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37519 discussions

setting $du in branch route has no effect in 5.3
by Juha Heinanen 30 Mar '20

30 Mar '20

I noticed that in 5.3 and master setting $du in branch route has no effect. It works fine in 5.2 using exactly the same config. -- Juha

2 12

Problem faced when Using Kamaiio as Session Refresher
by harneet singh 30 Mar '20

30 Mar '20

Hi, I am fairly new to Kamailio and had a question regarding how to use Kamailio to enable Session refresh functionality when Kamailio is acting as Sip Stateful Proxy. Kamailio Version used: *5.3.2* with *Call Load based routing* using the *dispatcher *module. * From what i understand from the documentation, Kamailio will probably not be acting as a session refresher, but Kamailio can tear down the call in case session refresh is negotiated, between the caller and the callee(via Kamailio Sip Proxy), and no message exchange happens in the duration set in Session-Expires header. *Is my understanding correct?* ** *I believe the above functionality is possible by using the *sst* and *dialog* module. I have set the same according to the documentation but I keep getting the following error: "dialog [dlg_handlers.c:681]: *get_dlg_timeout(): invalid AVP value, using default timeout*" Can someone share a working example? * When i tried hardcoding the timeout value by setting the timeout_avp to a specific value, Kamailio did sense a timeout and hence sent a BYE towards the caller and the Callee side both(which is what the requirement is), however, i do see the *dialog is still not cleared* in the "kamcmd dispatcher.list". Output excerpt below for reference: [root@CPaaSVM ~]# kamcmd dispatcher.list { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:172.27.44.121:5080 ;transport=tcp FLAGS: AP PRIORITY: 0 ATTRS: { BODY: duid=sample-cas;maxload=1000 DUID: sample-cas MAXLOAD: 1000 WEIGHT: 0 RWEIGHT: 0 SOCKET: } LATENCY: { AVG: 111.304000 STD: 1042.193000 EST: 2.385000 MAX: 9999 TIMEOUT: 1 } RUNTIME: { DLGLOAD: *1* } } } } } } It is noteworthy that in case the BYE is initiated by either the caller or the callee, the dialog is cleared properly and the DLGLOAD is set to 0 on call termination. Any pointers for the above questions would be highly appreciated. Regards, Harneet -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

3 17

Transaction Matching Failed - ACK BYE
by egemen ulus 29 Mar '20

29 Mar '20

Hi All, I'm trying to establish a topology with Kamailio which is located between Asterisk and Outside. My issue is "transaction matching failed" (t_check_trans failed). That means Kamailio does not recognise ACK & BYE message to forwards. I had a check on email list and google but found nothing. I'm using it with record_route. SIP signalling and part of kamailio.cfg is below. Is there any suggestion? route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { xlog("ack is came: $si:$sp\n"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { route(RELAY); exit; } else { exit; } } sl_send_reply("404","Not here"); exit; } route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } route(RTPPROXY); if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } Asterisk > Kamailio INVITE sip:user@outside_ip SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK1b56cce0 Max-Forwards: 70 From: "KUYRUK" <sip:0553847<tel:+10553847>aaaa@asterisk_ip>;tag=as3dde7cdb To: <sip:user@outside_ip> Contact: <sip:0553847<tel:+10553847>aaaa@asterisk_ip:5060> Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.32.3 Kamailio > Outside INVITE sip:user@outside_ip SIP/2.0 Record-Route: <sip:kamailio_ip;lr;nat=yes> Via: SIP/2.0/UDP kamailio_ip;branch=z9hG4bK7b3f.8e4716eeea4b0dc45cc51d21b5aded57.0 Via: SIP/2.0/UDP asterisk_ip:5060;rport=5060;branch=z9hG4bK1b56cce0 From: "KUYRUK" <sip:0553847<tel:+10553847>aaaa@asterisk_ip>;tag=as3dde7cdb To: <sip:user@outside_ip> Contact: <sip:0553847<tel:+10553847>aaaa@asterisk_ip:5060;alias=asterisk_ip~5060~1> Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.32.3 Outside > Kamailio SIP/2.0 200 OK Via: SIP/2.0/UDP kamailio_ip;branch=z9hG4bK7b3f.8e4716eeea4b0dc45cc51d21b5aded57.0 Via: SIP/2.0/UDP asterisk_ip:5060;rport=5060;branch=z9hG4bK1b56cce0 From: "KUYRUK" <sip:0553847<tel:+10553847>aaaa@asterisk_ip>;tag=as3dde7cdb To: <sip:user@outside_ip>;tag=kv8iml6ur9 Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060 CSeq: 102 INVITE Record-Route: <sip:kamailio_ip;lr;nat=yes> Contact: <sip:user@outside_ip:5060;alias=x.x.x.x~18005~6;transport=udp> Kamailio > Asterisk SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk_ip:5060;rport=5060;branch=z9hG4bK1b56cce0 From: "KUYRUK" <sip:0553847<tel:+10553847>aaaa@asterisk_ip>;tag=as3dde7cdb To: <sip:user@outside_ip>;tag=kv8iml6ur9 Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060 CSeq: 102 INVITE Record-Route: <sip:kamailio_ip;lr;nat=yes> Contact: <sip:user@outside_ip:5060;alias=x.x.x.x~18005~6;transport=udp> Asterisk > Kamailio THIS ACK IS NOT ROUTED TO OUTSIDE BY KAMAILIO!! ACK sip:user@outside_ip:5060;alias=x.x.x.x~18005~6;transport=udp SIP/2.0 Via: SIP/2.0/UDP asterisk_ip:5060;branch=z9hG4bK73366924 Route: <sip:kamailio_ip;lr;nat=yes> From: "KUYRUK" <sip:0553847<tel:+10553847>aaaa@asterisk_ip>;tag=as3dde7cdb To: <sip:user@outside_ip>;tag=kv8iml6ur9 Contact: <sip:0553847<tel:+10553847>aaaa@asterisk_ip:5060> Call-ID: 62a480bf71d2a1aa0ec94b454cd98945@asterisk_ip:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.32.3 Get Outlook for Android<https://aka.ms/ghei36>

2 6

Kamailiio TLS and Let'sEncrypt certs
by Igor Olhovskiy 29 Mar '20

29 Mar '20

Hi! Actually I’m trying to get Kamailio to work as MS Teams SBC following by perfect article https://skalatan.de/en/blog/kamailio-sbc-teams <https://skalatan.de/en/blog/kamailio-sbc-teams> It works well, but one thing is bothering me. I’m using Let’sEncrypt certs (actually, works well), but with setting in tls.conf verify_certificate = yes require_certificate = yes It’s giving an errors like /usr/sbin/kamailio[4551]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:1416F086:SSL routines:tls_process_server_certificate:certificate verify failed /usr/sbin/kamailio[4551]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f03e6d23d88 r: 0x7f03e6d23e08 (-1) They are resolved with setting these settings (verify/require) to off (actually, as mentioned here - https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ <https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/>), but I’m really curious - why? As I got, it’s using openssl verify on a background, but this check locally passed with openssl verify -CAfile issuer.crt myserver.crt myserver.crt: OK So, is there any tricks to lets encrypt or just some misconfig in tls.cfg? Now it looks like one from article [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/tls/myserver.key certificate = /etc/kamailio/tls/myserver.crt ca_list = /etc/kamailio/tls/issuer.crt [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/tls/myserver.key certificate = /etc/kamailio/tls/myserver.crt ca_list = /etc/kamailio/tls/issuer.crt — Regards, Igor

3 3

Kamailio OPTIONS Round-Trip
by Soltanici Ilie 27 Mar '20

27 Mar '20

Hi, With Asterisk, we are able to get some peer round-trip connection statistic by setting qualify=yes for the specified peer. It sends periodic OPTIONS to the peer and calculates the time round trip time. It's something like - "Status: OK (30 ms)". Is there any way to achieve this in Kamailio by using nathelper module, or any other? Thank you.

9 13

Segmentation fault in DMQ module
by Virgílio Cunha 27 Mar '20

27 Mar '20

Hello, I have in production environment with 4 proxies using DMQ to sync terminal registers and the DMQ configuration parameters is following: loadmodule "dmq.so" #modparam("dmq", "server_address", "sip:MY_IP_ADDRESS:MY_PORT_ADDRESS") modparam("dmq", "notification_address", "DMQ_HOSTS") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) In the lab environment with the same configuration that in prod, I can simulate this problem with 12000 terminal REGISTERS and, when doing several restarts on a proxy, one of them is generated a coredump. Analyzing the core, I can see that the problem is concurrency in the dmq synchronization. Job_queue_size (...) is called with a valid *queue *and, but inside this function, the *queue *is null. (gdb) up #1 0x00007fefe58ee246 in job_queue_size (queue=0x0) at worker.c:254 254 return atomic_get(&queue->count); (gdb) print queue $1 = (job_queue_t *) 0x0 (gdb) up #2 0x00007fefe58ed920 in add_dmq_job (msg=0x7ff028b453f8, peer=0x7fefebcc4df8) at worker.c:184 184 if(job_queue_size(workers[i].queue) == 0) { (gdb) print i $2 = 1 (gdb) print workers[i].queue $3 = (job_queue_t *) 0x7fefebce49b0 If I configure DMQ with num_workers=1 I can't reproduce this problem. I'm using the kamailio 5.0.8 release. Is this problem known to you? What is the right way to solve it? Best Regards, Virgílio Cunha

3 7

Multiple advertise addresses per socket
by Sergiu Pojoga 25 Mar '20

25 Mar '20

Hi there, Question: how can I advertise and later match multiple FQDNs per socket? Example: auto_aliases=no alias="mydomain.net":5060 alias="mydomain.net":5061 alias="client1.mydomain.net":5060 alias="client1.mydomain.net":5061 # looks like I can advertise only one address per socket? listen=udp:65.65.65.65:5060 advertise "mydomain.net":5060 listen=tls:65.65.65.65:5061 advertise "mydomain.net":5061 Adding RR before relaying downstream like so: record_route_preset("client1.mydomain.net:5061;transport=tls;r2=on", "client1.mydomain.net:5060;r2=on"); This is the warning (just warning, works fine otherwise) after receiving for example the ACK from upstream following the 200 OK: *WARNING*: {1 19082 ACK } rr [loose.c:768]: rr_do_force_send_socket(): no socket found to match second RR (sip:client1.mydomain.net:5061 ;transport=tls;lr;r2=on) Figured warning itself can be turned off with modparam "enable_socket_mismatch_warning" of RR, still I'd like to know if I'm doing something wrong or there's a better way. Much obliged, --Sergiu

2 3

Kamailio 5.3.2 - Signal 11 on startup
by José Seabra 25 Mar '20

25 Mar '20

Hello there, I'm facing an issue with kamailio 5.3.2 when it is starting up, it breaks with signal 11 and generates a coredump file: - https://pastebin.com/FSqyGFp8 This seems to be something related to dmq module, kamailio exits with signal 11 when it receives a DMQ notification_peer message. I did a few more tests and I noticed that problem disappears if I configure DMQ with one worker only, which takes me to think that it's something related to concurrency. Best regards -- Cumprimentos José Seabra

2 7

about kamailio.cfg NAT test
by Juha Heinanen 24 Mar '20

24 Mar '20

In kamailio/etc/kamailio.cfg NAT test is based on nat_uac_test("19"). 19 includes test 16: 16 - Test if the source port is different from the port in the “Via” header. If the “Via” header contains no port, it uses the default SIP port 5060 Based on a couple of tests using baresip, looks like that results in false positive when UA connects to SIP proxy via TCP. An example: T 2020/03/24 13:57:12.685441 192.26.134.10:38940 -> 192.26.134.1:5060 [AP] #127 REGISTER sip:test.tutpro.com SIP/2.0. Via: SIP/2.0/TCP 192.26.134.10:44717;branch=z9hG4bK5b6aa423104ef942;rport. Does someone know why test 16 is included? -- Juha

3 5

Transformations - urlencode
by João Vitor Arruda 24 Mar '20

24 Mar '20

Hello, I was looking in the transformation documentation ( https://www.kamailio.org/wiki/cookbooks/devel/transformations) something to URL encode a pv or var for use in a HTTP query but was not able to find it. Digging a bit in the source code I found that "s.urlencode.param" actually exists and from my initial tests it works just fine. I'm wondering if there is any reason why this is not mentioned in the documentation of if it should not be used at all. Thanks, Joao Arruda

3 3

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