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18 participants
37398 discussions

mid-registrar example
by Easy Rider 19 May '20

19 May '20

Hi everyone I'm trying to implement OpenSIPS-like mid registrar functionality using UAC module Up to now, I was simply performing authentication on my kamailio instance, saving to local domain and forwarding the request to another proxy I'd like to lower the number of requests sent to this other proxy, by forwarding only a few of the incoming requests. This is not a problem to do, but then comes the handling of expiration. If the SIP UAC is disconnected, the registration expires on my kamailio instance, but I don't know how to correctly notify the other proxy on the unregistration (I was trying to build and send a request using uac_send_req(), but this is useless as I can't set the CSeq value) While looking for a solution to my problem, I discovered that OpenSIPS had a mid registrar module doing exactly what I'd like to do (according to the documentation at least). I also saw old messages on this list that it was possible to achieve the same result with UAC module on kamailio In particular, I don't see why (or if) I need to populate the uacreg table. I don't see why it should be necessary, and it would have a great impact on performance to use a db for this matter. I also don't need (I think) to have different local and remote from and to URIs I think I need to build the a new REGISTER request using $uac_req, but I don't see any contact header in the outgoing request, despite having set reg_contact_addr value Thanks for any clue you could give me

4 7

nathelper without usrloc
by Daniel W. Graham 17 May '20

17 May '20

I’ve been missing with this for a while and seem to be missing something. Any suggestions on what is missing here? Trying to use set_contact_alias() and handle_ruri_alias() from nathelper module and nat_keepalive from nat_traversal module, without registrar. I had register keepalive working, that has since broke. When register keepalive was working, I was able to place call in either direction but ACK and BYE was not being routed past kamailio. Registrations are forwarded to the PBX using add_path() and is working. Also not included below is the routing to the PBX, that is just setting $du and t_relay, and is also working. Topology is: UA1 -> NAT -> kamailio -> PBX -> UA2 Using default config file as the example, modified with above changes. I also removed RTP config as that is a non-issue. request_route { …… # FLAG MESSAGES FROM PBX setflag(FLT_PBX); route(NATDETECT); …… route[NATDETECT] { if (nat_uac_test("19")) { force_rport(); set_contact_alias(); nat_keepalive(); } return; } route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { route(DLGURI); } else if ( is_method("ACK") ) { 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] { if(isflagset(FLT_PBX)) { handle_ruri_alias(); } if(!isflagset(FLT_PBX)) { set_contact_alias(); } return; } route[DLGURI] { if(!isdsturiset()) { handle_ruri_alias(); } return; } branch_route[MANAGE_BRANCH] { route(NATMANAGE); } onreply_route[MANAGE_REPLY] { if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; -dan

4 19

msilo dump, ordering
by Péter Barabás 16 May '20

16 May '20

Hi, I have a strange situation in messaging. I start to send messages continuously from one user to another on iOS devices. The messages are stored in silo since receiver is in background and registration does not exists. During the message flow, the receiver registers and the messages are dumped from silo when the sender still sends messages. The order of messages are important because of encryption, if an earlier message arrives after a later one to the receiver, a new key exchange is needed to decrypt messages, etc... So the question: * Is there any technique where I can delay or lock incoming messages on kamailio until silo messages successfully dumped in order to preserve message ordering? Thanks Peter

2 1

Facing issue with call disconnection BYE packets are not accepted by kamailio
by Voip support 15 May '20

15 May '20

Dear Community, I wrote before about my random issues with calls disconnection. We found some issue in our VMWare environment that much packets was lost. We resolved the issue by moving to other VMWare host however the issue is still present. Currently it is random and some calls do not disconnect due to no BYE forwarded by kamailio to other side. Here is the pcap of such call : https://www.dropbox.com/s/7bktz3p4im6ztld/bad-dialog-call.zip?dl=1 We use dialog module and dispatcher. dlg_manage(); is only executed in this block : # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_DLG 4 . . . # account only INVITEs if (is_method("INVITE")) { setflag(FLT_DLG); # create dialog << i added it setflag(FLT_ACCMISSED); # do accounting even if failed << i added it setflag(FLT_ACC); # do accounting #route(LIMIT_CALLS); dlg_manage(); sip_trace(); } I really have no idea i am unable to find differences between the bad and good call. The SIP packets looking good but the BYE is not processed. Please help me out how to debug it? I was thinking of adding a log for checking if request is BYE and if it is check if it match a dialog using is_known_dlg() method. Please let me know if you see what is wrong. Just to mention kamailio is listening on private IP with advertise to public IP. Best regards!

2 4

replace SIP methods
by AL RSM 15 May '20

15 May '20

Hi, I was wondering if Kamailio can replace SIP methods? i.e., when Kamailio receives SIP INVITE, change it to NOTIFY methods and fork it to multiple destination. Thanks, AL

2 1

Dispatcher - ds_select_dst based on attribute value
by rainolf 15 May '20

15 May '20

Hi everyone, Let's say i want to add additional attribute in every dispatcher database entry in form of: *Example:* extension_set=100-199;branch=NewYork Let's say also i have more than one entry in dispatcher database with these attributes and same priority. *EX:* IP=192.168.1.1;priority=1;attr=extension_set=100-199;branch=NewYork;Descrioption=PBX Production IP=192.168.1.2;priority=1;attr=extension_set=100-199;branch=NewYork;Descrioption=PBX Backup What i want to achieve is to select multiple destination with ds_select and algorithm 8(priority) based on dialed extension. I can get attributes with: $(xavp(_dsdst_=>attrs) but, off course, this may be possible only after destination selection has been made by dispatcher. Currently i've create a switch statement in order set statically a variable with the right priority like: # dial number selection switch($rU) { case /"^[1][0-9][0-9]$": # starting with 1XX xlog("L_NOTICE","THE DIALED NUMBER IS: $rU ... Searching for available destinations!!!\n"); $var(ds_priority) = 1; break; default: exit; } ds_select_dst($var(ds_priority), 8); #Get an up destination from dispatcher with priority 1 xlog("L_NOTICE","Calling Dispatcher $du -- $siz:$sp with PARAMETER ATTRIBUTE: (attrs: $xavp(_dsdst_=>attrs))\n"); xlog("L_NOTICE","THE BRANCH IS SET AS: $(xavp(_dsdst_=>attrs){param.value,branch})\n"); xlog("L_NOTICE","THE EXTENSION SET IS: $(xavp(_dsdst_=>attrs){param.value,extension_set})\n"); t_on_failure("DISPATCH_FAILURE"); but i would like to achieve something more dynamic which permit me to create a simple database entry. Probably there's a better way to do it maybe with the help of other modules or database tables. If someone can suggest me the best way or simply pointing me on the right direction. Appreciate Regards

1 0

Re: [SR-Users] Other issues encountered with MySQL 8 and Kamailio
by Yehuda Y 15 May '20

15 May '20

Hi Did you manage to use mysql8 with kamailio? Yehuda

1 0

Solution for Invite retransmission
by Duarte Rocha 14 May '20

14 May '20

Greetings, I have an issue where a client doesn't get the responses to the INVITE sent and as such keeps sending me retransmissions of the INVITE. While the transaction is still up, Kamailio does its job. However, when the transaction is closed Kamailio processes the request again as a first request (Doing Routing and Dispatcher operations again). In order to avoid the issue i've made the following code : // If it's the first INVITE if( is_method("INVITE") && !has_totag() ) { if(t_check_trans()) { xnotice("TRANS - INVITE Retransmission"); } else if ( is_known_dlg()) { xerr("KamTAG: INVITE in dialog without To Tag "); exit; } } Is this a good solution and still compliant with the SIP rules? Best Regards

3 3

Problem with Teams and Kamailio and calls hangup after 6 seconds
by sip user 13 May '20

13 May '20

Hi, I'm trying to connect Teams with Kamailio. I'd configured and it works, but a few days ago, calls Teams to kamailio no works. I call from Teams and in Kamailio I see the call, ring the extensions in Kamailio, but after 6 seconds, the call hangup. I received the INVITE, I send Trying and 200 Ok, but never received ACK from Teams. Anyone knows how can I check it? I have configurated SBC in teams like https://skalatan.de/en/blog/kamailio-sbc-teams Thanks

1 0

Issue of calls drop 32s
by Rex Lin (林昱頡) 13 May '20

13 May '20

Hi, It’s weird that the calls always drop after 32s while the callee is using public ip. Also, Server forwards ACK to itself with UDP, instead of forwarding to callee with TLS. Moreover, I cannot see “Received” in AOR as the callee finish the registration. I recognized it’s marked by nat_uac_test(“19”) and set with fix_nated_register(), the problem can be resolved. My scenario is as below, for your reference. 1.Kamailio version is 4.4.7, listening public ip. 2.Caller it’s behind NAT. 3.Clients of Linphone. Do you know the reason why it happened? Best Regards, Rex

2 1

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