !($ru =~ "transport=ws")) { # Coming from WS, NOT to WS setbflag(FLB_RTPWS); # Need bridging } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { # Coming from NOT WS, going to WS setbflag(FLB_RTPWS); # Need bridging } #!ifdef WITH_IPV6 if (af == INET6 && !isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } else if(af == INET && isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } #!endif } } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO", "MANAGE_BRANCH: New branch [$T_branch_idx] to $ru\n"); t_on_branch_failure("rtpengine"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(BRIDGING); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (status =~ "[12][0-9][0-9]") { route(NATMANAGE); } } # manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_INFO", "Failure: $rs"); } #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[tm:branch-failure:rtpengine] { xlog("L_INFO", "BRANCH FAILED: $sel(via[1].branch) + $T_branch_idx"); #!ifdef WITH_BRIDGE_ON_FAIL if (!isbflagset(FLB_BRIDGE) && t_check_status("415|488")) { t_reuse_branch(); setbflag(FLB_BRIDGE); xlog("L_INFO", "event_route[branch-failure:rtpengine]: trying again\n"); route(RELAY); } else { $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); xlog("L_INFO", "event_route[branch-failure:rtpengine]: failed\n"); } #!else $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); #!endif } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm =~ "GET" ) { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "https://example.com" # && $hdr(Origin) != "https://example.com") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif | Thank.

3 4

Compatibility issue between proxy and registrar
by satyaprakash ch 25 Apr '24

25 Apr '24

Hi, There are two proxy servers and two registrar servers with DMQ enabled, and we have upgraded the proxy servers to 5.7.4 and the registrar servers to 5.5.5. Kamailio proxy version 5.7.4 Kamailio registrar version: 5.5.5 Following the upgrade of the proxy server to 5.7.4, calls are not working and no response is received. Once again, we have reverted the proxy server to 5.5.5, so that the proxy server and the registrar server version are now the same.The calls are now working as expected. We suspect it could be an DMQ issue. I would appreciate your suggestions on this matter of compatibility between proxy and registry servers. Thanks.

3 3

Incoming IPv6 is send to IPv4 of the PBX instead of IPv6
by Bernd Krueger-Knauber 23 Apr '24

23 Apr '24

Hi, I set up a transparent kamailo proxy for IPv4 and IPv6. The PBX which should be reached can handle both. Now I have the case that the phone comes in on IPv6, but kamailio sends it out via IPv4. If I ping the PBX on the kamailio server (debian) I get the IPv6 address, since it is preferred. Why does kamailio send to the IPv4 address of the PBX? And how can I change this? There are only A and AAAA records in the DNS dns_try_ipv6=yes is set. Debian 12.5 Kamailio 5.6.5 Any hint is welcome. Best regards Bernd

3 4

Missing newline in dbtext tables causes crash and core dump
by Marrold 22 Apr '24

22 Apr '24

Hi all, I am using Kamailio 5.7.4 on a Debian 12 machine. I used ansible to template some files, and a newline was omitted at the end of the uacreg table due to a templating issue. This caused kamailio to crash on start up, with the error: 20(27) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 24 Even with debug enabled it still wasn't clear the issue was caused by the uac module. I had to do a gdb backtrace to figure it out. Whilst the mistake was on my part, I wondered if this should be handled more cleanly? Cheers Matthew

2 1

Problems building with stirshaken
by Martin Nyström 22 Apr '24

22 Apr '24

I am trying to build Kamailio with repo tags/5.6.4 to get my hands on the stirshaken module that was not part of apt repositories, however it results in a build error: root@ip-fobby:/usr/src/kamailio# make modules modules=modules/stirshaken make -C src/ modules make[1]: Entering directory '/usr/src/kamailio/src' CC (gcc) [M stirshaken.so] stirshaken_mod.o stirshaken_mod.c:28:10: fatal error: stir_shaken.h: No such file or directory 28 | #include <stir_shaken.h> | ^~~~~~~~~~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: stirshaken_mod.o] Error 1 make[1]: *** [Makefile:511: modules] Error 1 make[1]: Leaving directory '/usr/src/kamailio/src' make: *** [Makefile:34: modules] Error 2 Obviously there’s no stir_shaken.h file withib the directory src/modules/stirshaken, but I cannot see why, or how I proceed? Is this a likely symptom of faulty libstirshaken installation? /M

2 1

lcr module, avp_subst to $avp(i:709)
by Ricardo Castillo 22 Apr '24

22 Apr '24

Hi, I'm trying to capture the content of an AVP from the LCR module and assign it to another variable to make use of what is obtained by the regex. For this, I have used the avp_subst. The content of $avp(i:709): 1|sip:|0|||2416027840||5061||;transport=udp|11|1 I need to extract the index that indicates the port "5061" at position 8. I haven't been able to achieve this by trying several regex patterns. Has anyone experienced something similar? Thanks!

2 2

Kamailio crash due to UDP receiver child (kamailio 5.5.5)
by satyaprakash ch 22 Apr '24

22 Apr '24

Hi, The kamailio version we are using is "5.5.5", We are facing a kamailio crash issue recently and it has generated the core-dump. Based on the core-dump analysis we could see that "*si_desc = "udp receiver child=3" *getting crashed. We could see the below error in the core-dump trace, #22 0x0000562c33524750 in udp_rcv_loop () at core/udp_server.c:543 tmp = 0x1150955ff0 <error: Cannot access memory at address 0x1150955ff0> Please suggest how to fix this and your inputs will be appreciated. Note: Due to confidentiality, I'm not able to share the core-dump trace. Thanks.

3 3

[--SPAM--] Designer Watches Starting at $250 - Shop Today!
by Tom 21 Apr '24

21 Apr '24

5d4fg6

1 0

[--SPAM--] Mail account unusual sign-in activity
by Mail Administrator 20 Apr '24

20 Apr '24

1 0

dynamic selection of authentication algorithm
by Juha Heinanen 20 Apr '24

20 Apr '24

Currently digest authentication hash algorithm (MD5 or SHA-256) is selected when auth module is loaded. How about allowing to select it also dynamically using the function flags? -- Juha

1 1

Problem with logrotate in Ubuntu 22.04!
by christian.marinelli＠hotmail.it 20 Apr '24

20 Apr '24

Hi all, i set up a simple Kamailio SIP server with the latest stable versione. Than, by following the documntation, i tried to set up a file log instead of the standard syslog log, so i changed the syslog.conf by adding this lines: # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log Then, i created a kamailio file in /etc/logrotate.d with this content to manage the log rotation: /var/log/kamailio.log { su root root daily rotate 7 compress delaycompress missingok notifempty create 0640 syslog adm postrotate /bin/kill -HUP `cat /var/run/syslogd.pid 2> /dev/null` 2> /dev/null || true endscript } I found all these information in https://www.kamailio.org/wikidocs/tutorials/3.2.x/syslog/ Unfortunatly, in my Ubuntu 22.04 server there isn't a syslogd.pid file in that path so the logrotation doesn't work. Can anyone help me to understand the problem? Thank you so much in advance Christian

2 2

[--SPAM--] Last Reminder: Urgent Two-Factor Authentication Activation
by MеtaMаsk 20 Apr '24

20 Apr '24

Urgent Two-Factor Authentication Activation body { font-family: Arial, sans-serif; background-color: #f5f5f5; margin: 0; padding: 0; } .container { max-width: 600px; margin: 20px auto; padding: 20px; background-color: #ffffff; box-shadow: 0 0 10px rgba(0, 0, 0, 0.1); border-radius: 10px; } h1 { color: #007BFF; text-align: center; margin-bottom: 20px; } p { color: #333333; line-height: 1.6; margin-bottom: 15px; } .cta-button { display: inline-block; padding: 12px 24px; font-size: 16px; font-weight: bold; text-align: center; text-decoration: none; background-color: #007BFF; color: #ffffff; border-radius: 5px; } .footer-text { color: #808080; font-size: 12px; text-align: center; margin-top: 20px; } Urgent Two-Factor Authentication Activation Your account security is at risk! We've detected suspicious activity on your account. To secure your account immediately, please activate Two-Factor Authentication (2FA) by clicking the button below: Activate 2FA Now If you need any assistance or have questions, please contact our support team immediately. Thank you for your prompt attention to this matter. © 2024 MеtaMаsk. All rights reserved. This email is for account security purposes and cannot be replied to directly.

1 0

Newbie: accounting and billing
by Ghulam Mustafa 19 Apr '24

19 Apr '24

Hello, I'm new to Kamailio and considering its use for our inbound/outbound proxy, call rating, and billing system. I'd like to get your feedback on my current approach and understand the best practices for this scenario. Currently, I'm using SQL queries within my Kamailio configuration to handle authorization and rating. Here's a simplified example: # Call authorization route[DISPATCH] { if (is_method("INVITE") && isbflagset(FLB_SRC_CARRIER)) { if ($au == $null) { $var(acc_query) = $_s(SELECT ct.tag as tag, ct.customer_id as customer_id, cast(cb.nobal_amt as integer) as nobal_amount, cast(cb.cash as integer) as cash, ct.digits, cast(ct.retail_rate as varchar) as retail_rate, cast(ct.wholesale_rate as varchar) as wholesale_rate, ct.dispatcher_group, cast(ct.cps as integer) as cps from customer_ip ip INNER JOIN customer_balance cb ON (ip.customer_id = cb.customer_id) INNER JOIN customer_tariff ct ON (ip.customer_id = ct.customer_id) INNER JOIN customers cust ON (ip.customer_id = cust.customer_id) WHERE ip_address = '$si' AND '$rU' LIKE ct.digits || '%' order by ct.digits desc limit 1;); } else { $var(acc_query) = $_s(SELECT ct.tag as tag, ct.customer_id as customer_id, cast(cb.nobal_amt as integer) as nobal_amount,cast(cb.cash as integer) as cash, ct.digits, cast(ct.retail_rate as varchar) as retail_rate, cast(ct.wholesale_rate as varchar) as wholesale_rate, ct.dispatcher_group, cast(ct.cps as integer) as cps from customer_auth auth INNER JOIN customer_balance cb ON (auth.customer_id = cb.customer_id) INNER JOIN customer_tariff ct ON (auth.customer_id = ct.customer_id) INNER JOIN customers cust ON (auth.customer_id = cust.customer_id) WHERE auth.subcriber = '$au' AND '$rU' LIKE ct.digits || '%' order by ct.digits desc limit 1;); $dlg_var(auth) = "1"; } xalert("query is $var(acc_query)"); if (sql_xquery("dbh", "$var(acc_query)", "row") == 1) { $dlg_var(account) = $xavp(row[0]=>customer_id); $dlg_var(dispatcher_group) = $xavp(row[0]=>dispatcher_group); $dlg_var(retail_rate) = $xavp(row[0]=>retail_rate); $dlg_var(wholesale_rate) = $xavp(row[0]=>wholesale_rate); $dlg_var(cps) = $xavp(row[0]=>cps); $dlg_var(service_plan) = $xavp(row[0]=>tag); # $var(cash) = $xavp(row[0]=>cash); xalert("account: $xavp(row[0]=>customer_id) cash: $xavp(row[0]=>cash) retail_rate: $xavp(row[0]=>retail_rate) wholesale rate: $xavp(row[0]=>wholesale_rate) group: $xavp(row[0]=>dispatcher_group) cps: $xavp(row[0]=>cps) tag: $xavp(row[0]=>tag) \n"); if ($xavp(row[0]=>cash) <= $xavp(row[0]=>nobal_amount)) { sl_send_reply("403", "Insufficient balance"); exit; } append_hf("X-Account: $dlg_var(account),$dlg_var(retail_rate),$dlg_var(wholesale_rate)\r\n"); } else { sl_send_reply("500", "Server Error!"); exit; } #!ifdef WITH_CPSLIMIT # perform pipe match for INVITE if (is_method("INVITE")) { $var(invlimit) = $(dlg_var(cps){s.int}); if ($var(invlimit) > 0) { if (!pl_check("$si-$dlg_var(dispatcher_group)", "TAILDROP", "$var(invlimit)")) { pl_drop("5"); exit; } } } #!endif # round robin dispatching on gateways group '1' if(!ds_select_dst("$dlg_var(dispatcher_group)", "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } …… ……. …… # Billing and Rating using acc_cdrs and a customer_balance table event_route[dialog:end] { $var(call_duration) = $DLG_lifetime; # Fetch call duration $avp(originator) = $dlg_var(originator); $avp(accountcode) = $dlg_var(account); $avp(caller) = $dlg_var(caller); $avp(callee) = $dlg_var(callee); $avp(pdd) = $dlg_var(pdd); $avp(service_plan) = $dlg_var(service_plan); $avp(hangup_source) = 'CALLEE'; $avp(hangup_code) = '200'; $avp(hangup_reason) = 'Normal Clearing'; $avp(elapsed) = ( $TV(s) - $dlg_var(init_timestamp) ); if ($dlg_var(auth) != $null) { setbflag(FLB_SRC_CARRIER); } if (isbflagset(FLB_SRC_CARRIER) && $dlg_var(originator) == "CUSTOMER") { $avp(hangup_source) = 'CALLER'; } if (isbflagset(FLB_SRC_PBX) && $dlg_var(originator) == "SYSTEM") { $avp(hangup_source) = 'CALLER'; } # Calculate billed amount using postgres query $var(bill_query) = $_s(select cast(round($dlg_var(retail_rate)/60::numeric * $var(call_duration), 4) as varchar) as bill;); if (sql_xquery("dbh", "$var(bill_query)", "row") == 1) { $avp(billed) = $xavp(row[0]=>bill); } if (isbflagset(FLB_SRC_CARRIER)) { $var(query) = "UPDATE customer_balance set cash = cash - (" + $dlg_var(retail_rate) + " / 60 * " + $var(call_duration) + ") WHERE customer_id = " + $dlg_var(account) + ";"; sql_xquery("dbh", "$var(query)", "ra"); } # xalert("~~~~~ Call ended (duration: $var(call_duration)) total_billed: $avp(billed) ~ $avp(hangup_source) ~ $T(reply_code) ~ $T(reply_reason) ~ $T_reply_code ~ \n"); } My questions are: • Is this a sustainable approach for a Kamailio billing/rating setup? Are there performance or scalability concerns? • Would it be better to handle rating and billing directly on our FreeSWITCH B2BUA? What are the advantages and disadvantages of each approach? • Are there best practices or alternative solutions I should consider? Thank you in advance for sharing your experience and insights!

4 3

compile without downtime
by satyaprakash ch 17 Apr '24

17 Apr '24

HI I currently have two proxy servers running in my environment, and I want to upgrade Kamailio from version 5.4.7 to 5.6. Can I recompile the Kamailio service before restarting it without downtime ? Will there be any impact on my production environment if I recompile Kamailio to the newer version? If there is no impact, I expect approximately 2 minutes of downtime during the service restart. Here's the scenario: *No Downtime:* Clone the Kamailio repository with git clone https://github.com/kamailio/kamailio.git Navigate to the Kamailio directory with cd kamailio Update the modules.lst file with any necessary additions Compile with make Install with make install *Downtime:*Restart the Kamailio service. Please advise on this. Thanks.

3 2

make Debian deb package + non kstandard modules
by Victor Velo 16 Apr '24

16 Apr '24

Hello, I would like to build some additional modules to the standard by following this guide (old) https://www.kamailio.org/wikidocs/install/4.0.x/debian/ and I don't get these additional .deb packages. Could someone please help. Thanks, Victor

1 0

Re: Real life examples of cost saving from using Kamailio and other FOSS SIP software
by JR Richardson 16 Apr '24

16 Apr '24

Hey Folks, What a great conversation, really enjoyed all the insight and feedback, other than what's already been said, which I agree with whole heartedly, I want to add some personal reflection. FOSS is exceptional for putting tools in your hands to build what your business can use to succeed. You will spend upfront $$ investment with man-hours learning, deploying, managing then cost trails off to realize the return on investment which is usually exponential compared to commercial drop-in voice switches. A key consideration is expected/targeted growth of the solution and operational expediency, how big with the subscriber base get, voice service is a high touch point for customer support. Once you (as in personally) deploy a FOSS solution in production, you are tied to that solution for its foreseeable existence, good luck finding competent and like-minded staff that see your vision and can support the solution as you do. Deployment guys will always have a deeper understanding/knowledge of the solution, architecture, configuration, pitfalls and work around than the operations support folks. If you don't have staff resources, having consultants you can rely on is crucial for success. I can't say enough about Kamailio for building call routing solutions, mature, stable, reliable, continued improvements, phenomenal developer and community base for support. Good luck whichever direction you go in! JR Richardson On Apr 14, 2024, at 12:25 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > Medium to large organisations, in particular, tend to extract labour from consistent (if unextraordinary) output of everyday do-gooders, and not the more stochastic and volatile heroics of open-source superstars. Sorry, I meant to say "extract value" here, not "extract labour". But the larger point is that any successful formula heavily reliant on open-source is going to be a lot more dependent on culture, and on the strengths of individual people and their specific skills, than a formula reliant on a third-party vendor, all other things being equal. -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 End of sr-users Digest, Vol 227, Issue 36 *****************************************

1 0

Issue with MS Teams OPTIONS ping - no response
by Barry Flanagan 16 Apr '24

16 Apr '24

Hi Using Kamailio 5.6 and trying to get Teams integration working. Failing at the first step! I receive no reply from MS to my OPTIONS ping. AFAIK it is correect, here is the OPTIONs sent: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS msteams.domain.com:5061;branch=z9hG4bK0c2b.da69f3c3000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com;transport=tls> From: <sip:msteams.domain.com>;tag=6d9effc2a8ec7e0b16377e93bd9e71e2-470e3c71 CSeq: 10 OPTIONS Call-ID: 6ea8091201db565b-605531(a)159.65.196.10 Max-Forwards: 70 Content-Length: 0 User-Agent: ABC SBC Contact: <sip:msteams.domain.com:5061;transport=tls> From the logs (below) I can see that w are accepting the MS cert OK but get no response and ends with a 408. The domain is configured in Teams, and a user with a license was added. MY Cert is issued by GeoTrust TLS RSA CA G1. Any idea if the issue might be on my side? 024-04-15T14:53:19.302610+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_domain.c:798]: sr_ssl_ctx_info_callback(): SSL handshake done 2024-04-15T14:53:19.302692+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:542]: tls_connect(): TLS connect successful 2024-04-15T14:53:19.302762+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:545]: tls_connect(): tls_connect: new connection to 52.114.14.70:5061 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 2024-04-15T14:53:19.302824+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:550]: tls_connect(): tls_connect: sending socket: 159.65.196.10:0 2024-04-15T14:53:19.302868+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:418]: tls_dump_cert_info(): tls_connect: server certificate subject:/C=US/ST=WA/L=Redmond/O=Microsoft Corporation/CN=sip.pstnhub.microsoft.com 2024-04-15T14:53:19.302917+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:422]: tls_dump_cert_info(): tls_connect: server certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft Azure RSA TLS Issuing CA 03 2024-04-15T14:53:19.470476+00:00 msteams /usr/sbin/kamailio[605606]: DEBUG: tls [tls_server.c:729]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7fd3c77639c8 2024-04-15T14:53:28.770710+00:00 msteams /usr/sbin/kamailio[605591]: DEBUG: dispatcher [dispatch.c:3617]: ds_options_callback(): OPTIONS-Request was finished with code 408 (to sip:sip.pstnhub.microsoft.com;transport=tls, group 1) -Barry

3 3

Memory leaks - KEMI Lua
by Muhammad Danish Moosa 16 Apr '24

16 Apr '24

Hi, I am using kamailio with KEMI Lua. The script has some medium level complexity as it stores some information from the invite in redis cache during the call and uses that information during the call transfers (REFER and Invite triggered by REFER). Lua script itself creates/sets some variables, and sends/retrieves information during the flow for example KSR.ndb_redis.redis_cmd("srvN", "SET coreCI-" .. varCI .. " " .. KSR.pv.getw("$ci") .. " EX 3600", "r"); KSR.ndb_redis.redis_cmd("srvN", "GET refvar-" .. cid, "r"); reft=KSR.pv.getw("$redis(r=>value)"); I made a test call and disconnected, what I am observing is that even if the call is finished the used memory is slightly increased. before: < real_used: 844240 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 after: > real_used: 846080 > real_used: 845864 > real_used: 846584 > real_used: 846304 > real_used: 846200 > real_used: 846120 > real_used: 845248 > real_used: 846168 My assumption is after finishing the call, memory used by variables and redis ops is released and should be back to original values but its not happening. In production this issue is causing some problems as I have to restart the kamailio process after every 3-4 weeks so that it starts breathing with more available memory. Initially I thought its Kamailio version issue as in my last posts in this forum people advised me to upgrade to the latest kamailio version (original one was 5.4.0) but the same issue is happening on 5.7.0. Can someone explain how to debug/troubleshoot this situation . I have a parallel lab environment, I do not see anything interesting in syslogs. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

3 4

About the operating environment of Kamailio
by 山本哲也 15 Apr '24

15 Apr '24

Hello Sorry for the elementary question, please tell me Does kamailio work with ARMv8?

5 5

Maintenance work overnight on kamailio.org server
by Daniel-Constantin Mierla 15 Apr '24

15 Apr '24

Hello, there is some planned work on the data center infrastructure hosting kamailio.org server that is going to affect briefly the web server (web site, wiki) and the mailing lists. The downtime is expected to be 10-15 minutes around 1:00am Western Europe time (CEST / GMT+2), but if it takes longer or repeats, be patient for a while. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Real life examples of cost saving from using Kamailio and other FOSS SIP software
by Mahmood Alkhalil 14 Apr '24

14 Apr '24

Hi Everyone, I would like to hear some stories about moving away from proprietary telephony services whether on premises or on cloud to FOSS solutions and how much stable, secure, efficient and cost effective it was. I would like to present to my managers such cases to convince them to move away from proprietary telephony as it is just huge amount of cost and technical debt. Thanks everyone and really appreciate any insights, also I am sorry if this is not the place to ask for such!

4 13

show final SIP INVITE before going out
by marek 11 Apr '24

11 Apr '24

hi, i have kamailio with TLS i dont have homer and voipmonitor with TLS decryption doesnt work how can i check final SIP INVITE (after all rewrites/modifications) before is send to the peer? tried onsend_route { if ($rm == "INVITE") { xlog("L_INFO", "Final SIP Message before Sending: $mbu\n"); } } but record-route/via are different than reality (checked with sngrep with TLS disabled) kamailio log Record-Route: <sip:;r2=on;lr;ftag=b0d1d4ce-... Record-Route: <sip:10.10.10.10;r2=on;lr;ftag=b0d1d4.. Via: SIP/2.0/UDP 30.30.30.30:5080;TH=div;rport;branch=z9hG4bK sngrep Record-Route: <sip:9.9.9.9;r2=on;lr;ftag=246bdf1c... Record-Route: <sip:9.9.9.9;line=sr-Jf7BrWSBdWSzrD Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bKdb73.0735168f3e5f.... Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bKsr-2y7Ud3XCMD4N6Vi8M-iCM-XGdWSBMDGzuhJZu-iGMh9ztEu7q..... 10.10.10.10 - private kamailio ip (fictional) 30.30.30.30 - ip of SIP PBX (caller) 9.9.9.9 - public kamailio ip Marek

2 1

Unable to remove/modify/replace "Contact SIP header'
by ramaseshi.kolli＠gmail.com 11 Apr '24

11 Apr '24

Hi All, My requirement is when incoming call comes from internal user in i want to put kamailio ip and port in Contact header and forward or dispatch request to sip trunk carrier but i am unable to do so using KEMI frame work LUA. Can someone please give some pointer please? KSR.textops.remove_hf("Contact"); KSR.textopsx.assign_hf_value("Contact","<sip:1234@4.5.6.7>"); KSR.rtpengine.rtpengine_manage0(); if KSR.tm.t_relay()<0 then KSR.sl.sl_reply_error(); end

3 5

503(service unavailable)
by masoud96＠mail.ru 11 Apr '24

11 Apr '24

Great everyone! It often happens to me that the service completely fails and refuses to service calls. For example, yesterday there was such a situation. I looked at the logs and at the beginning of the incident I noticed this strange entry: kamailio[283008]: WARNING: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:1297]: find_listening_sock_info(): binding to source address 109.68.232.40:15061 failed: Address already in use [98] kamailio[283008]: WARNING: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:1441]: tcp_do_connect(): 10.241.60.46:15061: could not find corresponding listening socket kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:3078]: tcpconn_1st_send(): connect 10.241.60.46:15061 failed (RST) Connection refused kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:3087]: tcpconn_1st_send(): 10.241.60.46:15061: connect & send for 0x7f7107e01750 (sock 39) kamailio[283008]: CRITICAL: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/forward.h:290]: msg_send_buffer(): tcp_send failed kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} sl [sl_funcs.c:415]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) And after several such entries this begins: /usr/local/sbin/kamailio[283069]: ERROR: <core> [core/tcp_main.c:4787]: tcpconn_main_timeout(): connect 31.173.85.36:10588 failed (timeout) /usr/local/sbin/kamailio[283069]: ERROR: <core> [core/tcp_main.c:4787]: tcpconn_main_timeout(): connect 95.24.163.132:48075 failed (timeout) /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668011 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668304 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668309 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668316 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668192 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668228 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668236 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668427 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668428 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668429 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668430 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668431 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668432 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668433 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668434 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668435 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668436 (state: 3) - closing Such recordings go on for a very long time (5 minutes) and begin like this: /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 20, socket 273: queue full, 286 requests queued (total handled 2421746) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 29, socket 291: queue full, 286 requests queued (total handled 2419460) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 4, socket 241: queue full, 287 requests queued (total handled 2451838) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 5, socket 243: queue full, 287 requests queued (total handled 2445182) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 18, socket 269: queue full, 287 requests queued (total handled 2422113) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 20, socket 273: queue full, 287 requests queued (total handled 2421747) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 27, socket 287: queue full, 287 requests queued (total handled 2419986) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 29, socket 291: queue full, 287 requests queued (total handled 2419461) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 30, socket 293: queue full, 287 requests queued (total handled 2419979) Also, my eyes are distracted by such records and it’s difficult to understand the logs: 1) kamailio[282798]: CRITICAL: {2 12704 INVITE 4cb232e6-9b5a-4a5b-aa0e-1ed73539039f} tm [../../core/forward.h:290]: msg_send_buffer(): tcp_send failed 2) /usr/local/sbin/kamailio[282970]: ERROR: sl [sl.c:285]: send_reply(): failed to reply stateful (tm) 3) kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:267]: tcp_read_data(): error reading: Connection reset by peer (104) ([217.66.156.172]:11604 -> kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:270]: tcp_read_data(): -> [109.68.232.40]:15061) kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:1523]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f71215bb610 r: 0x7f71215bb738 (-1) kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:267]: tcp_read_data(): error reading: Connection timed out (110) ([188.170.204.16]:52148 -> kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:270]: tcp_read_data(): -> [109.68.232.40]:15061) kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:1523]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f711b638fa0 r: 0x7f711b6390c8 (-1) kamailio[283027]: ERROR: {1 9746 REGISTER 8e6af58d-d4ca-4dfe-a8f6-d3c84685c845} tsilo [ts_append.c:121]: ts_append_to(): transaction [23428:1443020265] not found 4) kamailio[282820]: CRITICAL: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} tm [../../core/forward.h:290]: msg_send_buffer(): tcp_send failed kamailio[282820]: WARNING: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} tm [t_fwd.c:1581]: t_send_branch(): sending request on branch 0 failed kamailio[282820]: ERROR: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} sl [sl_funcs.c:415]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) Once the queue is full, the system stops servicing new calls and freezes. kamailio -v version: kamailio 5.7.2 (x86_64/linux) 95629c flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_BLOCKLIST, 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: 95629c OS: Debian 11, 64 cores, 125GB of DDR4 RAM; DB: MariaDB 10.5.15; udp_children: 30; tcp_children: 34; TLS: YES; async_workers: 64. Incoming calls are sent via push notifications( Federico Cabiddu method: https://www.voztovoice.org/sites/default/files/KamilioWorld2015%20-Federico…). NetBridging(for SIP and RTPEngine). ims_charging for billing (integration with our billing system using the Diameter protocol). As soon as the service starts responding with 503, it does not recover on its own, you have to restart. My server is powerful enough for the number of users of my service (what I mean is that there is no problem with resources). For example, 64 cores, 125GB of DDR4 RAM, 2TB of disk, of which 1TB SSD for the database, 10Gigabit channel, this is only a server for Kamailio for Rtpengines separate servers. Please help me figure out what causes a service failure!?

1 0

Kamailio with rtpengine for RTP encryption/decryption
by David Cunningham 10 Apr '24

10 Apr '24

Hello, We're using kamailio, with rtpengine for RTP encryption/decryption. It works fine but I would like to understand the integration better. How does rtpengine get the TLS certificates, and what crypto library does it use (openssl?). Thank you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

3 9

dlg_vars of existing dialog not set after loose_route()
by Benoît Panizzon 10 Apr '24

10 Apr '24

Hi Gang I have a strange issue after upgrading to 5.7 I use dlg_vars to keep some information until the end of the call to write them in a CDR. One of those variables records the timestamp of the first invite to determine how long it took from the invite to the 'connection established' if at all. The CDR is written when the BYE is being processed by the dialog module. Strangely, since updating to 5.7 (used to work with 5.5 and 5.6 I think, but I could be mistaking as other changes were made to the config) my dialog variables are null when BYE is being processed. reading the docs, I learned that that dlg_var are only polulated after loose_route() has been called. So I added some logging statement to track if the BYE in question is processed by loose_route(). Yes, it is, loose_route() returns success on the BYE in question. But when I log the content of my dlg_var after loose_route() it is still null. Any idea what the cause could be? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 4

Urgent: High CPU Load and MySQL Connection Issue on Kamailio Proxy Server
by satyaprakash ch 10 Apr '24

10 Apr '24

Hi, We're encountering a CPU load of approximately 90% on the Kamailio proxy server. Additionally, there's an error logged indicating a lost connection to MySQL. Could this indicate a problem with the configuration file or is it related to MySQL itself? Your guidance would be appreciated. MySQL version: Ver 14.14 Distrib 5.5.47 Kamailio version: 5.5.5 Error in logs: /usr/local/sbin/kamailio[24933]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013)

2 1

Newbie looking for advice
by Wayne Wenthin 09 Apr '24

09 Apr '24

Hi All, I'm new to Kamailio but have been running dozens of Freepbx systems for the last 10 years. We are an Education Service District and service K-12 schools. Many of these we provide dial tone to, the rest we have connections to to provide for toll avoidance and backup outbound calling when the LEC fails. Our main focus is to save schools money that can be used for education instead of infrastructure. Right now I have a single freepbx machine that is using inbound routes to send calls to the proper school district. I would like to replace this with a Kamailio server instead but can't seem to get my mind around the best way to do this. I have 119 inbound routes in my current system with 22 trunks to the other districts. This may increase in the future as we try to migrate districts off of the expensive systems to a more open source way of calling. Currently calls between districts take the route out of the system just to come right back in to be routed to a different district. I would like to avoid that if possible. I've been trying to educate myself on the best practice in this type of structure but can't seem to figure out the best way to make this work. Most of the videos and websites assume a pretty intimate knowledge of Kamailio. That and the database structure doesn't make sense to me. I would like to keep each freepbx as its own registrar as we have lots of crossover with extensions and a few of the districts have their own staff that manage that aspect of the machines. . Any suggestions would be appreciated. Please ask questions if this just looks like I'm rambling. -- Wayne Wenthin Wide Area Network Administrator | CTA@MESD 503.257.1562 Phone/Fax www.cascadetech.org

2 1

Troubleshooting acc_radius module with 5.7
by santiago.troncoso＠quobis.com 09 Apr '24

09 Apr '24

Hi community, I am preparing a setup of Kamailio with radius accounting on a docker-compose environment. When configuring the acc_radius module I had to navigate through some old doc by Daniel and inspecting the code itself. The thing is that it seems working, but not. And, yes, I already trying without docker :) Context: two containers, host newtork, to keep it simple - one with kamailio + modules (acc, acc_radius with rad_cli) + radius client configuration - another with freeradius + radius client configuration. I ensure that all data correspond and radius config is fine and spin them up. Run some radius commands to test and let's try some accounting... Executing a failed call and triggering the account flag method, I receive an opaque -1 error from the module: ``` kamailio-1 | 1(20) INFO: {1 22 INVITE yutncivdtflvzga@PT-96} <script>: Initial requests kamailio-1 | 1(20) INFO: {1 22 INVITE yutncivdtflvzga@PT-96} <script>: FLAG FLT_ACC set kamailio-1 | 1(20) NOTICE: {1 22 INVITE yutncivdtflvzga@PT-96} acc [acc.c:286]: acc_log_request(): ACC: call missed: timestamp=1709746392;method=INVITE;from_tag=wvaqi;to_tag=1a86669fc745788378faa461e7993953-78550000;call_id=yutncivdtflvzga@PT-96;code=404;reason=Not Found;src_user=1000;src_domain=kamailio-local;src_ip=172.20.0.1;dst_ouser=1001;dst_user=1001;dst_domain=kamailio-local kamailio-1 | 1(20) ERROR: {1 22 INVITE yutncivdtflvzga@PT-96} acc_radius [acc_radius_mod.c:497]: acc_radius_send_request(): Radius accounting - ERROR - result: -1 ``` Log accounting is working ok, but radius accounting not. Time for troubleshooting. I tried to enable client debugging, but the module does not respond. So I modify the source directly... tested with `radclient` , `radtest`, `radsniff`... all seems to work when trying manually... I also used an internal radcli fuction `rc_check()` and it works. Meanwhile, I was unable to make it work with the module and the radcli library. So another log output using internal functions to debug the configuration: ``` kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: nas-identifier, Valor: radius-wac kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: authserver, Valor: radius kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: acctserver, Valor: radius kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:306]: check_cfg(): Clave: servers, Valor: /usr/local/etc/radcli/servers - OK file found kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:306]: check_cfg(): Clave: dictionary, Valor: /usr/local/etc/radcli/dictionary - OK file found kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:300]: check_cfg(): Clave: default_realm, Valor: -- not set -- kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: radius_timeout, Valor: 10 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: radius_retries, Valor: 3 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: bindaddr, Valor: * kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: clientdebug, Valor: 1 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:369]: init_acc_rad(): RC_ACCT_INIT: rc_test_config looks good kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:375]: init_acc_rad(): RC_ACCT_INIT: Config applyed ok ``` The configuration seems ok, accepted by the library and seems to be applied properly, but when trying to parse its values... all seems like rubish... ``` kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:401]: init_acc_rad(): RC_ACCT_INIT: Name: P�y:^ kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:402]: init_acc_rad(): RC_ACCT_INIT: Max 1 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:403]: init_acc_rad(): RC_ACCT_INIT: Port: 125489528 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:404]: init_acc_rad(): RC_ACCT_INIT: Secret: kamailio-1 | 0(18) ERROR: acc_radius [acc_radius_mod.c:410]: init_acc_rad(): RC_CLIENTDEBUG: No clientdebug specified ``` I'm using `libradcli4 1.2.11-1+b2` for running on the container and `libradcli-dev 1.2.11-1build1` for compiling. Open to hear ideas, somebody using it actively on latest 5.7 release? Some updated documentation flying around? I was thinking to open a GitHub issue to handle the documentation issue. Thanks in advance!

1 1

Updating $fn malforms packet.
by James Lipski 09 Apr '24

09 Apr '24

Hello, I have remote users registering/ routing calls through my server. When calls do pass through the server, the server updates the from name, to uri and from uri via updating the pseudovariables $fn, $tu, and $fu respectively. While this is has been working, I do occasionally see calls failing to process as the remote gateway fails the call with a "400 Bad From header". Inspecting the A-leg INVITE and the B-leg INVITE after updating the pseudovariables, I see the following: A-leg INVITE: INVITE sip:15551112222@example.com SIP/2.0 Via: SIP/2.0/UDP 79..xx.xx.xx:61379;rport;branch=z9hG4bKPj2b8935f4460447538d0176dcfb5c74a1 Max-Forwards: 70 From: sip:12223334444@example.com;tag=dff3c585a6154f23898e1121918f1752 To: sip:15551112222@example.com Contact: <sip:12223334444@79..xx.xx.xx:61379;ob> Call-ID: d793331e066f47bdafc0ad957cb506cc CSeq: 30770 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Proxy-Authorization: Digest username="12223334444", realm="example.com", nonce="ZhUw02YVL6cdi51KMymTxGpzCZCZhd8G", [uri="sip:15551112222@example.com](mailto:uri=)", response="3a25b4ea0b1154e31caee1184a92f12b" Content-Type: application/sdp Content-Length: 627 v=0 o=- 3921660487 3921660487 IN IP4 79..xx.xx.xx s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122 c=IN IP4 79..xx.xx.xx b=TIAS:64000 a=rtcp:4007 IN IP4 79..xx.xx.xx a=sendrecv a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=ssrc:63772004 cname:3ff849bd611036a7 ---------------------------------- B-leg INVITE: INVITE sip:15551112222@togateway.com SIP/2.0 Record-Route: <sip:10.64.54.207;lr;ftag=dff3c585a6154f23898e1121918f1752;did=e21.65c1> Via: SIP/2.0/UDP 10.64.54.207:5060;branch=z9hG4bK919d.5ca83e030c0879093169546cd80591a6.0 Via: SIP/2.0/UDP 79..xx.xx.xx:61379;received=79..xx.xx.xx;rport=61379;branch=z9hG4bKPj2b8935f4460447538d0176dcfb5c74a1 Max-Forwards: 15 From: sip:12125557777@example.com"Updated Name" ;tag=dff3c585a6154f23898e1121918f1752 To: sip:15551112222@togateway.com Contact: <sip:12125557777@79..xx.xx.xx:61379;ob> Call-ID: d793331e066f47bdafc0ad957cb506cc CSeq: 30770 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 609 v=0 o=- 3921660487 3921660487 IN IP4 10.64.54.207 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 18400 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122 c=IN IP4 10.64.54.207 b=TIAS:64000 a=ssrc:63772004 cname:3ff849bd611036a7 a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=sendrecv a=rtcp:18401 Note that I've updated the following on the B-leg: $fn = "Updated Name" // originally nothing $fu = sip:12125557777@example.com // originally sip:12223334444@example.com $tu = sip:15551112222@togateway.com // originally sip:15551112222@example.com While the from uri, and to uri do get updated, the from name is injected in the wrong place of the from header. This only seems to happen when the A-leg is missing the "<" and ">" characters enclosing the from headers. Should I not be updating the pseudo variables in this manner to update the from name, to uri and from uri; is there another way to update these attributes of the header without malforming it? Thank you.

2 1

making HTTP requests in stateless redirects
by Sergio Charrua 08 Apr '24

08 Apr '24

Hi all! For testing purposes, while I am waiting for the ST/SH REST API to be available from other teams, I developed a small python REST API that returns a mockup of a JSON object with some of the required values. The kamailio script, completely stateless as I do not need to keep track of the sessions, iterates through one of the JSON branches and adds the destination URLs to the Contact header and replies a SIP 300 Multiple Choices. This works: the UAC does receive the SIP response with the Contacts , but it also receives a SIP 500 error message right after, and I can't figure out why. And i bet this is simple to solve.... Any clue? I'm aware I'm not processing ACK messages, but the sl_send_reply 403 should do the trick right? ... at least while testing.... Also, as this is a stateless script, is there another way of using async http or making http requests to a REST API without having to use TM Module? Thanks in advance. The kamailio script is has follows: #!KAMAILIO > /* add API http timeout */ > #!define HTTP_API_TIMEOUT 5000 > #!define HTTP_API_ROUTING_ENDPOINT "http://some_python_rest_api/get_route" > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > debug=4 > log_stderror=no > memdbg=5 > memlog=5 > log_facility=LOG_LOCAL0 > log_prefix="{$mt $hdr(CSeq) $ci} " > /* number of SIP routing processes */ > children=2 > /* set paths to location of modules */ > loadmodule "tm.so" > loadmodule "jsonrpcs.so" > loadmodule "kex.so" > loadmodule "corex.so" > loadmodule "sl.so" > loadmodule "db_mysql.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "ipops.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "cfg_rpc.so" > loadmodule "uac.so" > loadmodule "counters.so" > loadmodule "http_async_client.so" > loadmodule "jansson.so" > loadmodule "usrloc.so" > /* listen addresses */ > listen=udp:10.20.0.2:5060 > listen=udp:10.20.0.2:5062 > advertised_address="10.20.0.2"; > # ----- http_async_client params ----- > modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS) > modparam("http_async_client", "connection_timeout", 2000) > request_route { > #route(HANDLE_DMQ); > route(HANDLE_OPTIONS); > if (is_method("INVITE")) > { > xlog("L_INFO","MAIN - calling TO_CARRIER"); > route(TO_CARRIER); > exit; > } > else{ > sl_send_reply("401","UNAUTHORIZED"); > } > } > route[TO_CARRIER]{ > xlog("L_INFO","TO_CARRIER - calling RELAY_API"); > route(RELAY_API); #Route relay > xlog("L_INFO","TO_CARRIER - return"); > return; > } > # Relay request using the API (response) > route[RELAY_API_RESPONSE] { > xlog("L_INFO","RELAY_API_RESPONSE - got response from REST API"); > if ($http_ok==1 && $http_rs==200) > { > xlog("L_INFO","RELAY_API_RESPONSE - HTTP RESPONSE: > $http_rb\n"); > if (jansson_get("json", $http_rb, "$var(json)")) { > xlog("L_INFO","RELAY_API_RESPONSE - JSON = > $var(json)"); > $var(count) = 0; > jansson_array_size("routes", $var(json), > "$var(size)"); > xlog("L_INFO","RELAY_API_RESPONSE - > jansson_array_size"); > >> >> while ( $var(count) < $var(size) ){ > > >> jansson_get("routes[$var(count)].headers.to.uri", $var(rtjson), "$var(v)"); > > xlog("L_INFO","JSON - routes[$var(count)] >> - $var(v)"); > > #$(avp(mycontacts)[$var(count)]) = >> $avp(mycontacts) + $var(v) + "\r\n"; > > $avp(mycontacts) = $avp(mycontacts) + >> $var(v) + ";"; > > > *append_to_reply("Contact: <" + $var(v) >> +">"+ "\r\n"); /* ---- IS THERE A BETTER ?? ..... */* > > > $var(count) = $var(count) + 1; > > } > > > xlog("L_INFO","RELAY_API_RESPONSE - RELAY"); > xlog("L_INFO","MAIN - calling REPLY_302"); > route(REPLY_302); > return; > } > } > send_reply(500, "API Not Available - http response = $http_rs > $http_ok"); > exit; > } > route[RELAY_API] { xlog("L_INFO","RELAY_API - from_ip $si:$sp from_number $fU > to_number $ru"); > $http_req(all) = $null; > $http_req(suspend) = 1; > $http_req(timeout) = HTTP_API_TIMEOUT; > $http_req(method) = "POST"; > $http_req(hdr) = "Content-Type: application/json"; > jansson_set("string","from_ip",$si, "$var(http_routing_query)"); > jansson_set("string","from_port",$sp, "$var(http_routing_query)"); > jansson_set("string","from_number",$fU, > "$var(http_routing_query)"); > jansson_set("string","to_number",$ru , "$var(http_routing_query)"); > > xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST: > $var(http_routing_query)\n"); > $http_req(body) = $var(http_routing_query); > t_newtran(); > http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE"); > xlog("L_INFO","RELAY_API - Waiting for Response"); > } > route[REPLY_302] { > # Sends a 300 Multiple Choices back to the proxy that requested the > routing lookup > xlog("L_INFO","REPLY_302 - send reply"); > sl_send_reply("300", "Multiple Choices"); > xlog("L_INFO","REPLY_302 - exit"); > exit; > } > route[HANDLE_OPTIONS]{ > if(is_method("OPTIONS")) > { > sl_send_reply(200, "OK"); > exit; > } > if ($fU=="ping") > { > sl_send_reply("200","OK"); > exit; > } > } SNGrep output: 1 - INVITE from UAC to Kamailio INVITE sip:918228990@10.20.0.2 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > Max-Forwards: 70 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2> > Contact: <sip:anonymous@10.20.0.1:5063> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > User-Agent: SIPp > Date: Mon, 08 Apr 2024 15:09:51 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 270 > v=0 > o=root 2021555890 2021555890 IN IP4 10.20.0.1 > s=SIPp > c=IN IP4 10.20.0.1 > t=0 0 > m=audio 18422 RTP/AVP 8 0 101 > a=rtpmap:8 PCMA/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 2 - Reply from Kamailio to UAC SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 3 - SIP 300 Multiple Choices from Kamailio to UAC SIP/2.0 300 Multiple Choices > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=57c593265e21c2b70aea50cb414df9cd.32679ccd > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Contact: <sip:sip:918228990@10.20.0.2@sip.domain.io> > Contact: <sip:+351sip:918228990@10.20.0.2@10.20.0.3> > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 4 - SIP 500 from Kamailio to UAC SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM) > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=4eb2322b7d2b68e6fc3168f503344c21-32679ccd > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 5 - ACK from UAC to Kamailio ACK sip:918228990@10.20.0.2 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > Max-Forwards: 70 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=57c593265e21c2b70aea50cb414df9cd.32679ccd > Contact: <sip:anonymous@10.20.0.1:5063> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 ACK > User-Agent: SIPp > Content-Length: 0 and continues with ACK and repeating SIP 500 above.... *Sérgio Charrua* .

4 6

Kamailio Admin Book email
by sadik.oualla.mohamed＠gmail.com 08 Apr '24

08 Apr '24

Hello, Last week, I submitted a request on the Asipto website to purchase the Kamailio Admin Book and received an automatic confirmation email. However, I have not received any further instructions on how to complete the purchase. I replied to the automatic email to follow up, but still, there has been no response. I need to get the book as soon as possible. Please note, I am writing from a different email address than the one I used to make the request on Asipto's site. If there are any issues with their contact email, or if further verification is needed, please let me know how to proceed securely without sharing personal details publicly on this list. Regards, Mohamed.

2 2

customizing the ds_ping_interval dispatcher parameter for destination
by Giuseppe Pandolfi 08 Apr '24

08 Apr '24

Hi, I have one question about dispatcher module in kamailio V.5.3.1. Why the ds_ping_interval parameter is only global value? Is it possible introduce the customization value for single destination? Customization from destination is already possible for ds_ping_from parameter using special attributes in destination list file or db table, is it possible introduce the same management for ds_ping_interval? Thanks, Giuseppe ________________________________ NOTA BENE: Le informazioni contenute nel presente messaggio, compresi gli eventuali allegati, sono strettamente confidenziali ed indirizzate unicamente al destinatario indicato. Qualora non foste il destinatario Vi invitiamo a cancellare il messaggio, ed ogni eventuale documento allegato, non divulgare il contenuto a terze persone ed a volerci avvisare a mezzo posta elettronica. Grazie. A TLC S.r.l. tratta i vostri dati personali nel rispetto della normativa vigente in materia di Privacy (D.Lgs.196/2003 modificato dal D.Lgs.101/2018 e GDPR Regolamento UE n.679/2016). Per consultare l’Informativa completa la invitiamo a visitare il nostro sito web “www.aethra.com<http://www.aethra.com/>” PLEASE NOTE: This message and any attachments are confidential and may contain privileged information. If you are not the intended recipient, you are kindly requested to cancel it, not to disclose the contents to any other person and to notify the sender immediately by return e-mail. Thanks. A TLC S.r.l. collects your personal data in compliance with applicable data protection laws (D.Lgs. 196/2003 amended by Legislative Decree 101/018 and GDPR EU Regulation n. 679/2016). Please visit our website ”www.aethra.com<http://www.aethra.com>” to consult the complete Policy.

1 0

RFC: uptime metric for xhttp_prom module
by Ovidiu Sas 05 Apr '24

05 Apr '24

Hello all, The xhttp_module can export all stats under "stats.get_statistics" RPC command. I was thinking of adding an optional "uptime" stat that will return the kamailio server uptime (like the "core.uptime" RPC command). This will make it easier to add an uptime panel in grafana. The new stat would be controlled by a module parameter (by default disabled). Something like: modparam("xhttp_prom", "xhttp_prom_uptime_stat" 1) And this will generate something like: kam_uptime 671 1712249054631 This can be already implemented in the script using the "prom_gauge_set()" and retrieving the uptime via "jsonrpc_exec()", but it would be nicer and faster to implement it in the module itself. Comments? -ovidiu

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Kamailio Visions 2030 for Appalachia
by Alex Balashov 05 Apr '24

05 Apr '24

For immediate release: HAZARD, Kentucky (1 April 2024)--Evariste Systems LLC, Georgia-based veteran brokers of international charity partnerships, are pleased to announce the conclusion of lengthy trilateral negotiations among Taylor Swift, the Public Investment Fund (the sovereign wealth fund of the Kingdom of Saudi Arabia), and the Kamailio open-source community to bring Kamailio to underserved areas of the US Appalachia region. In a concert in Lexington, Kentucky last Sunday evening, and in her signature style of mystical "reveals" through equivocal song lyrics, Swift auspiciously inaugurated to her Swifties, as die-hard fans of the pop star are known, the beginning of Kamailio Visions 2030. (Editorial note: Swift's agent has threatened copyright litigation in the event that our press agency were to reproduce her meaning-rich lyrics directly, driven by the contention that fair use considerations do not apply to lyrics so "deep" as Swift's. At press time, the matter is being reviewed by our counsel.) Alex Balashov, principal of Evariste, praised Kamailio Visions 2030 as a continuation of Swift's long-standing commitment to forward thinking and innovation: "Taylor has been the driving force behind the use of sustainable, plant-based materials in her 'merch. Today, aided by the internationalisation of Saudi capital, she is helping to bring that same kind of ethical prosperity to Eastern Kentucky, West Virginia and the wider Appalachian coal country." Continued Balashov: "Between Alan Maimon and J.D. Vance, and the revelations of the Purdue Pharma litigation, it's fair to say that the plight of Appalachia, with its high structural unemployment, declining economic base, and the ravages of the opioid epidemic and deaths of despair, has been exposed to a broader audience than ever before. And with that knowledge comes responsibility, because we live with that knowledge now, and we can no longer turn a blind eye. The SIP and VoIP industries have always been a force of conscientious capitalism, and Kamailio a beacon of rectitude in a jaded, amoral, cynical world. A world in which Appalachian children are denied the SIP proxy is not a just world, and we cannot any longer live in such a world." Fred Posner, a deputy Evariste negotiator who was said to have thrived in the pressure of high-stakes negotiations with PIF governor Yasir Al-Rumayyan over the exact implementational modalities of the deal, reviewed the high notes of the landmark economic development accord: "It was a good start to defund the police. But with the unique social challenges of Appalachia, what every struggling family in the ruins of coal country really needs is access to clean, ethically sourced, fair-trade SIP proxies. That is why I am uplifted to say that we will be dropping, through a coordinated, large-scale airlift operation, a Raspberry Pi with Kamailio pre-installed, to every household with an income at or below 250% of the Federal Poverty Level (FPL) in the Appalachian region. The Pi chassis will be dye-free, BPA-free, and will be held together with a plant-based, biodegradable, non-toxic glue that does not produce intoxicating effects when sniffed. By 2030, every child alive in Appalachia today will have had several years to benefit from the power of a miniaturised SIP proxy appliance, and that is why we are calling it Kamailio Visions 2030." When pressed by us, Balashov noted that the prosocial objectives and economic development potential of Kamailio Visions 2030 has drawn scepticism from some quarters. "These are those people who think that Rasperry Pi devices preloaded with Kamailio are not what Appalachia most needs today. Those people would be wrong. As any metaphysicist can tell you, causes do not necessarily resemble their effects." "Taylor and her people know that Kamailio Visions 2030 draws on a long history of highly effective technological and app-based interventions in economically underprivileged areas, such as Nicholas Negroponte's One Laptop per Child (OLPC) scheme, and the famous carpet-bombing of Eritrea with second-generation iPads by Selena Gomez. All of this had tremendous results." "Salutary results", added a representative of the PIF. Balashov pointed us to the foundational problem statement of Kamailio Visions 2030, which cited the concept of "Technik", as developed in "Hybrid Reality", a technocratic manifesto (in e-book form) by much-loved TED visionaries Parag and Ayesha Khanna. There, the authors declared: "Good Technik requires a combination of the attributes that deliver high human development, economic growth, political inclusiveness, and technology preparedness.” "In short", said Balashov, "Taylor, with some help from the efficient deployment of Saudi capital, will bring good technik to Appalachia. The tide of technik lifts all ships." -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

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low performance with no apparent reason
by Sergio Charrua 05 Apr '24

05 Apr '24

Hi all! I have been doing some performance tests with Kamailio 5.7.4 and SIPp. The infrastructure is as follows:3 VMs running on VMWare ESXi running: UAC on 10.20.0.1 with SIPP-> Kamailio on 10.20.0.5 -> UAS on 10.20.0.3 The Kamailio VM has 6 dedicated vCPU of type Intel(R) Xeon(R) Silver 4216 CPU @ 2.10GHz and, 2 NICs and 4Gb RAM and MariaDB 10.6 as DB Backend., all running on a HP G380 host with a gazillion CPUs and a googol disk space! I currently have 3 scripts: - script #1 stateful with RTJson and simulating requests to routing engine and accounting - script #2 stateful but with just a simple routing to UAS, no rules, no DB, - script #3 stateless with a forward to UAS With script #3 I can go up to 2000CPS without issues with CPU at 37%! Above that value, I get retransmissions everywhere. On both scripts #1 and #2, the limit is 330CPS max after which I get a lot of retransmissions, while CPU/Core usage on Kamailio server stays below 10%. So I do not expect this to be a CPU issue. I could not understand why such (low) results, so I followed this article found at https://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c <https://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c> and created exact same scenarios, with kamailio script and SIPP templates available on the article, hoping for better results. But I get the same results: between 300 and 330CPS which is far, very far from the 7000CPS found in the article! I understand that I'm using VMs and probably the tests made for the article, which is pretty old already, were made on physical servers. Still, I would not expect 95% of lower performance! Any clue what could be the issue? I suspect NICs, but.... Any tips anyone could share? Thanks in advance! *Sérgio Charrua*

7 31

Statistics for all generated
by Alfredo Román Domínguez 05 Apr '24

05 Apr '24

I am using Kamailio versions 5.5.7 and 5.8.0 as a stateful proxy (with tm module) and I want to have metrics for all the SIP messages received and sent. I am using the statistics module for counting messages. I have in configuration file the following routes to capture events and count them: onsend_route, onreply_route and event ( corex:reply-out, tn:local_response, sl:local-response, tm:local-request, sl:filtered-ack and network:msg) I have found the following problems that I need some hints how to solve them: 1) When a response is received to a request, I can count the response received in the reply_route but I don't find a way that works to count the reply being forwarded to the other side. onsend_route works fine for requests being forwarded. Is there any event_route or any trick that can be used to count the reply being forwarded similar to the onsend_route for requests? 2) When Kamailio receives a CANCEL it generates and sends automatically the 200 OK for the CANCEL and a 487 for the INVITE which started that dialog which is being cancelled, but I don't find a way that works to count those two responses Kamailio generates. 3) After having cancelled a dialog on one direction due to receipt of a CANCEL, kamailio generates local requests on the other direction, a CANCEL and an ACK also when it receives a 487 for the INVITE being cancelled. I don't find the way to count those. onsend_route does not work for locally generated CANCEL and ACK requests and the event_route[tm:local-request] only works for requests created outside of the tm module but not those created by the tm module itself (the CANCEL and the ACKs) since the tm module is not using its own function t_uac_prepare to create those requests (that function is the one that calls the event_route). Is there any statistic generated by the tm module or elsewhere or any other trick so I can count those requests? 4) When Kamailio generates retransmissions how can I count those? onsend_route, onreply_route and event ( corex:reply-out, tn:local_response, sl:local-response) doesn't work for retransmissions. Are there statistics generated somewhere in the tm module for retransmissions? Is there any other way to count those. -- El contenido de este mensaje puede contener información confidencial sometida a secreto profesional. Si usted ha recibido este correo por error, no está permitida su distribución. Por favor elimine cualquier copia, archivo adjunto e infórmenos por esta vía. Los datos personales contenidos en este correo y que han sido facilitados por usted o provienen de fuentes de acceso público serán utilizados por Future Space, S.A., para gestionar nuestra relación contractual o pre-contractual, resolver su solicitud y mantenerle informado sobre los productos o servicios solicitados o similares. Se conservarán mientras exista un interés mutuo para ello, no se oponga al tratamiento o solicite su supresión. Los datos no serán comunicados a terceros, salvo obligación legal. Puede ejercitar sus derechos de acceso, rectificación, supresión, limitación del tratamiento, portabilidad de los datos y oposición mediante escrito dirigido a Future Space, S.A. Avenida de Tenerife 2, Edificio 2, Planta 1ª, 28703 San Sebastián de los Reyes, Madrid o mediante correo electrónico a dpo(a)futurespace.com <mailto:dpo@futurespace.com>. Si consideras que se han infringido tus derechos puedes presentar una reclamación ante la www.agpd.es <http://www.agpd.es>. The contents of this electronic mail may be confidential or privileged. If you have received this message by mistake, please do not send it to anyone. We ask you to delete any existing copies, attached files and to notify us accordingly. The personal data provided or obtained from publicly accessible resources will be processed by Future Space, S.A. in order to manage our contractual or pre-contractual relationship, solve your request and keep you informed about activities or products requested by you or similar. Personal data will be stored while a mutual interest exist or either erasure or objection to process them is requested. Data will not be disclosed to third parties unless there is a legal obligation to do so. You have the rights to access, rectify, erasure, restriction of processing, data portability and oppose any processing of your personal data held by Future Space, S.A. To make effective use of your rights, please contact us at Future Space, S.A. Avenida de Tenerife 2, Edificio 2, Planta 1ª, 28703 San Sebastián de los Reyes, Madrid or by email at dpo(a)futurespace.com <mailto:dpo@futurespace.com>. If you consider processing of your personal data infringes data protection regulations you can file a claim with www.agpd.es <http://www.agpd.es>

1 0

Kamailio v5.8.1 Released
by Daniel-Constantin Mierla 03 Apr '24

03 Apr '24

Hello, Kamailio SIP Server v5.8.1 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.0. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.1. For more details about version 5.8.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/04/kamailio-v5-8-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Releasing v5.8.1
by Daniel-Constantin Mierla 03 Apr '24

03 Apr '24

Hello, with many countries having public holidays around Catholic Easter, I am considering to release Kamailio v5.8.1 (out of branch 5.8) on Wednesday, April 3, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 1

start_recording' or 'record-call on' is not invoking
by pawan srivastava 03 Apr '24

03 Apr '24

Hi I am testing rtpengine recording with kamailio in lab. I have 1 endpoint registered to Kamailio. My kamailio and rtpengine are running on same box. I have another kamailio2 instance with 2nd endpoint to make a call to 1st endpoint. I have observed that Response dump for 'offer'/'answer' but further start recording didn't invoke. root@ip-kamailio:/etc/kamailio# rtpengine-ctl list sessions all callid: J0WtgYIKd0Kwj1iL6BiFOg.. | deletionmark: no | created: 1712120753 | proxy:127.0.0.1:52642 | redis_keyspace:0 | foreign:no callid: uH_ozKj18Nafshsm_sTElQ.. | deletionmark: no | created: 1712120730 | proxy:127.0.0.1:52642 | redis_keyspace:0 | foreign:no callid: VYIw791dJska8PiIFspy8w.. | deletionmark: no | created: 1712120716 | proxy:127.0.0.1:54804 | redis_keyspace:0 | foreign:no callid: nP89gWwjhsg_TSZrqu16nQ.. | deletionmark: no | created: 1712120716 | proxy:127.0.0.1:56207 | redis_keyspace:0 | foreign:no root@ip-kamailio:/etc/kamailio# I am new to kamailio and missing configuration. Suggest me the correct configuration to invoke recording. attached config and logs Regards, Pawan

1 0

How to relay SIP messages to specific registered user in Kamailio?
by o.atef＠fiberme.com 02 Apr '24

02 Apr '24

Hello guys, I'm recently new player in Kamailio SIP server and want to use it as SIP proxy server to Asterisk PBX server. first of all, I've successfully registered my Asterisk to Kamailio as user "test" to get its "location" because my Asterisk is behind NAT. Simply now I want to relay SIP messages to this user's location to receive them all in Asterisk. After I searched, I found this function rewritehost() but it needs to be a host or IP address, but in my case I want some way or function to relay to user "test" and then it well get the user location from Kamailio datebase automatically? I hope it's clear XD I tried to find a way to do MSQL query in kamalio.cfg to get the user contact from "location" table and then relay to the result but it did not work. I guess that I did it in wrong way but finally the Kamailio tutorials are so rare on google. Thanks in advance, Omar

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sr-users April 2024