- sr-users - kamailio.org

How to ensure separate kamailio instances dispatchers make the same decision?
by George Diamantopoulos 20 May '20

20 May '20

Hello all, I need to have two separate kamailio instances' dispatcher modules make the same decisions when using algorithm 7 (hash over pvar). What do I need to do to ensure this? Note that for design reasons, the two instances cannot share a dispatcher table from db. If I ensure the "setid" group used for algo 7 in the respective cases contain the same group of hosts, is it enough? Do other things matter, such as the ordering of the group members in the table (i.e. different AUTO INCREMENT ids?). Does the setid need to be the same number? Do I need to ensure the 'destination' values are identical (i.e. not using IPs for dispatcher table A and hostnames for dispatcher table B)? If someone knows what the criteria is for matching a hashed pvar to a member of a dispatcher setid and how this can be made deterministic, I would be grateful. Thanks! BR, George

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get_record(): lookup failed
by Chris Atta 20 May '20

20 May '20

get_record(): lookup failed when try to add or invite a user to a private chat room with imc module. join to a public room works!

2 1

s.replace Transformation and Newline
by Carsten Bock 19 May '20

19 May '20

Hi, I am trying the following: - extract the message body - replace some content - store it in a hashtable - send it out in a reply in later request I do the following: [...] $sht(sdp=>$ci) = $(rb{s.replace,a=sendrecv,a=sendrecv\\r\\na=curr:qos local none}); [...] set_reply_body("$sht(sdp=>$ci)", "application/sdp"); send_reply("200", "OK"); [...] It works like a charm, however I'm facing one small issue: My SDP looks like this: [...] a=rtpmap:8 PCMA/8000 a=sendrecv\r\na=curr:qos local none a=rtcp:34949 [...] It seems that s.replace is escaping characters. Anyone any ideas? Thanks, Carsten $var(x) = "abababa"; $(var(x){s.replace,a,c} => "cbcbcbc" -- Carsten Bock I CTO & Founder ng-voice GmbH Trostbrücke 1 I 20457 Hamburg I Germany T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock

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rfc: types of announcements on sr-users list
by Daniel-Constantin Mierla 19 May '20

19 May '20

Hello, another topic discussed during the online devel meeting last week was about what kind of announcements should be allowed on this list. The purpose of sr-users list is to be an open community forum to help each other to install, configure and operate Kamailio instances. So far we announced here any new Kamailio versions (obvious) as well as releases to the open source (and free) applications/tools strictly developed for Kamailio (the kamcli and siremis), plus the events related to Kamailio community or project development. Last I am aware of being "accepted" to be announced on this list is the availability to hire Kamailio experts or to be hired, relying on common sense and not abuse of sending often such announcements (which happened so far). Now it came into discussions if announcements about other applications or projects that are built with Kamailio or useful for building/operating Kamailio-based services. Becoming aware of such things can be very valuable for the community, but because this may be interpreted differently (what's useful for me, might not be for others), we should be careful and set the rules to avoid "polluting/spamming" this group of discussions. On the online meeting was somehow considered acceptable to have one time announcement when launching such a project. Clearly this is not the place for announcing every new featured added in external projects or their releases. Moreover, the projects have to be related from real time communications perspective, like: - the project/application is built on top of kamailio (e.g., pbx system with kamailio inside and web ui) - completes what kamailio can do or needs for real time communication services (e.g., monitoring the sip/rtp traffic, billing/charging of sip calls, sip softphone, lib or media server ...) Again, only if these projects/applications are open source and free to use, plus the announcement only when launching the project (or better said, at the moment of the first usable version). Posts to this mailing lists can be done only by members, so who sends the announcement has to be part of the community. If people find it acceptable to get such announcements, I would also consider to revise the policy if we start having more than 2-3 such announcements per month. So, the question for now is if people here would find in useful to have such announcements. Let's see what arguments are for one or the other variant and get to conclusion good for most of us. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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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

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nathelper without usrloc
by Daniel W. Graham 18 May '20

18 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

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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

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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

Kamailio 5.2.7 Released
by Henning Westerholt 13 May '20

13 May '20

Hello, Kamailio SIP Server v5.2.7 stable release is out. This is a maintenance release of the previous stable branch (5.2), that includes fixes since the release of v5.2.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.7. For more details about version 5.2.7 (including links and guidelines to download the tar file or from GIT repository), visit: https://www.kamailio.org/w/2020/05/kamailio-5-2-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

Re: [SR-Users] [sr-dev] Unable to make SIP Calls (403 Forbidden - Domain Not Served)
by Pavithra M 13 May '20

13 May '20

Hi, I have my kamailio ims with NAT and rtpproxy enabled. When i make a call from outside , rtp packets is not going. Could you please tell me how to configure NAT to get rtp packets in pcscf . On Fri, Apr 17, 2020 at 4:30 PM David Villasmil < david.villasmil.work(a)gmail.com> wrote: > It doesn’t make a lot of sense to see looping calls “connected”. If they > connect, then they’re not looping, unless there is some counter somewhere > that answers the call when it sees a loop. > > That’s very weird. > > Use sngrep and trace on ALL interfaces. > > On Fri, 17 Apr 2020 at 11:46, Pavithra M <pavimohan3096(a)gmail.com> wrote: > >> Hi, >> In Wireshark , I couldn't able to see the looping packets .In debug logs >> of pcscf , I am able to see the looping occurs . But how then the call gets >> connected . That's where I am struck . >> >> On Fri, Apr 17, 2020, 4:11 PM David Villasmil < >> david.villasmil.work(a)gmail.com> wrote: >> >>> You need to trace sip. There’s a loop somewhere. Use tshark (wireshark) >>> or sngrep or ngrep, or some similar network tracing tool. >>> >>> On Fri, 17 Apr 2020 at 11:34, Pavithra M <pavimohan3096(a)gmail.com> >>> wrote: >>> >>>> Hi , >>>> >>>> I am getting the call connected and got 200 ok from the client . But >>>> the INVITE packet goes multiple times between terminating pcscf and s-cscf >>>> and gives too many hops error .Even After that call goes perfectly fine >>>> with 200 ok message. >>>> What would be the reason . I am not getting any errors in my logs with >>>> debug log level 6 . >>>> >>>> Kindly tell me what could be the issue . >>>> >>>> >>>> >>>> >>>> On Thu, Apr 16, 2020, 2:16 PM David Villasmil < >>>> david.villasmil.work(a)gmail.com> wrote: >>>> >>>>> That usually happens when you don’t set a next hop, or destination >>>>> where the packet should go, so the packet of forwarded to the same >>>>> destination, which is the proxy again. >>>>> >>>>> On Thu, 16 Apr 2020 at 06:56, Pavithra M <pavimohan3096(a)gmail.com> >>>>> wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> I have enabled all debug logs and i traced it and I found that as you >>>>>> said it is keep on looping the invite in pcscf instead of going to other UE >>>>>> . I couldn't able to find the reason.Can you please give me some input on >>>>>> this. >>>>>> >>>>>> Debug logs of pcscf : >>>>>> 0#015#012Via: SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.c9499ad5ce252a99a36ad8b9b70f04d4.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.a00d9260e4373806e691bfa2828ab215.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.6ab08c40989814cf3726b826bdef39d2.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.e650088370f5d61c0d902d1a40fc1252.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.44a3f1c7026e2d42810722ae5c4bf942.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.54030b75c055258251ec7e6980c78c74.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.9a08d87ec59a03501dcd55413ce046eb.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.f36a9dba7b795e3b96df4bb6b480a688.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.b49469c77e102314d82036734a413360.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.221403deba39c28c50f332a8fde22646.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.77f89368e68665c1e3ffdc1d1bea7bab.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060 >>>>>> ;branch=z9hG4bK7014.2e5981568975ea5c81876a9dded1bc08.0# >>>>>> 015#012Via: SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.19809abdb9c351fdce7e25f51e8c1754.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.ad73d18d15b6c9da96e13156a84d5522.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.fcb5555d6a683215aa989fb4254abd62.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.244b467757383234c76b1c66f98ddfe8.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.b2c353ef1d637907359822c64cb81b2a.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.d22dceb1af0320cfc769f94e0214a313.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.f5e9b7f9b209c82971ae1017b9ad7d1e.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.0f2c98bc92cc799fdce0a4090b626021.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.aa8a07d89f83a26cefd74e4770aef63c.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.94c376a77d066ffe113e84465f5f5311.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.2202e658b1d8b801b14ec432bae7fabe.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.55f7d6a5d9d2a9d064cf1c6683f12eca.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.6870e53f3bc573f140d1728401dd43ae.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.a84d83df8646a75ee0d3605dcacb45ad.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.486b6fb0d17077e20f8375188a8bee75.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.bc0321f98578ab09ad615b634c4c83cd.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.9b26fda9326632abf7bdb32a33d03ab2.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.060fa4afc70848b34ede46705b90f7cb.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.56134adb7c492f7a4f03b36dfbf713f5.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.f9ccc5e81175283c9b4b56f4a2252c0f.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.3c231176b0e95b8121637df41b504f18.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.95bd5b0e0bf8c2bda906a751816a9876.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.87e0fd0a1bd044147d22052d27e07527.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.73:4080;branch=z9hG4bK7014.0a319c71a9a26c8afa77713d6d4ecfed.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.79:4070;branch=z9hG4bK7014.1442d975c0a393f232020916a6fbfc15.1#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.73:4080;branch=z9hG4bK7014.e35a84678ec61f7cefa8cdacb6ba92d9.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.179:4060;branch=z9hG4bK7014.cc99617f0c0f5088f45e203f6a58dd57.0#015#012Via: >>>>>> SIP/2.0/UDP 10.45.6.174:36029;received=10.45.6.174;branch=z9hG4bK-524287-1---c856c42a33d1e959;rport=36029#015#012To: >>>>>> <sip:bob@sip.example.com>;tag=8409f602ccf3f3f0da6ebceab6ff6336.aa5bd74f#015#012From: >>>>>> <sip:alice@sip.example.com;transport=UDP>;tag=cdbaae3f#015#012Call-ID: >>>>>> I-58e6z5tN3aDpmdzq1oVA..#015#012CSeq: 1 INVITE#015#012Server: TelcoSuite >>>>>> Proxy-CSCF#015#012Content-Length: 0 >>>>>> >>>>>> As you see here , continuously invite is going on looping in >>>>>> 10.45.6.179 IP which is pcscf. What could be the reason .. Kindly do the >>>>>> needful. >>>>>> >>>>>> On Wed, Apr 15, 2020, 2:19 PM David Villasmil < >>>>>> david.villasmil.work(a)gmail.com> wrote: >>>>>> >>>>>>> You should be tracing, you’re probably not, otherwise you would see >>>>>>> the server is probably forwarding the INVITE to itself (removing one hop >>>>>>> every time as it must) and thus creating a loop. >>>>>>> >>>>>>> On Wed, 15 Apr 2020 at 09:42, Pavithra M <pavimohan3096(a)gmail.com> >>>>>>> wrote: >>>>>>> >>>>>>>> Hi , >>>>>>>> Yes , I have tried uncommenting that part . It is actually checking >>>>>>>> for hostname (sip.example.com:5060) in my scscf node. I am not >>>>>>>> sure why it is looking for sip.example.com instead of >>>>>>>> scscf.sip.example.com:4080 >>>>>>>> >>>>>>>> I have added one more alias in my scscf.cfg file .Now Domain not >>>>>>>> served issue has gone. >>>>>>>> After that I am facing 483 Too Many Hops Error. >>>>>>>> >>>>>>>> I have added scscf debug logs. >>>>>>>> >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: DEBUG: <core> >>>>>>>> [core/mem/q_malloc.c:482]: qm_free(): qm_free(0x7fc57b362010, >>>>>>>> 0x7fc57b514138), called from ims_registrar_scscf: lookup.c: lookup(225) >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: DEBUG: <core> >>>>>>>> [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7fc57b514100 alloc'ed >>>>>>>> from ims_registrar_scscf: lookup.c: lookup(103) >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: ERROR: *** >>>>>>>> cfgtrace:request_route=[FINAL_TERM] c=[/etc/kamailio/scscf/kamailio.cfg] >>>>>>>> l=1041 a=16 n=if >>>>>>>> * Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: DEBUG: <core> >>>>>>>> [core/socket_info.c:628]: grep_sock_info(): checking if host==us: 10==10 && >>>>>>>> [10.45.6.75] == [10.45.6.73]* >>>>>>>> * Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: DEBUG: <core> >>>>>>>> [core/socket_info.c:635]: grep_sock_info(): checking if port 4080 >>>>>>>> (advertise 0) matches port 32347* >>>>>>>> * Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: DEBUG: <core> >>>>>>>> [core/forward.c:412]: check_self(): host != me* >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: ERROR: *** >>>>>>>> cfgtrace:request_route=[FINAL_TERM] c=[/etc/kamailio/scscf/kamailio.cfg] >>>>>>>> l=1050 a=5 n=route >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: ERROR: *** >>>>>>>> cfgtrace:request_route=[apply_privacy] c=[/etc/kamailio/scscf/kamailio.cfg] >>>>>>>> l=821 a=16 n=if >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: ERROR: *** >>>>>>>> cfgtrace:request_route=[apply_privacy] c=[/etc/kamailio/scscf/kamailio.cfg] >>>>>>>> l=818 a=25 n=is_present_hf >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: ERROR: *** >>>>>>>> cfgtrace:request_route=[FINAL_TERM] c=[/etc/kamailio/scscf/kamailio.cfg] >>>>>>>> l=1052 a=24 n=t_relay >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: DEBUG: tm >>>>>>>> [t_lookup.c:1328]: t_newtran(): msg (0x7fc57b502a70) id=2/7139 global >>>>>>>> id=2/7139 T start=(nil) >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: DEBUG: tm >>>>>>>> [t_lookup.c:497]: t_lookup_request(): start searching: hash=38849, isACK=0 >>>>>>>> * Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: DEBUG: tm >>>>>>>> [t_lookup.c:455]: matching_3261(): RFC3261 transaction matching failed - >>>>>>>> via branch [z9hG4bK1c79.03e4ec0c2ca15f1aa14c067b33a29f10.1]* >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: DEBUG: tm >>>>>>>> [t_lookup.c:675]: t_lookup_request(): no transaction found >>>>>>>> Apr 15 13:59:12 tel-VirtualBox kamailio[7139]: DEBUG: <core> >>>>>>>> [core/mem/q_malloc.c:374]: qm_malloc(): qm_malloc(0x7fc57478b000, 5720) >>>>>>>> called from tm: h_table.c: build_cell(330) >>>>>>>> >>>>>>>> I am facing host is not me error . Here 10.45.6.73 -> scscf and >>>>>>>> 10.45.6.75 -> UE (bob user) >>>>>>>> I am not sure what alias it is asking for . Kindly help. >>>>>>>> >>>>>>>> On Wed, Apr 15, 2020 at 1:36 PM Pafel <pafels(a)gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi, >>>>>>>>> >>>>>>>>> Most likely this is because of the change. Revert back the changes >>>>>>>>> (uncomment the section I told you), turn on debug in scscf.cfg, restart all >>>>>>>>> nodes and test once again. Then have a look on the scscf logs. >>>>>>>>> >>>>>>>>> Regards, >>>>>>>>> Pavel Siderov >>>>>>>>> _______________________________________________ >>>>>>>>> Kamailio (SER) - Development Mailing List >>>>>>>>> sr-dev(a)lists.kamailio.org >>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Kamailio (SER) - Development Mailing List >>>>>>>> sr-dev(a)lists.kamailio.org >>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> >>>>>>> David Villasmil >>>>>>> email: david.villasmil.work(a)gmail.com >>>>>>> phone: +34669448337 >>>>>>> _______________________________________________ >>>>>>> Kamailio (SER) - Development Mailing List >>>>>>> sr-dev(a)lists.kamailio.org >>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>>>>>> >>>>>> _______________________________________________ >>>>>> Kamailio (SER) - Development Mailing List >>>>>> sr-dev(a)lists.kamailio.org >>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>>>>> >>>>> -- >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work(a)gmail.com >>>>> phone: +34669448337 >>>>> _______________________________________________ >>>>> Kamailio (SER) - Development Mailing List >>>>> sr-dev(a)lists.kamailio.org >>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Development Mailing List >>>> sr-dev(a)lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>>> >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work(a)gmail.com >>> phone: +34669448337 >>> _______________________________________________ >>> Kamailio (SER) - Development Mailing List >>> sr-dev(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>> >> _______________________________________________ >> Kamailio (SER) - Development Mailing List >> sr-dev(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >> > -- > Regards, > > David Villasmil > email: david.villasmil.work(a)gmail.com > phone: +34669448337 > _______________________________________________ > Kamailio (SER) - Development Mailing List > sr-dev(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >

1 0

Releasing Kamailio 5.2.7
by Henning Westerholt 13 May '20

13 May '20

Hello, as discussed in the last development meeting, I am planning to release next week (likely on Wednesday 13th May) the next minor version out of branch 5.2, version 5.2.7. Backports of fixed issues from git master will be integrated as usual. If you are aware of new issues which were not yet reported, let us know (by creating an issue on the tracker). Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

"kamdbctl create" scrip error "Invalid default value for 'action'"
by Вячеслав Панасов 13 May '20

13 May '20

Hi all I trying to install the test`s kamailio from repo in the new instance on the CentOS 7 with MariaDB 5.5.6 and get this ERROR message [user@testkamailio-002] # kamdbctl create MySQL password for root: INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... INFO: creating standard tables into kamailio ... INFO: Core Kamailio tables succesfully created. Install presence related tables? (y/n): y INFO: creating presence tables into kamailio ... INFO: Presence tables succesfully created. Install tables for imc cpl siptrace domainpolicy carrierroute drouting userblacklist htable purple uac pipelimit mtree sca mohqueue rtpproxy rtpengine secfilter? (y/n): y INFO: creating extra tables into kamailio ... ERROR 1067 (42000) at line 1: Invalid default value for 'action' ERROR: Creating extra tables failed at secfilter! after if i try reinit i get this error [user@testkamailio-002] # kamdbctl reinit MySQL password for root: This will drop your current database and create a new one. It is recommended to first backup your database. Continue with reinit? (y/n): y INFO: Database kamailio deleted INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... ERROR 1396 (HY000) at line 1: Operation CREATE USER failed for 'kamailio'@ 'localhost' ERROR 1396 (HY000) at line 1: Operation CREATE USER failed for 'kamailioro'@ 'localhost' ERROR: granting privileges to database kamailio failed! two weeks ago, the installation was successful, apparently there were some changes in the script cheers Vjacheslav Panasov

2 1

Determine correct port in record-route if kamailio is behind NAT
by Michal Popovic 12 May '20

12 May '20

Hello, our kamailio used for sip trunk interconnections is behind NAT and our cloud provider opens random outgoing ports for outbound connections. Our record-route is set to our external address and port 5060, that is probably incorrect, but we did not had any issues. One of our partners suddenly begin sending BYEs to the port advertised in record-route instead of port from where he received call. What is a correct approach here if we are not able to determine open port behind NAT? Bye, Michal

3 4

How to delete all values for a certain xavp index
by Sergiu Pojoga 11 May '20

11 May '20

Hi there, For some reason I can't figure this seemingly simple Subj, may be someone can help. In my particular case, trying to delete Contact[0] from the xavp storing Contacts after t_load_contacts() $xavp(tm_contacts[0]) = $null; Above results in all but the latest added value being deleted. Assigning $null to index[1] results in all but the latest 2 being delete and so on. Thanks.

3 8

Digest authentication w/ Kamailio & Freeswitch
by Edward Romanenco 11 May '20

11 May '20

Hi! I've using a SIP setup that includes both Kamailio & Freeswitch, invites are passed from Freeswitch and relayed by Kamailio to various dispatchers, I would like to have Kamailio authenticating when Proxy Authentication is required. As I understood, this can be achieved with the help of a failure route, problem is, when I'm utilizing this method - the 407 response gets reverted back to Freeswitch, which returns the revised invite filled with the default Freeswitch username/password, how can let Kamailio handle the authentication once receiving the 407? Can I work straight without relying on a failure route, but having the Proxy Authentication header on my original invite? This is my relevant configuration - route[RELAY] { 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("KAM_AUTH"); } } if (!t_relay()) { sl_reply_error(); } exit; } failure_route[KAM_AUTH] { if(t_check_status("401|407")) { $avp(auser) = "xxx"; $avp(apass) = "yyy"; t_on_failure("OUTGOING_FAILURE"); uac_auth(); t_relay(); exit; } } Edward

2 2

How to get cause from reason header?
by Luis Rojas G. 11 May '20

11 May '20

Hello, I am sure that this is very easy to do, but I have been googling and I can't find a way. Having a Reason header, for instance : Reason: SIP;cause=Some_Cause;text="Some_text" How can I get the value of "Some_Cause"? I need to compare it to some specific numeric values... Thanks in advance, Luis -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

2 4

imc module
by nicola defilippo 11 May '20

11 May '20

Hi, i'm a newbie about kamailio, i'd like create an imc chat. I load the module but i don't understand what's the sip uri to use to create the chat (i understand that there is a sip uri for the chat manager, but i don't know what it is). I' became a bit crazy. Thanks for help and sorry for the stupid question (if it's). N.

2 2

Incorrect Load Calculation by Dispatcher Module in Kamailio
by harneet singh 11 May '20

11 May '20

Hi All, I have run into an issue with the Dispatcher Module in Kamailio, which apparently seems to be running into issues randomly while calculating the dispatcher load. I have submitted an issue wrt the same at https://github.com/kamailio/kamailio/issues/2322 with all the details and potential fix as well, based on analysis and tests. Can some one please take a look into the same? Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

1 0

How S-CSCF route SIP INVITEs as per iFC ?
by Hamid Hashmi 11 May '20

11 May '20

I have added an Application Server in iFC and try to route one user to that Application server for testing But INVITE to that subscriber following the default iFC. I am using ng-voice HSS Server. I have been trying to differentiate between iFCs for two different subscribers and unable to find the store location of SAA xml on S-CSCF and I am Unable to find xml file of required subscribers in wireshark traces as well because it says in SAR that USER_DATA_ALREADY_EXISTS and SAA didn't contain any XML file for that specific users. Furthermore if you can describe SIP INVITEs routing mechanism of S-CSCF, on which params or SIP Headers S-CSCF route INVITEs to Application Servers according to iFC ? Regards Hamid R. Hashmi

2 1

Video Call for deaf
by www . deaf - tv . com 11 May '20

11 May '20

hello, my Server CentOs 7 with kamailio installed, Chat write good! Video call how modules ?? Kamailio V 5.3.3 Regards Denny Brendel

2 1

How Kamailio processes SIP ACK on local 4XX reply?
by Denys Pozniak 11 May '20

11 May '20

Hello! I have some misunderstanding regarding the processing of SIP ACK on local Kamailio 4xx reply (eg 407 Proxy Authentication Required). In the picture below you can see that such a message just goes through the proxy and moreover gets into the *request_route* section. I expect this SIP ACK to be consumed by the sl module without hitting script logic. [image: image.png] I put the part below to the very top of the *request_route* section: *request_route* { if ( is_method("ACK") ) { t_check_trans(); $var(error) = $rc; xlog("L_ERR", "[CATCH_ACK] m=$rm cseq=$cs ru=$ru err=$var(error) [$ci]"); } .... And as a result, I detected in syslog: May 7 16:15:34 test-edge /usr/sbin/kamailio[11642]: ERROR: <script>: [CATCH_ACK] m=ACK cseq=153 ru=sip:mod_sofia@10.0.7.3:5060 err=-1 [051e7c50-0b10-1239-2996-001dd8b71cb2] *SIP re-INVITE (hold) from Phone:* 2020/05/07 16:15:34.629973 176.6.4.6:1024 -> 185.9.7.7:5060 INVITE sip:mod_sofia@10.0.7.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.111:5060 ;branch=z9hG4bK00475de2da8eea118ec67caaaddc98fb;rport Route: <sip:185.9.7.7;r2=on;lr=on;did=f98.f3f4> Route: <sip:10.0.7.5;r2=on;lr=on;did=f98.f3f4> Route: <sip:10.0.7.6;lr=on;did=f98.fbe> From: <sip:312222222222@10.0.7.6>;tag=00dfd3d8da8eea118ec67caaaddc98fb To: "Denys VOIP" <sip:5008@185.9.7.7>;tag=54Q0XNHgXBUvp Call-ID: 051e7c50-0b10-1239-2996-001dd8b71cb2 CSeq: 153 INVITE Contact: <sip:312222222222@192.168.11.111:5060 ;gr=00DCB21F-BD8E-EA11-8E88-7CAAADDC98FB> Content-Type: application/sdp Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK Max-Forwards: 70 Supported: 100rel, replaces, from-change, gruu User-Agent: SIPPER for PhonerLite Content-Length: 313 v=0 o=- 3858283927 2 IN IP4 192.168.11.111 s=SIPPER for PhonerLite c=IN IP4 192.168.11.111 t=0 0 m=audio 5062 RTP/AVP 8 0 9 107 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:782125221 a=sendonly *SIP 407 from Kamailio:* 2020/05/07 16:15:34.632298 185.9.7.7:5060 -> 176.6.4.6:1024 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.11.111:5060 ;branch=z9hG4bK00475de2da8eea118ec67caaaddc98fb;rport=1024;received=176.6.4.6 From: <sip:312222222222@10.0.7.6>;tag=00dfd3d8da8eea118ec67caaaddc98fb To: "Denys VOIP" <sip:5008@185.9.7.7>;tag=54Q0XNHgXBUvp Call-ID: 051e7c50-0b10-1239-2996-001dd8b71cb2 CSeq: 153 INVITE Proxy-Authenticate: Digest realm="185.9.7.7", nonce="XrQaXjfjyrftyrtiyrft7uO" Server: kamailio (5.2.2 (x86_64/linux)) Content-Length: 0 *SIP ACK from Phone:* 2020/05/07 16:15:34.672872 176.6.4.6:1024 -> 185.9.7.7:5060 ACK sip:mod_sofia@10.0.7.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.111:5060 ;branch=z9hG4bK00475de2da8eea118ec67caaaddc98fb;rport Route: <sip:185.9.7.7;r2=on;lr=on;did=f98.f3f4> Route: <sip:10.0.7.5;r2=on;lr=on;did=f98.f3f4> Route: <sip:10.0.7.6;lr=on;did=f98.fbe> From: <sip:312222222222@10.0.7.6>;tag=00dfd3d8da8eea118ec67caaaddc98fb To: "Denys VOIP" <sip:5008@185.9.7.7>;tag=54Q0XNHgXBUvp Call-ID: 051e7c50-0b10-1239-2996-001dd8b71cb2 CSeq: 153 ACK Content-Length: 0 *SIP ACK (unwanted) from Kamailio to Core:* 2020/05/07 16:15:34.675546 10.0.7.5:5060 -> 10.0.7.6:5060 ACK sip:mod_sofia@10.0.7.3:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.0.7.5;branch=z9hG4bK0d46.c33c3556c3cffbac49e717352b3575f6.0 Via: SIP/2.0/UDP 192.168.11.111:5060 ;received=176.6.4.6;branch=z9hG4bK00475de2da8eea118ec67caaaddc98fb;rport=1024 Route: <sip:10.0.7.6;lr=on;did=f98.fbe> From: <sip:312222222222@10.0.7.6>;tag=00dfd3d8da8eea118ec67caaaddc98fb To: "Denys VOIP" <sip:5008@185.9.7.7>;tag=54Q0XNHgXBUvp Call-ID: 051e7c50-0b10-1239-2996-001dd8b71cb2 CSeq: 153 ACK Content-Length: 0 loose-route: done *IP addresses and call-flow:* Core (10.0.7.6) --> [ Kamailio (10.0.7.5) -> Kamailio (185.9.7.7) ] --> Phone (176.6.4.6) [root@test-edge kamailio]$ kamailio -v version: kamailio 5.2.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES 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: unknown compiled on 19:29:40 Jun 20 2019 with gcc 4.8.5 -- BR, Denys Pozniak

2 1

kamdbctl create - failing create kamailio database on secfilter table action column default value
by Voip support 11 May '20

11 May '20

Hello Community, I had created for my solution an installer using ansible. It worked great all the time but suddenly it started to crash on step where i run the command : kamdbctl create root@nlb:/home/nlb# kamdbctl create -e \E[37;33mINFO: creating database kamailio ... mysql: [Warning] Using a password on the command line interface can be insecure. -e \E[37;33mINFO: granting privileges to database kamailio ... mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. -e \E[37;33mINFO: creating standard tables into kamailio ... mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. -e \E[37;33mINFO: Core Kamailio tables succesfully created. -e \E[37;33mINFO: creating presence tables into kamailio ... mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. -e \E[37;33mINFO: Presence tables succesfully created. -e \E[37;33mINFO: creating extra tables into kamailio ... mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. mysql: [Warning] Using a password on the command line interface can be insecure. ERROR 1067 (42000) at line 1: Invalid default value for 'action' -e \E[37;31mERROR: Creating extra tables failed at secfilter! Its saying some error about 'action' column default value. The installation is made using ubuntu 16.04 from repository: deb http://deb.kamailio.org/kamailio53 xenial main deb-src http://deb.kamailio.org/kamailio53 xenial main Maybe the issue is this change i dont know (it mention default value 0 for action column): https://github.com/kamailio/kamailio/commit/b0f1486f8a656922aac02020c9e3141… Best regards, Tom

2 1

DMQ Module Crash
by Ilie Soltanici 11 May '20

11 May '20

Hello, I'm trying to replicate a hash table to another 2 nodes by using DMQ Module. That's the related configuration: loadmodule "tm.so" loadmodule "tmx.so" loadmodule "dialog.so" loadmodule "dmq.so" loadmodule "htable.so" modparam("dmq", "server_address", "sip:MY_INTERNAL_IP_ADDR:5062") modparam("dmq", "notification_address", "sip:dmq.domain.local:5062") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 90) modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) modparam("htable", "htable", "test=>size=10;initval=0;dmqreplicate=1;") if(is_method("KDMQ")) { dmq_handle_message(); exit; } After starting Kamailio - everything seems to be working fine, Kamailio is able to discover other nodes and replications is working fine, until after a few minutes I'm getting such errors in the Kamailio Log file: May 8 17:14:26 kamailio kamailio[28064]: CRITICAL: <core> [core/mem/q_malloc.c:521]: qm_free(): BUG: freeing already freed pointer (0x7f0ac62493e0), called from dialog: dlg_var.c: free_local_varlist(145), first free dialog: dlg_var.c: set_dlg_variable_unsafe(205) - ignoring Then, if I'm getting any SIP messages from any Device - Kamailio is crashing with the following info in the log file: May 8 16:03:19 kamailio kam[10963]: CRITICAL: {REGISTER (1) | Seq=45534 | Source IP=192.168.1.5 (tcp) | Call ID=Kyb-KJjf.U9ttdOwQNYu6mUWCKb2bvk7} <core> [core/mem/q_malloc.c:150]: qm_debug_check_frag(): BUG: qm: prev. fragm. tail overwritten(0, 0)[0x7f62cc746a10:0x7f62cc746a48]! Memory allocator was called from dialog: dlg_var.c:147. Fragment marked by dialog: dlg_var.c:108. Exec from core/mem/q_malloc.c:511. May 8 16:03:21 kamailio kam[10966]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 59 May 8 16:03:21 kamailio kam[10966]: DEBUG: <core> [core/tcp_main.c:3825]: handle_ser_child(): dead child 37, pid 10963 (shutting down?) May 8 16:03:21 kamailio kam[10966]: DEBUG: <core> [core/io_wait.h:599]: io_watch_del(): DBG: io_watch_del (0xa9e1c0, 59, -1, 0x0) fd_no=48 called May 8 16:03:21 kamailio kam[10966]: DEBUG: <core> [core/tcp_main.c:3593]: handle_tcp_child(): dead tcp child 1 (pid 10963, no 37) (shutting down?) May 8 16:03:21 kamailio kam[10966]: DEBUG: <core> [core/io_wait.h:599]: io_watch_del(): DBG: io_watch_del (0xa9e1c0, 62, -1, 0x0) fd_no=47 called May 8 16:03:21 kamailio kam[10919]: ALERT: <core> [main.c:767]: handle_sigs(): child process 10963 exited by a signal 6 May 8 16:03:21 kamailio kam[10919]: ALERT: <core> [main.c:770]: handle_sigs(): core was generated May 8 16:03:21 kamailio kam[10919]: INFO: <core> [main.c:792]: handle_sigs(): terminating due to SIGCHLD May 8 16:03:21 kamailio kam[10919]: DEBUG: <core> [main.c:794]: handle_sigs(): terminating due to SIGCHLD May 8 16:03:21 kamailio kam[10966]: INFO: <core> [main.c:847]: sig_usr(): signal 15 received This is what i got from the dump core file: uri = {user = {s = 0x3 <Address 0x3 out of bounds>, len = -264573566}, passwd = {s = 0x788 <Address 0x788 out of bounds>, len = 8208}, host = {s = 0x393100000000 <Address 0x393100000000 out of bounds>, len = 1588950199}, port = {s = 0x2 <Address 0x2 out of bounds>, len = 24086112}, params = {s = 0x7ffcbb1600c0 "\200", len = -261173312}, sip_params = {s = 0x7ffcbb1601a8 "\300\317n\360b\177", len = 24086112}, headers = {s = 0x0, len = -261173312}, port_no = 424, proto = 47894, type = 32764, flags = (URI_SIP_USER_PHONE | unknown: 4031456108), transport = { s = 0x16f88cc "16:03:19 kamailio[10963]: CRITICAL: {REGISTER (1) | Seq=45534 | Source IP=192.168.1.5 (tcp) | Call ID=Kyb-KJjf.U9ttdOwQNYu6mUWCKb2bvk7} <core> [core/mem/q_malloc.c:150]: qm_debug_check_frag(): BUG: qm: pr"..., len = -261163168}, ttl = { s = 0x8 <Address 0x8 out of bounds>, len = -263517049}, user_param = { s = 0x16f88cc "16:03:19 kamailio[10963]: CRITICAL: {REGISTER (1) | Seq=45534 | Source IP=192.168.1.5 (tcp) | Call ID=Kyb-KJjf.U9ttdOwQNYu6mUWCKb2bvk7} <core> [core/mem/q_malloc.c:150]: qm_debug_check_frag(): BUG: qm: pr"..., len = -264350686}, maddr = { s = 0x3336393031000080 <Address 0x3336393031000080 out of bounds>, len = 8187}, method = {s = 0x0, len = 24086112}, lr = {s = 0x820ce8 "%s: %s%s(): cseq <%.*s>: <%.*s> <%.*s>\n", len = -1156185384}, r2 = {s = 0xb <Address 0xb out of bounds>, len = 21}, gr = { s = 0xffffffff <Address 0xffffffff out of bounds>, len = -264780793}, transport_val = {s = 0x1ff4 <Address 0x1ff4 out of bounds>, len = 24086720}, ttl_val = {s = 0x7f62f06f0760 <main_arena> "", len = 128}, user_param_val = {s = 0x16f88b0 "`\002", len = 144}, maddr_val = {s = 0x15 <Address 0x15 out of bounds>, len = -264573566}, method_val = {s = 0x788 <Address 0x788 out of bounds>, len = 8208}, lr_val = {s = 0x0, len = 1588950199}, r2_val = {s = 0x2 <Address 0x2 out of bounds>, len = 24086112}, gr_val = { s = 0x7ffcbb1601f0 "\006", len = -261173312}} next_hop = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x7f62f06eef20 <_nl_C_LC_NUMERIC+32> "", len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x7f620000000b <Address 0x7f620000000b out of bounds>, len = 0}, sip_params = { s = 0x7f62f03762cd <vfprintf+19661> "\200\275\360\372\377\377", len = 0}, headers = {s = 0x0, len = -1}, port_no = 25293, proto = 61495, type = 32610, flags = (URI_USER_NORMALIZE | URI_SIP_USER_PHONE), transport = {s = 0x7ffcbb160038 "\267t\265^", len = 24086112}, ttl = {s = 0xffffffff <Address 0xffffffff out of bounds>, len = 0}, user_param = {s = 0x7d62e8 "%s: %.*s%s%s(): preparing to run routing scripts...\n", len = 11}, maddr = {s = 0xbd <Address 0xbd out of bounds>, len = 3440}, method = {s = 0x7d631c "", len = -263511186}, lr = {s = 0x16f88cc "16:03:19 kamailio[10963]: CRITICAL: {REGISTER (1) | Seq=45534 | Source IP=192.168.1.5 (tcp) | Call ID=Kyb-KJjf.U9ttdOwQNYu6mUWCKb2bvk7} <core> [core/mem/q_malloc.c:150]: qm_debug_check_frag(): BUG: qm: pr"..., len = -261163168}, r2 = { s = 0x8 <Address 0x8 out of bounds>, len = -263517049}, gr = { s = 0x16f88cc "16:03:19 kamailio[10963]: CRITICAL: {REGISTER (1) | Seq=45534 | Source IP=192.168.1.5 (tcp) | Call ID=Kyb-KJjf.U9ttdOwQNYu6mUWCKb2bvk7} <core> [core/mem/q_malloc.c:150]: qm_debug_check_frag(): BUG: qm: pr"..., len = -264350686}, transport_val = { s = 0x7d62ef "s%s%s(): preparing to run routing scripts...\n", len = 8187}, ttl_val = {s = 0x0, len = -264345823}, user_param_val = {s = 0x3000000010 <Address 0x3000000010 out of bounds>, len = -1156184608}, maddr_val = { s = 0x7ffcbb160520 "\341+:\213'H\027\203\341+\342ޙ>\356|", len = 21}, method_val = {s = 0x7ffcbb160250 "\023", len = -1156186080}, lr_val = {s = 0x7f62f06eef80 <_nl_C_LC_TIME> "r\036K\360b\177", len = -1156185648}, r2_val = {s = 0x17ff710 "IST", len = -1156186048}, gr_val = {s = 0x0, len = -1156186058}} u = 0x400001ff4 By disabling DMQ Module i haven't noticed any crashes or issues like that, so i suppose that Kamailio is crashing because of this module, but i can be wrong. I've tried with 2 versions of Kamailio (5.2.6 and 5.3.4), both of them having the same issue. Thanks

2 3

db_redis and PUA
by Carsten Bock 08 May '20

08 May '20

Hi, quick question: Did anyone ever attempt to use PUA with the new db_redis? If yes, could someone share the definition for this? Thanks, Carsten -- Carsten Bock I CTO & Founder ng-voice GmbH Trostbrücke 1 I 20457 Hamburg I Germany T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock

3 2

RTCP
by David Villasmil 08 May '20

08 May '20

Hello guys, We have clients/vendor who exchange RTP directly to each other. But we'd like to get RTCP packets (and proxy them to the other side). Is this at all possible? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 5

Kamailio with dispatcher and asterisks real time
by ospos web 08 May '20

08 May '20

Hi all folks, Hello people, I'm new and I've been reading, so I have some but not much knowledge. I must integrate as work, two asterisk to the kamailio, a month ago I started with the real time guide [1] with only one kamailio, register, start the call, etc.. I am with the basics and it works. Now I must put two asterisk and using dispatcher, but although I read the documentation of the module.. use it and so then configure a list of file (with the asterisk where to dispatch) and not in the database.. MY HELP REQUEST QUESTION: How do I use dispatcher in load balancing mode, but taking in consideration my already working realtime-asterisk worling setup? THE PROBLEM: since with what I did it simply sends the call to both asterisk. i setup only 4 easy steps.. load module, then setup params (where i set the asterisk lists by file, and not db) and then added a "ds_select_dst(1, 4);" before the "route(RELAY);" line. please help! as i know.. the ds_select_dst(1, 4); prepare wicht asterisk will be choose to use and then when the routing RELAY happends no cares if in the realtime previously guide are a main asterisk? right? NOTE: I know it's a big world... but I can't keep reading the kamailio theory, i must goon forward due i have to lear later once are property working.. [1] https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

3 5

secfilter table error
by Social Boh 07 May '20

07 May '20

Hello, any help with this error? ERROR 1067 (42000) at line 1: Invalid default value for 'action' ERROR: Creating extra tables failed at secfilter! Kamilio version 5.3.3 compiled from source. Thank you -- --- I'm SoCIaL, MayBe

2 4

Kamailio v5.3.4 Released
by Daniel-Constantin Mierla 06 May '20

06 May '20

Hello, Kamailio SIP Server v5.3.4 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.4. For more details about version 5.3.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/05/kamailio-v5-3-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Calls from Kamailio to Teams
by sip user 06 May '20

06 May '20

Hi, I have connected Kamailio like SBC with Teams, and Calls from Teams to Kamailio works.. Now, I have problems with calls from Kamailio to Teams, not works. How have I send the call? I set record_route_preset, but how send the call to Teams? Thanks so much

2 4

Releasing Kamailio v5.3.4
by Daniel-Constantin Mierla 06 May '20

06 May '20

Hello, based on the discussions from yesterday's online devel meeting, I am planning to release next week (likely on Wednesday, May 6) the next minor version out of branch 5.3, respectively 5.3.4. Should anyone be aware of issues not reported to the bug tracker or commits for fixes in master not backported to 5.3 branch, let us know. For new issues, the best is to report them on the tracker. For commits, you can provide links/hashids or make pull requests. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Map Diversion to History-Info
by Duarte Rocha 05 May '20

05 May '20

Greetings, Does Kamailio have any method to transform an History-Info with redirecting info into a Diversion or the other way around? Best Regards

2 1

PUA and db_redis
by Carsten Bock 05 May '20

05 May '20

Hi, has anyone been using db_redis with the Presence/PUA modules? If yes, can you share your experience and the proper key definition? Thanks, Carsten -- Carsten Bock I CTO & Founder ng-voice GmbH Trostbrücke 1 I 20457 Hamburg I Germany T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock

1 0

Roadmap to the next major release - Kamailio v5.4.0
by Daniel-Constantin Mierla 05 May '20

05 May '20

Hello, I am going to send a few emails on different topics to inform the broader community on mailing lists about what was discussed during the last online development meeting that happened last Wednesday, so everyone becomes aware and add own opinions and suggestions. First here is about the next major release, which is considered for the mid of summer. To allow the usual 1-1.5 months of testing time frame, the development has to be frozen in the first part of June (more or less one month from now on). If there will be no other proposals in the near future that end up in postponing the release target date, then try to push to master branch (or do pull requests) any new features that you want in Kamailio 5.4.x in the next 4-5 weeks. Once the code is frozen, you would have to wait for another release cycle to get new features in a stable branch. During the online meeting there was no major missing feature reported, if anyone here knows any, describe it and maybe some developer has time to implement it. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

info: in-memory presentity records
by Daniel-Constantin Mierla 05 May '20

05 May '20

Hello, the presentity records (PUBLISH data) stored by the presence module needed always an external storage backend (sql/no-sql database), for subscriptions (active watchers) it was possible to use memory caching only. I recently added support for cache-only module for PUBLISH data as well, more details are presented in the news article: * https://www.kamailio.org/w/2020/05/in-memory-presentity-records/ That could help simplifying network architecture on distributed platforms leveraging cloud infrastructure and using replications mechanisms (e.g., dmq, evapi+pua_json, nsq, ...) between the Kamailio nodes. The impact on the code was quite significant, given that any PUBLISH triggers a series of internal tasks (apply xcap rules, update watchers, ...), therefore I am calling for community testing to be sure old operational modes were not affected as well as testing the new mode to discover eventual issues -- just open an item on bug tracker when a problem is discovered. Special credits to Fred Posner (lod.com) for supporting the development of this feature and to Emmanuel Schmidbauer for testing and troubleshooting in the early phase of development. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

rtpproxy and NAT
by Pavithra M 05 May '20

05 May '20

Hi, kamailio version - 5.3.3 I have configured rtpproxy in pcscf . rtpproxy 17002 1 0 18:40 ? 00:00:00 /usr/bin/rtpproxy -s udp:10.40.10.3 7722 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 10.45.4.22 -d DBUG LOG_LOCAL0 I have configured rtpproxy in kamailio.cfg as well. modparam("rtpproxy", "rtpproxy_sock", "udp:10.40.10.3:7722") 10.40.10.3 - private IP 10.45.4.22 - public IP I have enabled NAT configuration also. When i make a call from zoiper client , rtp packets are going to some other public ip Could anyone please help me in this issue. Kindly help.

1 3

load_crl takes 90 seconds to complete it execution with crl file size of 4MB !!!
by mahesh b 03 May '20

03 May '20

Hi All, Am using Kamailio 5.1.9 version, In my tls.cfg i have one client and server profile, along with default client and server profile. I have crl enabled for the non default client and server profile , the crl file size is 4 MB in my case. I have 22 child tcp process. With this what i observe is load_crl is taking close to 90 seconds to finish its execution and return. It is very clear from the code, its because of this for loop. procs_no=get_max_procs(); for(i = 0; i < procs_no; i++) { if (SSL_CTX_load_verify_locations(d->ctx[i], d->crl_file.s, 0) != 1) { ERR("%s: Unable to load certificate revocation list '%s'\n", tls_domain_str(d), d->crl_file.s); TLS_ERR("load_crl:"); return -1; } store = SSL_CTX_get_cert_store(d->ctx[i]); X509_STORE_set_flags(store, X509_V_FLAG_CRL_CHECK | X509_V_FLAG_CRL_CHECK_ALL); } Is there a way this can be enhanced or as per the current kamailio design this is a must to do for each and every profile and the child process array list. Please suggest, is it justifiable for load_crl to take 90 seconds , to complete its execution, with one client and server profile and 22 child tcp process and crl file size of 4MB. More info of my setup: ----------------------------- Kamailio 5.1.9 version is running on the below Linux Kernel version : 3.10.0-693.el7.x86_64 Centos version : CentOS Linux release 7.4.1708 (Core) CPU : 2 cores with model name : Intel(R) Core(TM) i7-6700 CPU @ 3.40GHz [root@miv5000 ~]# cat /proc/meminfo MemTotal: 3882076 kB MemFree: 811244 kB MemAvailable: 2320356 kB Openssl verison : OpenSSL 1.0.2k-fips 26 Jan 2017 Regards, Mahesh.B

2 2

Build-variable name for route[name]
by Karsten Horsmann 02 May '20

02 May '20

Hi List, my google skills dont help me. I know there is a kind of build-in variable that you can use for example to return the name of the route you are referring to right now. here some pseudo kamailio-scripting-language: route[FANCY_ROUTE] { xlog("L_INFO", "$??? something happened\n"; } I search for $??? (as placeholder) which should return the route-name "route[FANCY_ROUTE]" or something like that? Take care. -- Cheers *Karsten Horsmann*

3 7

WebRTC control plane: how do you handle trickle ICE?
by Anthony Alba 01 May '20

01 May '20

Hello, For those of you integrating with WebRTC and using a websocket control plane like SIP.js how do you handle trickle ICE? If you use trickle ICE any recommended RTP media servers (FreeSWITCH, Kurento, ....)? SIP.js does some trick like delay XXX ms before sending out the offer assuming the browser has collected all the ICE candidates, or maybe even wait for the IceGatheringDone event: what do you think of such a pattern? I see that rtpengine source code mentions "trickle" in a few places, but the rtpengine module doesn't make it clear how to send a ICE candidate for an existing call. Cheers Anthony Alba

3 2

Kamailio & rtpproxy with more than one external IP interfaces
by Leonid Fainshtein 30 Apr '20

30 Apr '20

Hi, My server has several external IP interfaces and one internal interface. For some calls I would like to handle RTP via Ext1<->Int interfaces and for other calls: via Ext2<->Int. In order to achieve this, I don't define the '-l' rtpproxy parameter, and depending on the call, I set the required IP address as a parameter of rtpproxy_manage(). I didn't succeed to get the desired result. On my testbed, I have the following interfaces: Ext1: 192.168.0.31 Ext2: 192.168.0.32 Int: 10.159.65.1 When I try to create Ext2<->Int (192.168.0.32<-> 10.159.65.1), the packets that accepted on the internal interface are sent out from the Ext1(192.168.0.31) instead of Ext2 (192.168.0.32). It looks like the decision about the source address is made based on the IP routing table information (?). Also, I would like to have an advertised address for at least one external interface. Is it achievable with just one instance of rtpproxy? Or, it is necessary to run two rtpproxies in the bridged mode wit configured advertize addresses? Thank you in advance, Leonid

3 7

Call from Asterisk - kamailio - Teams
by sip user 30 Apr '20

30 Apr '20

Hi... I have integrated Asterisk with Teams using Kamailio like SBC, thanks this list.. I can make call from Teams to Asterisk throught Kamailio, I have not Callerid from Teams but I'm working in this.. The problem that I have is call from Asterisk to Teams.. So, I have configurated a PSTN in Kamailio to my Asterisk. In Kamailio.cfg I have: * PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "IP ASERISK" desc "PSTN GW Address" pstn.gw_port = "PORT ASTERISK" desc "PSTN GW Port" In request_route, when I received an INVITE, I have: if (is_method("INVITE|SUBSCRIBE")) { if(src_ip==IP ASTERISK) { record_route_preset("centralita.arcanos.es:5061 ;transport=$ add_rr_param(";r2=on"); } else { route(PSTN); } } So, If I received an INVITE from a src_ip!=IP ASTERISK, I route to PSTN and call to my Asterisk(works, without callerid, but works). I don't know if there are any best way to do that.. If I received an INVITE from a src_ip==IP ASTERISK, I set record_route_preset, but how can I send the call to Teams? What is wrong? What are I doing wrong? Thanks

1 0

Time taken for tls_fix_domains_cfg initialization to complete is 8 seconds
by mahesh b 30 Apr '20

30 Apr '20

Hi All, I have 10 client profiles and 10 server profiles along with default client and server profile in my tls.cfg These are the numbers on time taken for tls_fix_domains_cfg initialization to complete based on the number of client and server profiles i have. Number of profiles Time diff (Sec) No of tcp child processes Start time End time 5 *3.99* 42 16:47:01.836 16:47:05.826 6 4.69 42 16:59:29.335 16:59:34.027 7 *5.347* 42 17:04:30.271 17:04:35.618 8 *6.10* 42 17:18:41.109 17:18:47.204 9 *6.614* 42 17:22:36.869 17:22:43.483 10 *7.379* 42 17:27:46.676 17:27:54.055 Kamailio 5.1.9 version is running on the below Linux Kernel version : 3.10.0-693.el7.x86_64 Centos version : CentOS Linux release 7.4.1708 (Core) CPU : 2 cores with model name : Intel(R) Core(TM) i7-6700 CPU @ 3.40GHz [root@miv5000 ~]# cat /proc/meminfo MemTotal: 3882076 kB MemFree: 811244 kB MemAvailable: 2320356 kB Openssl verison : OpenSSL 1.0.2k-fips 26 Jan 2017 Now my question is : Is it ok for the api to take "8 seconds" to complete initialization in case of having 10 client and 10 server profiles along with default profiles in tls.cfg ? The time units taken for initialization to complete should it be in the order of milliseconds range or in seconds range ? Are there any more parameters related to tls configuration like "crl " which would play a factor in time taken for the initialization to complete. Currently at my application level , timeout is happening as the initialization is taking more time when more number of profiles are enabled. Please share your inputs on whether it is justifiable with my setup cpu/memory, it is ok for tls_fix_domains_cfg to take 8 seconds for the initialization to complete. Regards, Mahesh.B.

1 0

Next online devel meeting - Wed, Apr 29, 2020, 14:00UTC
by Daniel-Constantin Mierla 29 Apr '20

29 Apr '20

Hello, in order to plan the roadmap to next major release, it is time to organize another online devel meeting, so the developers and community members can syncronize and plan what should be done till Kamailio v5.4 will be out. Because freenode.net service is now requiring user registration to join IRC channels on their servers, we are considering some alternatives that would allow guest participation, one of them being using a matrix chat room. More details about it will be posted soon. I created a wiki page for the event at: * https://www.kamailio.org/wiki/devel/irc-meetings/2020a Feel free to add there topics that you want to be discussed. The proposed date is next Wednesday, April 29, 2020, at 14:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

4 7

NAT and RTP
by Pavithra M 29 Apr '20

29 Apr '20

Hi, I have installed kamailio as a ims server in openstack and enabled WITH_NAT and rtpproxy . I have installed pcscf, icscf, scscf and hss in four different ubuntu instances . server - kamailio 5.3.3. When i make a call using zoiper client from external network , call is becoming a missed call without giving ACK and BYE. Openstack - pcscf - 10.40.10.3 (floating ip - 10.45.4.22) icscf - 10.40.10.12 scscf - 10.40.10.5 hss - 10.40.10.31 zoiper1 - 10.0.2.15 zoiper2 - 10.0.2.15 kindly help me in this regard. when i installed without using NAT and rtpproxy, the call went well. Since currently i am using Opensatck , it is my prerequisite to use zoiper client externally .so for me it is mandate to enable NAT and rtpproxy. Help me please!!!

1 0

change module parameter in routing logic
by Ali Taher 28 Apr '20

28 Apr '20

Hi, Is it possible to change a module parameter inside the routing logic ? For example I want to change sqlcon parameter of sqlops module based on the originating IP. Regards,

2 5

DNS SRV issue
by Melek Oktay 28 Apr '20

28 Apr '20

Hi All, I have a user, who has a SRV records like below, srv:betaproxy.siptelekom.com 1- SRV betaproxy siptelekom 4 hrs 20 0 5060 beta-proxy-dc01.siptelekom.com (X.Y.87.150) 2- SRV betaproxy siptelekom 4 hrs 10 0 5060 beta-proxy-dc02.siptelekom.com (X.Y.87.160) And I want to send the call to him/her, but one of the potential IP address could be failed. Assume that, first record (X.Y.87.150) is failed when I'm trying to send the packet. Later on, timeout occurs and our Kamailio try to send the packet to second one (X.Y.87.160) after period ot time. My question: Next time when I want to send another call to the same user, how Kamailio could use second (X.Y.87.160) SRV instead of trying to first one (X.Y.87.150) ? What is the elegant way to implement this ? Best Regards, Melek

1 0

RTP relay
by hugues 28 Apr '20

28 Apr '20

Hello I'm trying to run Kamailio in front of an asterisk server due to security issues. Both servers have a public ip address. It works fine but the RTP voice traffic goes directly from the VOIP client to the asterisk server without going through the kamailio server. I want to shut down the asterisk server so that it only talks to the kamailio server. UA -> Kamailio public IP addr <-> Asterisk public ip addr How do I force RTP traffic to go through kamailio. Apparently I don't have NAT so I don't need rtpproxy. But you need something that relays RTP traffic? If anyone has an idea I'll take it. Thank you for your help. Hugues. -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

4 4

DMQ broadcasting crashes kamailio
by SamyGo 28 Apr '20

28 Apr '20

Hi, Is there a way to broadcast KDMQ to the cluster but not expect a reply back !?as far as I've read the source code dmq_bcast_message is exactly like dmq_send_message in a way that it expects a callback to be executed on response i.e expects a reply. So, the situation I'm facing is I'm broadcasting message to cluster and I do not want a reply back. The following two options result in crash & core dump. 1 - If my script doesn't respond back, by use of dmq_handle_message, it marks the destined servers as "inactive" and stops usrloc sync process which isn't desirable. 2 - If I respond back with the dmq_handle_message it crashes the Kamailio which just received this broadcasted message. Here is how its done in script: *broadcasting message to cluster:* dmq_bcast_message("userOnline", "$fu", "text/plain"); *Receiving and handling a broadcast message:* route[DMQ_HANDLE] { if(!(is_method("KDMQ") || $rm == "KDMQ")) return; if(is_method("KDMQ") || $rm == "KDMQ"){ if($rU =~ "userOnline"){ //user came online in cluster, resume transactions if-any suspended $avp(remoteUser) = $rb; } dmq_handle_message(); exit; } } *Related log lines:* Apr 23 21:15:48 kamailio[916]: ALERT: <script>: [da2c1-2f499] ------ DMQ_HANDLE: UserOnline Event Received ------ Apr 23 21:15:48 kamailio[916]: DEBUG: dmq [message.c:53]: ki_dmq_handle_message_rc(): dmq_handle_message [KDMQ sip:userOnline@9.8.7.123:5060] Apr 23 21:15:48 kamailio[916]: DEBUG: dmq [message.c:66]: ki_dmq_handle_message_rc(): dmq_handle_message peer found: userOnline Apr 23 21:15:48 kamailio[916]: DEBUG: <core> [core/receive.c:437]: receive_msg(): request-route executed in: 401461 usec Apr 23 21:15:48 kamailio[935]: DEBUG: dmq [worker.c:87]: worker_loop(): dmq_worker [2 935] lock acquired and crash/segfault.. Core dump: https://pastebin.com/S7ekCPfF Any help or pointers to solve this would be really appreciated. Best Regards, Sammy

5 13

RFC: C standard requirements
by Daniel-Constantin Mierla 27 Apr '20

27 Apr '20

Hello, (cross-posting because it is relevant to know where users need to run Kamailio as well as what developers tend to use for coding, but replies can go to one or the other list). Somehow I tried to stay close to ANSI-C/C90 with C coding style, for the reason of being compatible as much as possible with old compiles and niche/embedded systems. But I noticed that there are couple of C99 extensions in the core, like inline functions, variadic macros (the debugging macros), use of 'long long' or line comments starting with "//" (not so many, but still such comment style is used). Although I haven't searched for occurrences in core, but I remember seeing designated initializers (or initializing a structure by field names). These are part of C99, as listed by: * https://en.wikipedia.org/wiki/C99#Design Although likely GCC had some of them like custom extensions before the standard was out. On the other hand, newer standard C11 made some of those extensions optional: * https://en.wikipedia.org/wiki/C11_(C_standard_revision)#Optional_features So, my email here is to figure out if we should consider C99 as the requirement for adding C code or be more restrictive because there are still cases of usage on very specific systems that do not have such C compiler. We can eventually make a list of what is accepted or not from the recent versions of C standard to stay as much as possible portable across Unix'es. >From develeopers perspective, it would be good to learn if they use (in other projects, etc.) extensions of the new standards that they found useful and simplify their programming work. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

how to use get my domain name in VIA header (INVITE) instead of IP address.
by Nasida Yuriy 27 Apr '20

27 Apr '20

Hi guys, It sounds simple but I can not get domain name in VIA header (INVITE) instead of IP address. It is just FS ---INVITE --> Kamailio --INVITE--> endpoint Kamailio gets INVITE, and adds one more VIA with it's IP address. How to get domain name instead ? Please advice.

6 12

Multi INVITE issue
by Pavithra M 27 Apr '20

27 Apr '20

Hi, I have configured kamailio IMS in vm based images. I have configured pcscf,icscf,scscf,hss and ue in five different vms.I am able to successfully make a call via IMS. But when making a INVITE from alice to bob, it goes via ue -> pcscf pcscf -> scscf scscf -> icscf icscf -> scscf scscf -> pcscf (here multiple INVITE is happening and giving 483 Too Many Hops error and then the call is going to UE successfully) pcscf -> ue Can anybody help me what am i missing . why multi INVITE is happening there alone and giving me too many hops issue. Even after that issue how the call reaches the other UE. Kindly anyone help me in this regard atleast where to check this scenario.

2 1

Dialog module problem.
by Voip support 24 Apr '20

24 Apr '20

Hello, i am using dialog module in-memory to get all dialogs and show in a website. I use for this kamctl dialog show which is run quite often for autorefresh. I noticed that some calls get hung on kamailio while on caller side and callee side calls are disconnected. I captured a wireshark trace and for same caller IP i get many good calls as well some bad calls (not disconnected properly). When comparing the BYE packets i could not notice any difference. However in the "BAD" calls it looks like the BYE from caller get stuck on kamailio and is not sent/forwarded to callee side. I get BYE retransmissions from caller and all BYE requests got stuck on kamailio. After few seconds Callee side sends BYE and that BYE also get stuck. I am wondering how to trace this issue (differently than taking a wireshark trace) as i dont see any difference in the good BYE and the bad BYE. It looks like for any reason from time to time BYE is not forwarded to the other side and when the other side sents BYE its also not forwarded to call initiator. Please give me a clue how to continue on this. I am using dispatcher module to send the calls. Best regards Tom

2 2

nat_uac_test(8)
by Sergiu Pojoga 24 Apr '20

24 Apr '20

When receiving an INVITE over a specific LTE carrier, I'm seeing 'c=IN IP4 192.0.0.4' in SDP, which isn't technically a RFC1918 or RFC6598 IP address and thus nat_uac_test(8) fails. What elegant workaround can be done to catch such specific cases? Thanks.

5 11

Kamailio like SBC with Teams
by sip user 24 Apr '20

24 Apr '20

Hello good morning ... I am new to this list and I was starting to mess with Kamailio, mainly to set it up as SBC against Teams, in this case. But I can't get it to work for me. If I launch a call from the Teams, in the Kamailio I see: 1.- In syslog: Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 52.114.7.24 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 4160, type 3 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 171:1857:1187, 30 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0xa1f240, 23, 2, 0x7f90f2438f80), fd_no=17 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0xa1f240, 23, -1, 0x0) fd_no=18 called Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:4219]: handle_tcpconn_ev(): tcp: DBG: sending to child, events 1 Apr 15 11:11:41 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:3902]: send2child(): selected tcp worker 1 10(23159) for activity on [tls:SBC_IP:5061], 0x7f90f2438f80 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1507]: handle_io(): received n=8 con=0x7f90f2438f80, fd=9 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: tls [tls_server.c:184]: tls_complete_init(): Using TLS domain TLSs<default> Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: tls [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake started Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f90f2438f80 n=1468 fd=9 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf= Apr 15 11:11:41 vps793907 kamailio[23122]: #026#003#003 Apr 15 11:11:41 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9c1700, 9, 2, 0x7f90f2438f80), fd_no=1 Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7f90f2438f80, FD 9 Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1291]: tcp_read_req(): tcp_read_req: EOF Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9c1700, 9, -1, 0x10) fd_no=2 called Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1434]: release_tcpconn(): releasing con 0x7f90f2438f80, state -1, fd=9, id=30 Apr 15 11:11:42 vps793907 kamailio[23122]: 10(23159) DEBUG: <core> [tcp_read.c:1435]: release_tcpconn(): extra_data 0x7f90f2432b40 Apr 15 11:11:42 vps793907 kamailio[23122]: 13(23167) DEBUG: <core> [tcp_main.c:3331]: handle_tcp_child(): handle_tcp_child: reader response= 7f90f2438f80, -1 from 1 Apr 15 11:11:42 vps793907 kamailio[23122]: 13(23167) DEBUG: tls [tls_server.c:604]: tls_h_close(): Closing SSL connection 0x7f90f2432b40 2.- With TCPDUMP: 11:13:09.311797 IP SBC_IP .1024 > SBC_IP .eu.sip-tls: Flags [S], seq 261244614, win 64240, options [mss 1440,nop,wscale 8,nop,nop,sackOK], length 0 11:13:09.311898 IP SBC_IP .eu.sip-tls > 52.114.76.76.1024: Flags [S.], seq 812357247, ack 261244615, win 29200, options [mss 1460,nop,nop,sackOK,nop,wscale 7], length 0 11:13:09.340358 IP 52.114.76.76.1024 > SBC_IP .eu.sip-tls: Flags [.], ack 1, win 2053, length 0 11:13:09.340560 IP 52.114.76.76.1024 > SBC_IP .eu.sip-tls: Flags [P.], seq 1:187, ack 1, win 2053, length 186 11:13:09.340578 IP SBC_IP .eu.sip-tls > 52.114.76.76.1024: Flags [.], ack 187, win 237, length 0 11:13:09.341361 IP SBC_IP .eu.sip-tls > 52.114.76.76.1024: Flags [P.], seq 1:1469, ack 187, win 237, length 1468 11:13:09.369606 IP 52.114.76.76.1024 > SBC_IP .eu.sip-tls: Flags [.], ack 1469, win 2053, length 0 11:13:12.451498 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [S], seq 309084204, win 64240, options [mss 1440,nop,wscale 8,nop,nop,sackOK], length 0 11:13:12.451587 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [S.], seq 3275066862, ack 309084205, win 29200, options [mss 1460,nop,nop,sackOK,nop,wscale 7], length 0 11:13:12.707119 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [.], ack 1, win 2053, length 0 11:13:12.707311 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [P.], seq 1:187, ack 1, win 2053, length 186 11:13:12.707345 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [.], ack 187, win 237, length 0 11:13:12.708273 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [P.], seq 1:1469, ack 187, win 237, length 1468 11:13:12.965101 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [.], ack 1469, win 2053, length 0 11:13:12.965156 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [F.], seq 187, ack 1469, win 2053, length 0 11:13:12.965688 IP SBC_IP .eu.sip-tls > 52.114.7.24.1216: Flags [F.], seq 1469, ack 188, win 237, length 0 11:13:13.220871 IP 52.114.7.24.1216 > SBC_IP .eu.sip-tls: Flags [.], ack 1470, win 2053, length 0 I can't "receive" anything. I have generated the certificates and configured in the tls.cfg of the Kamailio: [server:default] method = TLSv1.2 verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/ssl/cert.key certificate = /etc/letsencrypt/ssl/cert.crt ca_list = /etc/letsencrypt/ssl/ca.crt Within Kamailio itself I have it configured to return a 200 KeepAlive to Teams when it receives an OPTIONS: event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip: SBC_DNS :5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } And I have measured the record_route for this new one: record_route_preset(" SBC_DNS :5061;transport=tls", " SBC_IP :5061"); I have created a dispatcher.list: # setid(integer) destination(sip uri) flags (integer, optional), priority(int,opt), attrs (str,optional) 1:1:sip:sip.pstnhub.microsoft.com;transport=tls 0 3 socket=tls: SBC_IP :5061;ping_from= sip:SBC_DNS 2:1:sip:sip2.pstnhub.microsoft.com;transport=tls 0 2 socket=tls: SBC_IP :5061;ping_from=sip: SBC_DNS 3:1:sip:sip3.pstnhub.microsoft.com;transport=tls 0 1 socket=tls: SBC_IP :5061;ping_from=sip: SBC_DNS 4:1:sip:sip-all.pstnhub.microsoft.com;transport=tls 0 1 socket=tls: SBC_IP :5061;ping_from=sip: SBC_DNS I think that one of the problems is that I do not send the OPTIONS to the Teams well, since it is on their panel, it indicates that the SBC is INACTIVE. I don't know if you could help me straighten this out a bit ... Thank you so much for everything.. a greeting

5 15

Check if Kamailio is going to CANCEL branch
by Ivan Ribakov 24 Apr '20

24 Apr '20

Hi all, Upon receiving a provisional response from callee in a branch route, is there a way of telling if Kamailio is going to automatically CANCEL that branch because a final reply was received from another branch before that? The closest I got is finding “cancel_b_method” parameter of the TM module, but that is not what I’m after. Previously I was recommended to debug via logging onreply_route but I’m not really sure what information may contain hints of intention to send CANCEL. Does anyone have any suggestions/guesses in which direction I should be digging? Thanks, Ivan

1 0

[info] matrix chat room for kamailio
by Daniel-Constantin Mierla 23 Apr '20

23 Apr '20

Thanks to the efforts made by Fred Posner, a matrix.org server has been installed on Kamailio project infrastructure (matrix.kamailio.dev - server sponsored by lod.com), which enables us to create chat rooms as we need for our community of users and developers. It tries to be an alternative to the existing IRC #kamailio channel, given that a while ago freenode.net had to enforce user registration to join the channels on their service (not complaining about it, they did it for good reasons to protect against spammers and D/DoS attacks). The #kamailio channel on freenode.net will still be in use by the people that like it, there is no plan to consider it obsolete. One of the main benefits of running own matrix server is the ability to allow guest access during the devel online meetings, so people that do not want to create an account can still participate by using a web browser. Also, it offers modern encryption and privacy, along with richer collaboration features (e.g., file sharing), many considering it a viable open source alternative to Slack or other private team collaboration services. The first room created is #kamailio:matrix.kamailio.dev : * https://riot.kamailio.dev/#/room/#kamailio:matrix.kamailio.dev The matrix server is federated, so you can join if you have an account on any other public federated matrix server or one that you can host yourself. The server on matrix.kamailio.dev is not allowing public registrations. If you are a registered developer (have git commit access) and need a matrix account, let us know and we will find solutions. It is the first phase now and based on the interest from the community, its usage can be increased (e.g., creating rooms for development notifications (commits, pull requests, issue); bridge to irc channel; etc...) or the server can be shut down if proves not useful. We will try to do the next online devel meeting (scheduled for Wednesday, April 29, 2020, at 14:00UTC) on the matrix chat room mentioned above. More details at: * https://www.kamailio.org/w/2020/04/matrix-chat-room-for-kamailio/ Stay safe and healthy! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio, TSILO and "local" calls
by German Cancio 23 Apr '20

23 Apr '20

Dear All, does somebody know up to which point TSILO can handle calls from one UA to another UA, when both are registered on the same Kamailio instance? Is a routing through a backend (such as Asterisk) required in that case? thanks and best regards, Germán

2 2

Integrating Push Notifications with iOS
by German Cancio 22 Apr '20

22 Apr '20

Dear All, we are working on the integration of VoIP Push Notifications with iOS devices via Kamailio. From what we observe, the time window between receiving a VoIP APNS notification that wakes up the client app and the client app being sent back to the background by iOS is extremely narrow. It is around 1-2 seconds, with a (seemingly random) variance of say ~0.5s. So 1-2s is the time window available for getting the client app to properly REGISTER and to receive the INVITE back from Kamailio (using e.g. TSILO); otherwise the call is lost. (On Android, the equivalent time window extends to 10s or more.) Have others observed the same on iOS? We are using a client app that still uses the iOS 12 SDK (with Xcode10). Can we expect changes in that regard with iOS13/Xcode13? many thanks and cheers, Germán

3 5

doubt about xflags
by David Escartin 22 Apr '20

22 Apr '20

Dear all one quick question, reading the module corex doc, seems that xflag are message(transaction) flags. But I made a test and seems for some reason the flag is not seeing activated at the onreply_route, when it's activated on the request route. Seemed more like a script flag behaviour. Maybe I'm missing something? thanks a lot and regards david

2 8

Parallel forking - first responder wins
by Ivan Ribakov 21 Apr '20

21 Apr '20

Hi all, What I’m trying to achieve is to have Kamailio fork an INVITE to multiple endpoints in parallel but only maintain the branch that responds first (first to respond with 200 OK I guess). I’ve read the TM module documentation on forking (https://www.kamailio.org/docs/modules/stable/modules/tm.html#tm.serial_fork… <https://www.kamailio.org/docs/modules/stable/modules/tm.html#tm.serial_fork…>) and as far as I understood a combination of “seturi()” + “append_branch()” + “t_relay()” command calls will allow me to send multiple forked INVITEs in parallel. What I couldn't find information about in the documentation (please point me to it in case I missed it) is what controls (if any) do I have over forked requests. Do I need to keep track of the branches myself and cancel others when first succeeds or does Kamailio have some kind of setting for implementing such behaviour? Thanks, Ivan

7 12

NAT pinging after service restart
by Jacob Greene 21 Apr '20

21 Apr '20

Hello everyone! When I restart the Kamailio service, NAT pinging for all existing location records stops and does not start until the UA registers again. I'm using db_mode 2, and all of the registrations will be there, but for some reason Kamailio wont restart the pinging cycle. Is it possible to configure Kamailio to restart NAT pinging in the event of a service restart? Thanks for reading! Jake Greene Voice Engineer

1 1

Join us for SIP Chronicles Live #1, featuring Giovanni Maruzelli
by Maxim Sobolev 20 Apr '20

20 Apr '20

Dear Real-Time Friends and Colleagues! As many of you we have been totally devastated that we will have no chance to see you in the next few months to come. :-/ Some people in the community believe it might be years. I don’t necessarily agree with that opinion myself. Over the course of the last few years our team had a great time extending live coverage for some of those events that have been affected, got some experience and equipment. Instead of just waiting for the virus to clear, we decided to organize a series of bi-weekly live casts with some of the speakers that we have hoped to see at those events presenting their latest developments live and then answering questions from the audience. So without further ado, let me introduce our first guest Giovanni Maruzzelli, who is going to introduce his newest project SaraPhone ( https://github.com/gmaruzz/saraphone). Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask Giovanni a question about his project live: https://youtu.be/mF9elIcVGE8 Or if you miss that opportunity, you can always watch the recording later on Sippy Labs channel on YouTube and email Giovanni your question at < gmaruzz(a)gmail.com>. SaraPhone is a bare bone SIP WebRTC phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) Stay healthy, optimistic and productive! Also share, like and subscribe. See you soon!!! Regards, Max

1 0

INVITE forking - uniform way to identify branches
by Ivan Ribakov 20 Apr '20

20 Apr '20

Hi all, I have a prioritised list of endpoints, where endpoints with the same priority should be sent in parallel: - endpoint_1_uri; p = 1 - endpoint_2_uri; p = 2 - endpoint_3_uri; p = 4 - endpoint_4_uri; p = 5 As some of the lower priority endpoints are not known upfront, I can not load entire list into the contact list and use TM module's “forking based on Q value” to take care of the rest. So my plan is to load all endpoints with the same priority and fork them in parallel. If none succeed - obtain next endpoint(s) and repeat the process. As I go through the above process, I need to update some counters each time delivery to one of the endpoints succeeds or fails. To be able to perform these “housekeeping” tasks I need to be able to clearly identify branches and link them to the destination URI (endpoint URI) of the forked INVITE that started them. Ideally I would like to be able to maintain relationship “branch_id” => “destination URI” where “branch_id” is the identifier that is appended to Via branch parameter. What is the best/most reliable way to achieve this? I tried logging and observing the value of $du - it was available just before sending out a forked INVITE, but in branch_failure and onreply_route it was <null>. I tried logging and observing $T_req($du) - I only expected it to work from onreply_route, but it was also <null>. I tried logging and observing $T_branch_idx - while it was showing meaningful data before sending forked INVITE and during branch_failure route, I couldn’t access it from onreply_route so I couldn’t tell which of the branches current reply is for. Are there any more variables that could help me build “branch_id” => “destination URI” relationship and access information about “branch_id” and/or “destination URI” from branch_failure and onreply_route? Apologies in advance for a lengthy description, I did my best to summarise the situation and what I’m trying to achieve. Regards, Ivan

1 0

Re: [SR-Users] (no subject)
by ahmed moghazy 20 Apr '20

20 Apr '20

Excuse me sir, if i want to make integration between kamailio and asterisk to make kamailio could use land line , i followed steps but i got errors i want help , was asterisk is only way to make kamailio use landline (fxs,fxo) Sent from my iPhone > On Mar 10, 2020, at 9:52 PM, ahmed moghazy <ahmed.m.moghazi(a)gmail.com> wrote: >

2 1

LCR and fallback
by Edward Romanenco 20 Apr '20

20 Apr '20

Hi! I would like to implement LCR and fallback capabilities for my Kamailio installation. I'm currently using the dispatcher module with a simple configuration file. As I understood, there are multiple modules (carrier route, dynamic routing) that can be used to achieve that, what I'm aiming for is - * Country-based (or prefix-based) routing with up to 4 different carriers (could be expanded in the future, but I won't have more than a dozen), while also maintaining a fallback route if the 1st priority is not available. * Each of my carriers have his own requirements for headers and formatting, for instance, for one of our carriers, having a '+' prefix before dialing to a non-US number is mandatory, while on the others - it is not, or another example - one of my carriers doesn't allow having a P-Asserted-Identity header in my invite, while others do. I would love to have the option to manipulate headers based on the route that I have chosen, both on the default and fallback routes. * It would be nice to receive SMTP or SNMP alerts if a fallback does occur. * Easy management, having the ability to load and reload the routing table on a monthly basis, once the pricing list is changed. What would be your suggestion? Edward

2 1

How to install the PHONENUM module?
by Edward Romanenco 20 Apr '20

20 Apr '20

Hi, I'm trying to add the PHONENUM module into my Kamailio installation. For this, I've cloned the main branch of Libphonenumber<https://github.com/google/libphonenumber/tree/master/cpp> and followed the installation rules as they appear in the relevant README file: Building and testing the library -------------------------------- $ cd libphonenumber/cpp $ mkdir build $ cd build $ cmake .. $ make $ ./libphonenumber_test It all went through and the library was created, but when I try to make and install the module itself, I'm getting the following error. Looks like the library wasn't included in the building context, can you please lend me a hand and tell me how do I include it? root@kamaillioegress:/usr/local/src/kamailio-5.1/kamailio/src/modules/phonenum# make Compiling cphonenumber.cpp g++ -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -Wall -Wno-write-strings -Wno-deprecated -Wno-unused-function -Wno-sign-compare -Wno-strict-aliasing -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -Wall -DNAME='"kamailio"' -DVERSION='"5.2.0-dev6"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 5.4.0"' -D__CPU_x86_64 -D__OS_linux -DVERSIONVAL=5002000 -DCFG_DIR='"/usr/local/kamailio-5.1/etc/kamailio/"' -DRUN_DIR='"/var/run/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DMEM_JOIN_FREE -DF_MALLOC -DQ_MALLOC -DTLSF_MALLOC -DDBG_SR_MEMORY -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DUSE_SCTP -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_IP_MREQN -DUSE_RAW_SOCKS -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -I/opt/local/include -DKAMAILIO_MOD_INTERFACE -DMOD_NAME='"phonenum"' -c cphonenumber.cpp -o cphonenumber.o cphonenumber.cpp:24:65: fatal error: phonenumbers/geocoding/phonenumber_offline_geocoder.h: No such file or directory compilation terminated. Makefile:22: recipe for target 'cphonenumber.o' failed make: *** [cphonenumber.o] Error 1 Edward

2 1

Free Persian Asterisk Book and COVID-19 Pandemic
by Mojtaba 19 Apr '20

19 Apr '20

In the context of the COVID-19 pandemic, Nasim Telecom decided to publish the first six-captures of Persian Asterisk book that was written by me and Mr Najafi, published in 2017. Also it was introduced by Mr. David Duffett in AstriCon 2017 on Florida. It would be useful for all people which understand Persian language, Also all other people could use it by going code commands on it and examples. For downloading it go the following link: https://www.nasimtelecom.com/en/persian-asterisk-book/ So, be careful and safety at first, Stay at home and read your favourite books. With Best Regards. --Mojtaba Esfandiari.S

3 2

rtpenigne & STUN
by Ahed Saed 17 Apr '20

17 Apr '20

Hi, I'm using kamailio 5.3 & rtpengine ( both kamailio and rtpengine resides on public IP on same machine) am using below logic to handle clients behind NAT if(nat_uac_test("8")){ rtpengine_manage("SIP-source-address replace-origin replace-session-connection "); }else{ rtpengine_manage("trust-address replace-origin replace-session-connection "); } I don't want all the rtp packet to go over the server, i just want to make it use the rtpengine just in case it is needed. but am not sure how this can be done using kamailio, from the client side i'll try to use STUN, but i understand STUN doesn't work all the time ( ~90% of the cases it will work) can anyone help me with this ? another question - if one client works with STUN and the other is behind NAT how this should be handled? Cheers

1 0

TCP/TLS connection (id: 0) for WebSocket could not be found
by Mack Hendricks 17 Apr '20

17 Apr '20

Hey All, I get this message when trying to route request to a WebSocket client: Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: <core> [core/msg_translator.c:1762]: check_boundaries(): no multi-part body Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: <core> [core/msg_translator.c:429]: clen_builder(): content-length: 651 (651) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: WARNING: <core> [core/msg_translator.c:2786]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: <core> [core/msg_translator.c:2002]: build_req_buf_from_sip_req(): could not create Via header Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: tm [t_fwd.c:1738]: t_forward_nonack(): failure to add branches Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: tm [t_funcs.c:337]: t_relay_to(): t_forward_nonack returned error -2 (-2) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: tm [t_funcs.c:355]: t_relay_to(): -2 error reply generation delayed Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: exec: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=1176 a=24 n=sl_reply_error Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: No error (2/SL) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: exec: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=1178 a=2 n=exit Here is what my location table looks like. It should try to send the call to sip:98.209.240.245:56291;transport=ws *************************** 2. row *************************** id: 124 ruid: uloc-5e9685e8-5650-92 username: 2000 domain: ws-test.com contact: sips:2000@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss received: sip:98.209.240.245:56291;transport=ws path: NULL expires: 2020-04-15 05:03:17 q: -1.00 callid: 7ef0a550-b2b9-24c6-5c3b-8e0c667f5533 cseq: 50096 last_modified: 2020-04-15 04:59:57 flags: 0 cflags: 524352 user_agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 socket: NULL methods: NULL instance: NULL reg_id: 0 server_id: 0 connection_id: -1 keepalive: 1 partition: 18 The Websocket list looks like this: Server01:~# kamcmd ws.dump { connections: { 8: wss:98.209.240.245:56291 -> wss:134.122.27.49:4443 (state: OPEN, last used 47s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } I’m not sure what I’m doing wrong. Any ideas? Thanks in advance -Mack

2 8

Kamailio in osm
by Pavithra M 17 Apr '20

17 Apr '20

Hi, Do anybody have worked orchestrating kamailio as an IMS server through osm by launching vnfs in openstack . If so kindly reply me as I need some inputs on that.

1 0

Dealing with mobile client state while using Push Notification and TSILO
by Hubert Odziemczyk 17 Apr '20

17 Apr '20

Hello! We are trying to setup a call-flow with TSILO and Push Notifications for solution with a mobile SIP client (Linphone), based on presentation by Federico Cabiddu (http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-…). We have an issue with "managing" state of the client (active or not) when there is a call coming. To be more specific, lookup in location table doesn't always give a proper answer, as contact expiration is different from actual lifetime of the app (which also differs between iOS and Android). One of proposed solutions is setting Expires=1 for REGISTER and then always rely on Push Notifications, but this scenario seems to be unreliable with some clients that are in foreground, as Push Notification doesn't trigger REGISTER again. Question is: did you perceive similar problems? What are your solutions to deal with them? Thank you in advance! Best regards, Hubert

2 1

Announcing SaraPhone, SIP WebRTC Open Source business phone
by Giovanni Maruzzelli 16 Apr '20

16 Apr '20

My fellow VoIPers, I am pleased to announce the early availability of: SaraPhone ------------------ SaraPhone is a bare bone SIP WebRTC voice phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, Kamailip, OpenSIPS, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) You an find it in GitHub ( https://github.com/gmaruzz/saraphone ). Anyone interested can play with it :). Have fun, giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

2 2

kamailio integration with MS teams.
by Nasida Yuriy 16 Apr '20

16 Apr '20

Hi guys, Do you have some expirience with kamailio integration with MS teams ? I follow the instructon from here https://skalatan.de/en/blog/kamailio-sbc-teams TLS part is configured correctly. I also got OPTIONS pings working between MS teams and kamailio by following of this instruction. That is very good. But... There is issues with outbound calls from MS teams to kamailo. MS side send INVITE, kamailio responds with 180 and 200 OK, but looks like MS ignores us. I know that I probably should use record_route_preset function here to modify Record-Route headers to satisfy MS wishes. But nothing helps. I also tried to modify contact header in 200 OK like it was done for OPTIONS - no luck. Anyone has a working example of kamailio files for MS team ? Or at least sucessful pcap with them. I can be wrong but looks like MS doesn't respect RFC at all. Thanks. Regards, Team lead<https://www.odesk.com/companies/Nasida_~0168bc9fd46660bcca> [https://odesk-prod-portraits.s3.amazonaws.com/Companies:1712760:CompanyLogo…]

4 6

Weird dialplan result
by David Villasmil 15 Apr '20

15 Apr '20

Hello guys, So i have the following dialplan (almost literally, 2 records) +----+------+-----+----------+-----------+-----------+-----------+----------+---------------------------------------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+-----+----------+-----------+-----------+-----------+----------+---------------------------------------+ | 10 | 1 | 100 | 0 | 910000000 | 0 | | | 1.2.3.4:5060 | | 11 | 1 | 100 | 0 | 910000001 | 0 | | | sip.somedomain.net:5060 | And with kamcmd: kamcmd> dialplan.translate 1 s:910000000 { Output: Attributes: 1.2.3.4:5060 } and kamcmd> dialplan.translate 1 s:910600000 error: 500 - No translation So that works... now, when i get an INVITE I do: dp_match("1", "$rU"); xlog("[DIALPLAN] '$rU' was translated to '$var(destination)'\n"); if ( $var(destination)!=0 ) { $ru = "sip:" + $fU + "@" + $var(destination); route(RELAY); } And this prints: <script>: [DIALPLAN] '910600000' was translated to '195.114.216.188:5060' What am I doing wrong? Kind of confused here... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

Kamailio propagates 180 and 200 OK OUT OF ORDER
by Luis Rojas G. 15 Apr '20

15 Apr '20

Good day, I am testing the dispatcher module, using Kamailio as stateless proxy. I have a pool of UAC (scripts in SIPP) and a pool of UAS (also scripts in SIPP) for the destinations. Kamailio version is kamailio-5.3.3-4.1.x86_64. Problem I have is, if UAS responds 180 and 200 OK to Invite immediately, sometimes they are propagated out of order. 200 OK before 180, like this : UAS is 172.30.4.195:5061. UAC is 172.30.4.195:5080. Kamailio is 192.168.253.4:5070 Difference between 180 and 200 is just about 50 microseconds. My guess is that both messages are received by different instances of Kamailio, and then because of context switches, even though the 180 is received before, that process ends after the processing of 200. However, I had the idea that in order to avoid these problems the kamailio processes synchronized with each other using a shared memory. I tried using stateful proxy and I obtained the same result. By the way, anyone has any idea about how Kamailio's share memory is implemented? It clearly does not use the typical system calls shmget(), shmat(), because they are not shown by ipcs command. Before posting here I googled, but I couldn't find anything related to this. I can't believe I am the only one who ever had this problem, so I guess I am doing something wrong... Please, any help. I'm really stuck on this. Thanks. -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

6 11

Re: [SR-Users] [EXT] UE client for IMS
by Woscek, Martin W. 15 Apr '20

15 Apr '20

https://code.google.com/archive/p/boghe/downloads Starting with version labeled: Boghe_2.0.113.744.sfx.exe<https://storage.googleapis.com/google-code-archive-downloads/v2/code.google…> works with IPv6. From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Pavithra M Sent: Tuesday, April 14, 2020 4:28 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [EXT] [SR-Users] UE client for IMS Hi, Could anyone suggest me what UE client can be used to test the normal ims call flow . Currently I am using zoiper client and facing issues of 403 forbidden - domain not served . So kindly help me in this regard so that if I change my UE client it works . Thanks

1 0

UE client for IMS
by Pavithra M 15 Apr '20

15 Apr '20

Hi, Could anyone suggest me what UE client can be used to test the normal ims call flow . Currently I am using zoiper client and facing issues of *403 forbidden - domain not served *. So kindly help me in this regard so that if I change my UE client it works . Thanks

2 1

kamailio integration with MS teams.
by Yuriy Nasida 14 Apr '20

14 Apr '20

Do you have some expirience with kamailio integration with MS teams ? I follow the instructon from here https://skalatan.de/en/blog/kamailio-sbc-teams TLS part is configured correctly. I also got OPTIONS pings working between MS teams and kamailio by following of this instruction. That is very good. But... There is issues with outbound calls from MS teams to kamailo. MS side send INVITE, kamailio responds with 180 and 200 OK, but looks like MS ignores us. I know that I probably should use record_route_preset function here to modify Record-Route headers to satisfy MS wishes. But nothing helps. I also tried to modify contact header in 200 OK like it was done for OPTIONS - no luck. Anyone has a working example of kamailio files for MS team ? Or at least sucessful pcap with them. I can be wrong but looks like MS doesn't respect RFC at all. Thanks.

1 0

H264 codec licence
by Prakash Ganesan 14 Apr '20

14 Apr '20

Hi, We are planning to use kamilio/RTP engine for our WebRTC calls. I would like to know how the H264 codec licences are handled as i know there is a royalty fee associated with this video codec from the server perspective. As of now there is plan to use H264 pass through and vp8/vp9 transcoded to H264 by the kamilio/RTP engine. -- Cheers Prakash

2 1

Using dispatcher module : Modifying message before dispatching with send_udp()
by Luis Rojas G. 13 Apr '20

13 Apr '20

Hello, I am using the dispatcher module, Kamailio version 5.3.3. The point is, I am using send_udp() to forward the message, because I don't want Kamailio to add headers Via nor Record-Route. I am just dispatching the Invite message, and the rest of communication to be made just between the endpoints, not through Kamailio. However, I need to add before a custom header, using append_hf(). It seems, send_udp() really changes nothing at all in the message, not even something added by the script with append_hf(). It works if I use forward(), but not if I use send_udp(). Is there a way to accomplish this? Best regards, -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

2 2

Lifetime of Invite with no Ack
by Duarte Rocha 13 Apr '20

13 Apr '20

Greetings, How much time will Kamailio keep a transaction active if the ACK to a INVITE is never received by Kamailio? Best Regards,

3 7

Using Stir/Shaken with Kamailio
by Alexandru Tripon 13 Apr '20

13 Apr '20

Hi, After I watched the talk about Stir/Shaken(https://vimeo.com/396940577), I wanted to try the module(secsipid), but I couldn't make it work. I built the Kamailio from source using the following steps: https://kamailio.org/docs/tutorials/devel/kamailio-install-guide-git/ . And then built the secsipid following the instruction found here: https://www.kamailio.org/docs/modules/devel/modules/secsipid.html#idm90 . But when I start the Kamailio with the module loaded I get the following error: ` kamailio: ERROR: <core> [core/sr_module.c:325]: version_control(): no version info in module </usr/local/lib64/kamailio/modules/secsipid.so>: /usr/local/lib64/kamailio/modules/secsipid.so: undefined symbol: module_version ` How can I fix this? Thank you in advance, Alexandru Tripon

2 2

Callee BYE routing issue
by Daniel W. Graham 13 Apr '20

13 Apr '20

Have an issue with BYE routing in a lab setup, any suggestions on what I should be looking at? Topology: UA1 <-> NAT device 1 <-> Proxy1 <-> Registrar/B2BUA <-> Proxy1 <-> NAT device 1 <-> UA2 All UA to UA calls flow through B2BUA. All IP’s are public except for UA1 and UA2. -Both endpoints can call each other. -If the caller hangs up first, BYE is routed properly. -If the callee hangs up first, BYE causes proxy to send 478. On calls where BYE results in 478: if(!isdsturiset()) { ……….…………………….not executed handle_ruri_alias() } return; Example issue call: UA2 to UA1, UA1 hang-up. Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[DLGURI] c=[/usr/local/etc/kamailio/kamailio.cfg] l=540 a=16 n=if Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[DLGURI] c=[/usr/local/etc/kamailio/kamailio.cfg] l=532 a=24 n=isdsturiset Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[DLGURI] c=[/usr/local/etc/kamailio/kamailio.cfg] l=540 a=2 n=return ……….. Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} tm [ut.h:245]: uri2dst2(): bad_uri: [sip::;transport=] Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} tm [t_fwd.c:1732]: t_forward_nonack(): failure to add branches Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=275 a=24 n=sl_reply_error Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} sl [sl_funcs.c:401]: sl_reply_error(): stateless error reply used: Unresolvable destination (478/SL) Apr 10 00:31:28 edgedev /usr/local/sbin/kamailio[5317]: ERROR: {1 2 BYE 24dc84a4592515245177483c2a657042@B2BUA_IP:5160} *** cfgtrace:dbg_cfg_trace(): request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=277 a=2 n=exit 2020/04/10 00:31:28.625060 NAT_Device_IP:57808 -> Proxy_IP:5060 BYE sip:398@B2BUA_IP:5160 SIP/2.0 Via: SIP/2.0/UDP UA1_IP:52884;branch=z9hG4bK-524287-1---43b5f31829f20952 Max-Forwards: 70 Route: <sip:Proxy_IP;lr=on;ftag=as78671fd9;did=b1b.a27> Contact: <sip:399@UA1_IP:52884> To: "Dan Test" <sip:398@B2BUA_IP:5160>;tag=as78671fd9 From: <sip:399@UA1_IP:52884;rinstance=705e0af5f4be485c;alias=NAT_Device_IP:~57808~1>;tag=64f4b712 Call-ID: 24dc84a4592515245177483c2a657042@B2BUA_IP:5160 CSeq: 2 BYE User-Agent: xxxxxxxxxxxxxxxx Content-Length: 0 ######################################### For comparison, here is a BYE from another test call: UA1 to UA2, UA1 hang-up. This BYE was routed correctly. 2020/04/10 00:44:02.688608 NAT_Device_IP:57808 -> Proxy_IP:5060 BYE sip:398@B2BUA_IP:5160 SIP/2.0 Via: SIP/2.0/UDP UA1_IP:52884;branch=z9hG4bK-524287-1---ce36df02e8a80a48 Max-Forwards: 70 Route: <sip:Proxy_IP;lr=on;ftag=9078f232;did=49e.8c01> Contact: <sip:399@UA1_IP:52884> To: <sip:398@B2BUA_IP>;tag=as5874901f From: <sip:399@B2BUA_IP:5060>;tag=9078f232 Call-ID: 103104NGU1NjkxNGZkYmM4MzA3Y2FkMzY1OGNkZTZmMTMyZDU CSeq: 3 BYE User-Agent: xxxxxxxxxxxxxxxx Authorization: Digest username= xxxxxxxxxxxxxxxx Content-Length: 0 -dan

4 7

Kamailio as IMS (version - 5.3.3) Error : 403 Forbidden Domain not served
by Pavithra M 13 Apr '20

13 Apr '20

Hi David, Kindly help me in this regard. I am not getting like it is because of pval issue or some other issue which causes this issue. I have configured kamailio as an IMS server as VM based . PCSCF - 10.x.x.179 ICSCF - 10.x.x.79 SCSCF - 10.x.x.73 HSS - 10.x.x.75 I am using zoiper client for testing the call flow . The *registration is happening properly* in my both zoiper clients for hss configured users (alice and bob). When i am making an *INVITE , *I am getting *403 Forbidden - Domain not served *error from SCSCF. The Call Flow for invite is as follows . call from alice(a)sip.example.com zoiper1 (UE) -> PCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) PCSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> PCSCF (100 trying) SCSCF -> ICSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) ICSCF -> SCSCF (100 trying) ICSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> ICSCF (100 trying) SCSCF -> ICSCF (*403 Forbidden - Domain not served*) ICSCF -> SCSCF (Request: ACK sip:bob@sip.example.com | ) ICSCF -> SCSCF ( 403 Forbidden - Domain not served ) SCSCF -> PCSCF ( 403 Forbidden - Domain not served ) PCSCF -> UE ( 403 Forbidden - Domain not served ) Below are the alias configuration i have done in pcscf,icscf,scscf nodes *PCSCF.CFG* # IP-Adress for incoming SIP-Traffic, in the following format: # SIP / UDP listen=udp:10.x.x.179:4060 # SIP / TCP listen=tcp:10.x.x.179:4060 # SIP / TCP/TLS listen=tls:10.x.x.179:4060 # IPSEC / UDP #!define IPSEC_LISTEN_ADDR "0.0.0.0" #!define IPSEC_CLIENT_PORT 5062 #!define IPSEC_SERVER_PORT 5063 #alias="sip.example.com:4060" alias=pcscf.sip.example.com:4060 #!define MY_WS_PORT 80 #!define MY_WSS_PORT 443 #!define PCSCF_URL "sip:pcscf.sip.example.com:4060" #!define TCP_PROCESSES 8 #!subst "/NETWORKNAME/sip.example.com/" #!subst "/HOSTNAME/pcscf.sip.example.com/" *ICSCF.CFG* # SIP / UDP listen=udp:10.x.x.79:4070 # SIP / TCP listen=tcp:10.x.x.79:4070 # SIP / TCP/TLS listen=tls:10.x.x.79:4071 alias="icscf.sip.example.com:4070" #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "icscf.sip.example.com" *SCSCF.CFG* # SIP / UDP listen=udp:10.x.x.73:4080 # SIP / TCP listen=tcp:10.x.x.73:4080 # SIP / TCP/TLS listen=tls:10.x.x.73:4081 #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "scscf.sip.example.com" #!define URI "sip:scscf.sip.example.com:4080" alias=scscf.sip.example.com:4080 # ENUM-Server to query: #!define ENUM_SUFFIX "scscf.sip.example.com." Kindly let me know where i am missing . Kindly need help in this issue since the registration is happening successfully and dropping in the INVITE level. Kindly help.

1 0

kamailio error
by Pavithra M 13 Apr '20

13 Apr '20

Hi, This is the lvalue.c file where i am facing the issue. kindly help. if (unlikely(ret<0)){ LM_ERR("assignment failed at pos: (%d,%d-%d,%d)\n", rve->fpos.s_line, rve->fpos.s_col, rve->fpos.e_line, rve->fpos.e_col); } else { if(unlikely(_log_assign_action!=NULL)) _log_assign_action(msg, lv); } rval_destroy(rv); return ret; error: if (rv) rval_destroy(rv); return -1;

1 0

Kamailio - 5.3.3 Error : 403 domain not served
by Pavithra M 13 Apr '20

13 Apr '20

Hi , I have configured kamailio as an IMS server as VM based . PCSCF - 10.x.x.179 ICSCF - 10.x.x.79 SCSCF - 10.x.x.73 HSS - 10.x.x.75 I am using zoiper client for testing the call flow . The *registration is happening properly* in my both zoiper clients for hss configured users (alice and bob). When i am making an *INVITE , *I am getting *403 Forbidden - Domain not served *error from SCSCF. The Call Flow for invite is as follows . call from alice(a)sip.example.com zoiper1 (UE) -> PCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) PCSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> PCSCF (100 trying) SCSCF -> ICSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) ICSCF -> SCSCF (100 trying) ICSCF -> SCSCF ( INVITE sip:bob@sip.example.com;transport=UDP ) SCSCF -> ICSCF (100 trying) SCSCF -> ICSCF (*403 Forbidden - Domain not served*) ICSCF -> SCSCF (Request: ACK sip:bob@sip.example.com | ) ICSCF -> SCSCF ( 403 Forbidden - Domain not served ) SCSCF -> PCSCF ( 403 Forbidden - Domain not served ) PCSCF -> UE ( 403 Forbidden - Domain not served ) Below are the alias configuration i have done in pcscf,icscf,scscf nodes *PCSCF.CFG* # IP-Adress for incoming SIP-Traffic, in the following format: # SIP / UDP listen=udp:10.x.x.179:4060 # SIP / TCP listen=tcp:10.x.x.179:4060 # SIP / TCP/TLS listen=tls:10.x.x.179:4060 # IPSEC / UDP #!define IPSEC_LISTEN_ADDR "0.0.0.0" #!define IPSEC_CLIENT_PORT 5062 #!define IPSEC_SERVER_PORT 5063 #alias="sip.example.com:4060" alias=pcscf.sip.example.com:4060 #!define MY_WS_PORT 80 #!define MY_WSS_PORT 443 #!define PCSCF_URL "sip:pcscf.sip.example.com:4060" #!define TCP_PROCESSES 8 #!subst "/NETWORKNAME/sip.example.com/" #!subst "/HOSTNAME/pcscf.sip.example.com/" *ICSCF.CFG* # SIP / UDP listen=udp:10.x.x.79:4070 # SIP / TCP listen=tcp:10.x.x.79:4070 # SIP / TCP/TLS listen=tls:10.x.x.79:4071 alias="icscf.sip.example.com:4070" #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "icscf.sip.example.com" *SCSCF.CFG* # SIP / UDP listen=udp:10.x.x.73:4080 # SIP / TCP listen=tcp:10.x.x.73:4080 # SIP / TCP/TLS listen=tls:10.x.x.73:4081 #!define NETWORKNAME "sip.example.com" #!define HOSTNAME "scscf.sip.example.com" #!define URI "sip:scscf.sip.example.com:4080" alias=scscf.sip.example.com:4080 # ENUM-Server to query: #!define ENUM_SUFFIX "scscf.sip.example.com." Kindly let me know where i am missing . Kindly need help in this issue since the registration is happening successfully and dropping in the INVITE level. Kindly help.

1 0

Kamailio 5.3.2 crashing continuously
by harneet singh 13 Apr '20

13 Apr '20

Hi All, As part of our very low cps load at 50 calls per second through Kamailio Version 5.3.2 (Dispatcher Module used with Call Load based Routing - Algorithm 10. Machine is a Centos: 7.7 with kernel version 3.10), we are seeing the kamailio continuously crashing. Please see the data below for 3 types of segfaults seen: Crash-1: -------- Apr 3 19:32:25 FE-A07-34-VM6 kernel: kamailio[10382]: segfault at 7ff300000078 ip 00007ff300000078 sp 00007ffe5f2eafc8 error 14 in libbz2.so.1.0.6[7ff331582000+f000] Apr 3 19:32:25 FE-A07-34-VM6 mysqld: 2020-04-03 19:32:25 6131 [Warning] Aborted connection 6131 to db: 'kamailio' user: 'kamailio' host: 'localhost' (Got an error reading communication packets) Apr 3 19:32:25 FE-A07-34-VM6 /usr/sbin/kamailio[10423]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 13 Apr 3 19:32:25 FE-A07-34-VM6 /usr/sbin/kamailio[10381]: ALERT: <core> [main.c:767]: handle_sigs(): child process 10382 exited by a signal 11 Apr 3 19:32:25 FE-A07-34-VM6 /usr/sbin/kamailio[10381]: ALERT: <core> [main.c:770]: handle_sigs(): core was not generated =============================== Crash-2: ------- Apr 3 19:36:11 FE-A07-34-VM6 kernel: kamailio[12838]: segfault at 7efe00000078 ip 00007efe00000078 sp 00007ffdc876b868 error 15 in zero (deleted)[7efdc407f000+40000000] Apr 3 19:36:11 FE-A07-34-VM6 mysqld: 2020-04-03 19:36:11 6175 [Warning] Aborted connection 6175 to db: 'kamailio' user: 'kamailio' host: 'localhost' (Got an error reading communication packets) Apr 3 19:36:11 FE-A07-34-VM6 /usr/sbin/kamailio[12874]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 18 Apr 3 19:36:11 FE-A07-34-VM6 /usr/sbin/kamailio[12826]: ALERT: <core> [main.c:767]: handle_sigs(): child process 12838 exited by a signal 11 Apr 3 19:36:11 FE-A07-34-VM6 /usr/sbin/kamailio[12826]: ALERT: <core> [main.c:770]: handle_sigs(): core was not generated ============================ Crash-3: -------- Apr 3 19:40:53 FE-A07-34-VM6 kernel: kamailio[13542]: segfault at 80 ip 000000000065e193 sp 00007ffdae10c1f0 error 4 in kamailio[400000+476000] Apr 3 19:40:53 FE-A07-34-VM6 mysqld: 2020-04-03 19:40:53 6222 [Warning] Aborted connection 6222 to db: 'kamailio' user: 'kamailio' host: 'localhost' (Got an error reading communication packets) Apr 3 19:40:53 FE-A07-34-VM6 /usr/sbin/kamailio[13516]: ALERT: <core> [main.c:767]: handle_sigs(): child process 13542 exited by a signal 11 Apr 3 19:40:53 FE-A07-34-VM6 /usr/sbin/kamailio[13558]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 38 Apr 3 19:40:53 FE-A07-34-VM6 /usr/sbin/kamailio[13516]: ALERT: <core> [main.c:770]: handle_sigs(): core was not generated Any pointers for resolution are most welcome as we need to quickly resolve this. Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

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IPSec and USIM
by Mohammad Elahi 12 Apr '20

12 Apr '20

Hi All, When my VoLTE Mobile is registered to the Kamailio IMS that support IPSec by AKAv1 algorithm, inside the ESP packet is clear text (not encrypted). Is that because my mistake in configuration or for my USIM card? thanks in advance. Mohammad

1 1

kamailio as SBC for Teams
by Aidar Kamalov 10 Apr '20

10 Apr '20

Hello, I read https://skalatan.de/en/blog/kamailio-sbc-teams and trying to use kamailio for teams. But I faced that MS not responding ACK to my 200OK. Somethin is wrong, but I don't know what. I call from teams to kamailio. Can you help me please This is initial INVITE from Teams: 2020/04/09 22:55:17.346092 52.114.7.24:1024 -> KAMAILIO_IP:5061 INVITE sip:+xxxxxxxxx@MY_KAMAILIO_FQDN:5061;user=phone;transport=tls SIP/2.0 FROM: Aidar Kamalov<sip:+xxxxxxxxx@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=de28a5b5333b451a97312097f91f5564 TO: <sip:xxxxxxxx@MY_KAMAILIO_FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: 3d5ad3cdd01457be94d127fa20478cf1 MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.7.24:5061;branch=z9hG4bKdb903e2c RECORD-ROUTE: <sip:sip-du-a-as.pstnhub.microsoft.com:5061;transport=tls;lr> CONTACT: <sip:api-du-a-euno.pstnhub.microsoft.com:443 ;x-i=903eeaac-b19e-43a0-8e75-bb222049922a;x-c=3d5ad3cdd01457be94d127fa20478cf1/d/8/d3d2c2e3f3c14f88bb2fe843b4bc70e4> CONTENT-LENGTH: 2426 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.4.3 i.ASEA.5 CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY v=0 o=- 0 0 IN IP4 52.114.252.146 s=session c=IN IP4 52.114.252.146 b=CT:50000 t=0 0 m=audio 52939 RTP/SAVP 104 9 111 18 0 8 103 97 13 118 119 101 a=rtcp:52925 a=ice-ufrag:rKBm a=ice-pwd:BkjUnkfIvYV5pjteSteevL1O a=rtcp-mux a=candidate:4 1 UDP 184547839 52.114.252.146 52939 typ relay raddr 31.13.144.55 rport 57780 a=candidate:4 2 UDP 184547326 52.114.252.146 52925 typ relay raddr 31.13.144.55 rport 57781 a=candidate:1 1 tcp-act 2121006590 100.78.247.179 1024 typ host a=candidate:1 2 tcp-act 2121006590 100.78.247.179 1024 typ host a=candidate:2 1 UDP 2130705919 100.78.247.179 39182 typ host a=candidate:2 2 UDP 2130705406 100.78.247.179 39183 typ host a=candidate:3 1 UDP 1694497791 31.13.144.55 57780 typ srflx raddr 100.78.247.179 rport 39182 a=candidate:3 2 UDP 1694497278 31.13.144.55 57781 typ srflx raddr 100.78.247.179 rport 39183 a=candidate:5 1 tcp-act 1684796926 31.13.144.55 9761 typ srflx raddr 100.78.247.179 rport 41775 a=candidate:5 2 tcp-act 1684796926 31.13.144.55 9761 typ srflx raddr 100.78.247.179 rport 41775 a=candidate:6 1 tcp-pass 174453758 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=candidate:6 2 tcp-pass 174453758 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=candidate:7 1 tcp-act 174846462 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=candidate:7 2 tcp-act 174846462 52.114.252.130 59925 typ relay raddr 31.13.144.55 rport 9761 a=x-candidate-info:4 network-type=WWAN a=x-candidate-info:1 network-type=WWAN a=x-candidate-info:2 network-type=WWAN a=x-candidate-info:3 network-type=WWAN a=x-candidate-info:5 network-type=WWAN a=x-candidate-info:6 network-type=WWAN a=x-candidate-info:7 network-type=WWAN a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:DY3r3DWRojJm94CJQ83OAavmlRCwPf5ciRxHCT40|2^31 a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:DY3r3DWRojJm94CJQ83OAavmlRCwPf5ciRxHCT40|2^31 a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:DY3r3DWRojJm94CJQ83OAavmlRCwPf5ciRxHCT40|2^31|1:1 a=rtpmap:104 SILK/16000 a=fmtp:104 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 SILK/8000 a=fmtp:103 useinbandfec=1; usedtx=0 a=rtpmap:97 RED/8000 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=rtpmap:119 CN/24000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:200 and my 200OK to MS is: 2020/04/09 22:55:19.976427 KAMAILIO_IP:5061 -> 52.114.7.24:1024 SIP/2.0 200 OK Record-Route: <sip:KAMAILIO_FQDN_DOMAIN:5061;transport=tls;ftag=de28a5b5333b451a97312097f91f5564;lr=on> Record-Route: <sip:KAMAILIO_IP:5060;ftag=de28a5b5333b451a97312097f91f5564;lr=on> Via: SIP/2.0/TLS 52.114.7.24:5061;rport=1024;branch=z9hG4bKdb903e2c Record-Route: <sip:ENDPOINT_BEHIND_KAMAILIO_IP;line=sr-QyfKA15Kw1rKw1hKEoPljM0oE1VLX10RQ1VLX10gkY37WJufE1BBjJhVE6E6r14VEJ5Nj6ElE1UNjyrNEJrVjnr.AyudmMV1rg52rJrlAyNBkMVNmSE*> Record-Route: <sip:ENDPOINT_BEHIND_KAMAILIO_IP;r2=on;lr=on;ftag=de28a5b5333b451a97312097f91f5564;did=cba.af1;nat=yes> Record-Route: <sip:KAMAILIO_IP;r2=on;lr=on;ftag=de28a5b5333b451a97312097f91f5564;did=cba.42a1;nat=yes> Record-Route: <sip:KAMAILIO_IP:5061;transport=tls;r2=on;lr=on;ftag=de28a5b5333b451a97312097f91f5564;did=cba.42a1;nat=yes> Record-Route: <sip:sip-du-a-as.pstnhub.microsoft.com:5061;transport=tls;lr> To: <sip:+xxxxxxxxxx@KAMAILIO_FQDN_DOMAIN :5061;user=phone>;tag=3795432917-393759697 From: <sip:+xxxxxxxxxx@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=de28a5b5333b451a97312097f91f5564 Call-ID: 3d5ad3cdd01457be94d127fa20478cf1 CSeq: 1 INVITE Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL Content-Type: application/sdp Accept: application/sdp Content-Length: 282 User-Agent: MY_DOMAIN Contact: <sip:KAMAILIO_IP;line=sr-DHZToOtmMdn7MODy9AnmD10TGLpUrL4noJCE4A03DdQE9LUaMXr*> v=0 o=ETPI-MSX1 0 0 IN IP4 KAMAILIO_IP s=sip call c=IN IP4 KAMAILIO_IP t=0 0 m=audio 12176 RTP/SAVP 8 b=AS:64 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtcp:12177 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:03ZmcR+CxxB7k7C8QvWBgseYq7cH7py7VxqfLM8j a=ptime:20 -- Aydar A. Kamalov

4 8

Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Luis Rojas G. 10 Apr '20

10 Apr '20

Hello, I just realized that I had the dispatcher configured using a hash of Call-ID. That means, after recvfrom there must be an extra processing finding the Call-ID header in message, to calculate a hash and then forward() message. The more the processing, the more cases when 200 could arrive before 180. I just changed it to round robin, and the amount decreased a lot, but it's still there. If I send a burst of 1000 messages, about 5 of them leave out of order every time. Best regards, Luis On 4/9/20 1:48 PM, Luis Rojas G. wrote: > Hello, > > I have a lot of experience developing mutithreaded applications, and I > don't see it so unlikely at all that a process loses cpu just after > recvfrom(). It's just as probable as to lose it just before, or when > writing on a cache or just before of after sendto(). If there are many > messages going through, some of them will fall in this scenario. if I > try sending a burst of 100 messages, I see two or three presenting the > scenario. > > Just forward() with a single process does not give the capacity. I'm > getting almost 1000caps. More than that and start getting errores, > retransmissions, etc. And this is just one way. I need to receive the > call to go back to the network (our application is a B2BUA), so I will > be down to 500caps, with a simple scenario, with no reliable > responses, reinvites, updates, etc. I will end up having as many > standalone kamailio processes as the current servers I do have now. > > I really think the simplest way would be to add a small delay to 200 > OK. Very small, like 10ms, should be enough. Simple and it should > work. As Alex Balashov commented he did for the case with ACK-Re-Invite. > > I have to figure out how to make async_ms_sleep() work in reply_route(). > > Thanks for all the comments and ideas > > Best regards, > > Luis > > > > . On 4/9/20 12:17 PM, Daniel-Constantin Mierla wrote: >> >> >> MICONDA(a)GMAIL.COM appears similar to someone who previously sent you >> email, but may not be that person. Learn why this could be a risk >> <http://aka.ms/LearnAboutSenderIdentification> >> Feedback <http://aka.ms/SafetyTipsFeedback> >> >> Hello, >> >> then the overtaking is in between reading from the socket and getting >> to parsing the call-id value -- the cpu is lost by first reader after >> recvfrom() and the second process get enough cpu time to go ahead >> further. I haven't encountered this case, but as I said previously, >> it is very unlikely, but still possible. I added the route_locks_size >> because in the past I had cases when processing of some messages took >> longer executing config (e.g., due to authentication, accounting, ..) >> and I needed to be sure they are processed in the order they enter >> config execution. >> >> Then the option is to see if a single process with stateless sending >> out (using forward()) gives the capacity, if you don't do any other >> complex processing. Or if you do more complex processing, use a >> dispatcher process with forwarding to local host or in a similar >> manner try to use mqueue+rtimer for dispatching using shared memory >> queues. >> >> Of course, it is open source and there is also the C coding way, to >> add a synchronizing mechanism to protect against parallel execution >> of the code from recvfrom() till call-id lock is acquired. >> >> Cheers, >> Daniel >> -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

3 4

t_relay failure when dropping all branches in same destination set
by Sergiu Pojoga 09 Apr '20

09 Apr '20

Hi there, Can someone help figure out why there's this t_relay error, if I decide to drop the single master branch either all branches belonging to the same destination set, even though there are more serial sets to try later? *Error:* *tm [t_funcs.c:337]: t_relay_to(): t_forward_nonack returned error -6 (-6)* *tm [t_funcs.c:355]: t_relay_to(): -6 error reply generation delayed * *Example:* request_route { seturi("sip:a@mydomain.net"); append_branch("sip:b@mydomain.net", "0.5"); append_branch("sip:c@mydomain.net", "0.5"); append_branch("sip:d@10.22.0.30", "1.0"); # append_branch("sip:e@mydomain.net", "1.0"); # no error if I add this branch with same Q=1.0 t_load_contacts(); t_next_contacts(); t_on_branch("check_branch"); t_on_failure("serial"); t_relay(); break; } branch_route[check_branch] { if($rd=="10.22.0.30") drop; } failure_route["serial"] { if (!t_next_contacts()) exit; } t_on_branch("check_branch"); t_on_failure("serial"); t_relay(); } Thanks in advance. --Sergiu

1 1

Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Alex Balashov 09 Apr '20

09 Apr '20

On Thu, Apr 09, 2020 at 02:26:45PM +0100, David Villasmil wrote: > How about just not forwarding the 180 if it’s coming right after the > 200ok? I know it’s a hack, but a 180 is a provisional response > indicating the ring status, not the actual audio for the ring... so if > a 200 is received, the call is already established, no need to forward > the 180. There is a general principle, particularly in PSTN interworking, that signalling should be reliably conserved end-to-end. I think dropping messages is the most cavalier option available, perhaps worth considering only if every other logical possibility has been exhausted first. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Daniel-Constantin Mierla 09 Apr '20

09 Apr '20

On 09.04.20 09:47, Daniel-Constantin Mierla wrote: > [...] >> >> Any idea why Async is not allowed in reply_route()? >> Didn't notice this question in the first place to respond in my previous email. Probably the developer that did it needed it for SIP requests. However, you can make a patch (pull request) to make it work for other cases. The developers will review the changes and if they don't break anything, it will be merged. Cheers, Daniel

2 1

kamailio - IPSEC
by Pavithra M 09 Apr '20

09 Apr '20

Hi, Is there any possibility to disable ipsec_pcscf in kamailio 5.3.3.

2 2

Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER
by Daniel-Constantin Mierla 09 Apr '20

09 Apr '20

Hello, it was a reply to my email where I mentioned the route_locks_size parameter. As he said he looked at that parameter, I assumed it was about the route_locks_size, because there was not other parameter listed in the emails. So using the route_locks_size parameter doesn't require to use dialog module. Cheers, Daniel On 09.04.20 10:29, Henning Westerholt wrote: > > Hello, > > > > I mentioned in some of earlier e-mails as one possible option to track > the state of a dialog and to act depending on it. > > > > Cheers, > > > > Henning > > > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com <https://gilawa.com/> > > > > *From:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *Sent:* Thursday, April 9, 2020 9:48 AM > *To:* luis.rojas(a)sixbell.com; Kamailio (SER) - Users Mailing List > <sr-users(a)lists.kamailio.org>; Henning Westerholt <hw(a)skalatan.de> > *Subject:* Re: [SR-Users] Kamailio propagates 180 and 200 OK OUT OF ORDER > > > > Hello, > > On 08.04.20 23:03, Luis Rojas G. wrote: > > Hello, Daniel, > > > > I looked into that parameter, but I need to use with the dialog > module, and I'm pretty afraid to use that. > > who said or where is written than you need to load the dialog module? > You definitely don't. > > Cheers, > Daniel > > > > I was looking more into the stateless proxy, because I need to > process a lot of traffic. > > > > My target is 4200CAPS. with duration between 90s and 210. Let's > say, 150 seconds. That would mean 630.000 simultaneous dialogs. I > don't think the solution can go that way. > > it would really help me to be able to use completely stateless > proxy plus Async in reply_route(), to introduce an artificial > delay before forwarding 200 OK to Invite.. As someone mentioned, > it would help me on request_route(), for race conditions between > ACK and Re-Invite. > > Any idea why Async is not allowed in reply_route()? > > Best regards, > > > > Luis > > > > On 4/8/20 1:07 PM, Daniel-Constantin Mierla wrote: > > Hello, > > you have to keep in mind that Kamailio is a SIP packet router, > not a telephony engine. If 180 and 200 replies are part of a > call is not something that Kamailio recognize at its core. Its > main goal is to route out as fast as possible what is > received, by executing the configuration file script. Now, a > matter of your configuration file, processing of some SIP > messages can take longer than processing other. And the > processing is done in parallel, a matter of children parameter > (and tcp_children, sctp_children). > > With that in mind, a way to try to cope better with the issue > you face is to set route_locks_size parameter, see: > > * > https://www.kamailio.org/wiki/cookbooks/devel/core#route_locks_size > <https://nam02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.kamai…> > > Probably is what you look for. > > But if you want more tight constraints, like when receiving a > 180 after a 200ok and not route it out, you have to make the > logic in configuration file by combining modules such as > dialog or htable (as already suggested). > > Cheers, > Daniel > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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