sr-users August 2021

sr-users@lists.kamailio.org

49 participants
52 discussions

by David Villasmil

Hello guys, I know SEMS can provide conference, voicemail, and other services. In theory it’s also a B2BUA. Could I use a python script to provide a simple routing service? I.e.: receive an invite and send it somewhere else based on some routing logic? Thanks all David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 years, 1 month

manually append_branch to lookup result

by David Villasmil

Hello guys, I'm trying to dynamically add a branch after doing lookup. The user is found, but in some cases I need to add a branch and do parallel forking. So i'm basically doing: route[LOCATION] { if (!lookup("location")) { .... } if (something) { route(BRANCH_TO_EXTRA); } route(RELAY); } route[BRANCH_TO_EXTRA] { $fs = MY_SOCKET; append_branch("sip:$tU@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port)); return; } For some reason only the branch appended is being used (I have append_branches=1) Ideas? David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 years, 2 months

Manage multiple RTP streams with different TO-tags (forking)

by B. Tietz

Hello, following situation. I have a Kamailio (5.4) using rtpengine to loadbalance calls. If a call from Alice comes in, Kamailio decides to send the call to Carrier B from Bob. Bobs Phone is ringing and the carrier B send a 183 Session Progress with SDP and To-tag=abcd. The SDP has G722 as codec and port 1234. A few moments later carrier B send a second 183 Session Progress with SDP and TO-tag=fghi. The SDP has G711 as codec and port 5678. This is done, to play some funky music as ringtone -.- If Bob answers the call, carrier B sends a 200 OK WITHOUT SDP and TO-tag=abcd. So this should instruct our Kamailio to switch to the first G722 and port 1234. But sadly, this is just not working as expected. We tried to set the flags media-handover and port-latching for the rtpengine options and additionally set a to-tag when using rtpenging_manage. But this doesn't solve the codec change, so we have only audio when Bob answers the call, but no ringtone-music. If we allow G711 only in the outgoing INVITE to Bob, we have also tha ringtone-muisic, because there is no codec-change. Carrier B tells us, they are using a fork-mechanism. Is there something we can do, to support the codec change in 183? Or enforce carrier B to send SDP in 200 OK? Or anything else? Carrier B can not change anything in the ringtone-music-backend. They are stuck on G711. Thanks!

3 years, 2 months

Re: [SR-Users] t_drop_replies not working with t_suspend in failure route

by Huber Andreas

Hi, We’re still using kamailio 4.4 but we’ll be migrating to 5.0 soon. Cool so it will be fixed when we migrate ! Thanks, Andreas From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Federico Cabiddu Sent: vendredi 12 mai 2017 11:56 To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] t_drop_replies not working with t_suspend in failure route Hi, which version are you using? A similar case had been reported some months ago and it should be fixed in 5.0. Regards, Federico On Fri, May 12, 2017 at 11:44 AM, Huber Andreas <andreas.huber(a)nagra.com<mailto:andreas.huber@nagra.com>> wrote: Hello, We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE. We’d like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect. The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch. But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something? Kind Regards, Andreas _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

3 years, 2 months

How to drop negative final response and send ACK

by Koray Vatansever

Hi, I have a problem with the following scenario: I have web and mobile sip clients for the same username. If both clients are idle, INVITE is forked and both clients ring. If web client is busy and mobile client is online and registered, the mobile client rings and the caller hears ringback tone. If web client is busy and mobile client is offline, before the mobile client receives push notification and registers, web client sends 486 busy response and caller hears busy tone. In this last scenario, I want to wait some time for the mobile client to register and ring. I tried to drop 486 response in onreply_route[x], but I found that final responses cannot be dropped in this route. I dropped 486 in reply_route, but I couldn't send an ACK for this transaction. Is there a way to implement this scenario? I would appreciate if you have any suggestions. Thanks, Koray

3 years, 2 months

dns_srv_lb on or yes

by Juha Heinanen

Wiki tells: dns_srv_lb = yes | no (default no) and doc/tutorials/dns.txt tells: dns_srv_lb = on | off Which one is correct or does it matter? -- Juha

3 years, 2 months

Info: Tomorrow Live - Kamailio World 2021 - Online

by Daniel-Constantin Mierla

Hello, couple of details for those that want to watch live or even participate to Kamailio World Online Conference, Sep 1-2, 2021 - the schedule is available at: * https://www.kamailioworld.com/k09-online/ There is no registration required, participation is completely free. You can watch live the presentations starting with 13:00UTC (14:00 London, 15:00 Berlin/Paris/Rome/Madrid, 09:00 New York) via KamailioWorld youtube channel: * https://www.youtube.com/c/KamailioWorld If you want to participate to the live video conferencing room, the link is: * https://meet.kamailioworld.com/live It is recommended to do it only if you want to address questions and interact with the speakers, otherwise the capacity of the video conferencing system may be filled and we will have to restrict the access in order to ensure good quality for presentations. You can also address questions via youtube chat that will be available next to the live stream or Matrix room: * #kamailio:matrix.kamailio.dev * https://riot.kamailio.dev/#/room/#kamailio:matrix.kamailio.dev To interact with other community member, join the Matrix chat room. No registration is required for Matrix to join the room, however the web client at the link above does not support the guests feature, you will have to use another client application that supports it or make an account with any federated Matrix server, like the one from element.io: * https://app.element.io/#/welcome Looking forward to chatting with many of you during the next two days! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 years, 2 months

Add PIDF-LO object to INVITE

by Adarsh Chauhan

Hi All, I am trying to add PIDF-LO XML to INVITE, however i have had little success so far. My end goal is to create an invite like mentioned in https://docs.microsoft.com/en-us/openspecs/office_protocols/ms-sipre/4010da… I have the following config: request_route{ xlog("L_DBG","REQUEST RECEIVED \n ($mb) \n"); if (is_method("INVITE")) { xinfo("$ua\n"); regex_substring("$ua", "([a-f0-9]{12})",0, 1, "$var(asd)"); if($var(asd)!="") { http_client_query("http://127.0.0.1/location/$var(asd)","$var(result)"); xdbg("$rc ,[$var(result)]\n"); if(!has_body("application/pidf+xml")) { set_body_multipart("$rb", "application/sdp", "delimiter"); msg_apply_changes(); append_body_part($(var(result){s.unquote}), "application/pidf+xml"); msg_apply_changes(); } }else { xerr("NO MAC ADDRESS FOUND\n"); } } route("forward"); } which achieves the following but doesn't have the xml in the correct format. I have explored most of the modules but couldnt find a handy function for this. INVITE sip:service@10.0.1.133:5060 SIP/2.0 Via: SIP/2.0/TCP 169.254.172.2;branch=z9hG4bK3de6.d3b4cddbcbdba59a3e190c41b06ada15.0;i=1 From: sipp <sip:sipp@1.1.1.1:45907>;tag=21113SIPpTag001 To: service <sip:service@10.0.1.133:5060> Call-ID: 1-21113(a)1.1.1.1 CSeq: 1 INVITE Contact: sip:sipp@1.1.1.1:45907 Max-Forwards: 70 Subject: Performance Test Content-Length: 937 Content-Type: multipart/mixed;boundary="delimiter" Mime-Version: 1.0 --delimiter Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP4 172.31.87.143 s=- c=IN IP4 1.1.1.1 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 --delimiter Content-Type: application/pidf+xml <?xml version=\"1.0\" encoding=\"UTF-8\"?>\n<presence xmlns:xsd=\"http:/ /www.w3.org/2001/XMLSchema\" xmlns:xsi=\"http://www.w3.org/2001/ XMLSchema-instance\" entity=\"sip:U SER9(a)SBSBVR00Test0m.ABC.com\" xmlns=\"urn:ietf:params:xml:ns:pidf\">\n<tuple id=\"tuple0\">\n<status>\n<geopriv xmlns=\"urn:ietf:params:xml:ns:pidf:geopriv10\">\n<civicAddre ss xmlns=\"urn:ietf:params:xml:ns:pidf:geopriv10:civicAddr\">\n['<STS>Ave</STS>\\n', '<A1>WA</A1>\\n', '<A3>Redmond</A3>\\n', '<RD>163rd</RD>\\n', '<PC>98052</PC>\\n', '<POD >NE</POD>\\n', '<NAM>Contoso Corporation </NAM>\\n', '<HNO>3910</HNO>\\n', '<country>US</country>\\n']\n</civicAddress>\n</geopriv>\n</status>\n</tuple>\n</presence> --delimiter-- Regards, Adarsh Chauhan

3 years, 2 months

What does "tls.reload" actually do?

by Sebastian Damm

Hi, I noticed a strange behavior on some of our proxy servers, all running Kamailio 5.3.8. After running for some time (weeks), our monitoring system sporadically starts reporting errors. The check connects via tls and registers to an Asterisk behind the proxy server. When this happens, the Kamailio log shows the following line: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1409441B:SSL routines:ssl3_read_bytes:tlsv1 alert decrypt error When restarting Kamailio, the problem goes away only to come back after some weeks uptime again. On one host, I tried to find something using kamcmd, and I don't know why but I also issued "tls.reload". And from that point, the monitoring system has not reported the system as faulty anymore. I repeated the same thing on other hosts when the problem occured there, all with the same result. "tls.reload" helps. But from the documentation, I don't know why. Does anybody have an explanation for it? Regards, Sebastian

3 years, 2 months

Slack Group

by Daniel Donoghue

Hello everyone Does anybody know if the Slack workspace at kamailio.slack.com (as mentioned on https://www.kamailio.org/w/contact-us/) is still active? If it is, who should we contact for access? With every blessing, — Daniel Donoghue

3 years, 2 months

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sr-users August 2021