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37394 discussions

xlog print sdp parameters
by Angel Elena 11 Sep '20

11 Sep '20

Hi, good moorning. I have a Kamailio 4.4.7 in a debian box and I need print with xlog (if is possible) the o,c and m values from SDP headers. Is possible make a xlog() print this parameters with sdpops module or another ? Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://diablo.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Kopano Webapp.

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Using Kamalio as a proxy for internal servers
by Moshe Katz 11 Sep '20

11 Sep '20

Hello all, (Note: I previously posted a more detailed version of this question on StackOverflow at https://stackoverflow.com/q/63760506/829970 . This version is simplified to fit better in an email.) I have Kamailio 5.4.1 (and RTPEngine) running on an internal server with a private IP address 172.31.7.96 and One-to-one NAT to an external IP address. The external IP is 192.0.2.100. (Note: The internal IP addresses are all unedited, but the public IPs have been replaced with TEST-NET-1 and TEST-NET-2 example addresses.) I will eventually be doing transcoding with RTPEngine, but for now this is a simple SIP Proxy. Kamailio is installed on Ubuntu 18.04 using the DEB packages from dev.kamailio.org/kamailio54 and is using the stock configuration that comes with those packages, except for the following changes: #!define WITH_NAT #!define WITH_RTPENGINE #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_IPAUTH listen=udp:0.0.0.0:5060 advertise 192.0.2.100:5060 #!define DBURL "mysql://kamailio:REAL_PASSWORD_HERE@127.0.0.1/kamailio" I have internal SIP servers with private IP addresses in the 172.31.7.0/24 range that I want to have send all SIP traffic through the Kamailio server. The internal servers are running a Java SIP client with the `OUTBOUND_PROXY` setting set to 172.31.7.96. The problem I have is that the SIP `200 OK` message sent by Kamailio to my SIP server has its `Record-Route` header set to the public IP address `192.0.2.100` instead of the private address `172.31.7.96`. The SIP client therefore tries to send the `ACK` message back to the public address, but it has no route to the public address so the ACK never gets sent. How can I configure Kamailio to use the public IP for external traffic but the private IP for communicating with internal machines on the same subnet? I tried setting `mhomed=1`, but the machine isn't actually multi-homed so that didn't work. I thought of adding a second listen line `listen=udp:172.31.7.96:5061` and having the internal servers talk to port 5061, but that doesn't work because Kamailio uses the 5061 definition for the external side too. I see in the docs that it is possible to name the listener lines, but I don't understand how to use those names in a way that would be relevant to my issue. Thank you very much for your help, Moshe

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How to handle BYE in case of serial fork
by Björn Klasen 11 Sep '20

11 Sep '20

Hi we are using Kamailio 4.3.7 and I have a question concerning BYE handling. We have the following situation: A carrier a call to us. We are forwarding the call to the destination, but it does not pick up the call. We have call-forward active with use of tm module, so we forward the call to the new destination. The first call generates a to-tag in 180, let's call it TT1. The second call, so the c-destination also answer with 180 but with another to-tag (TT2). Both ringing are passed to the originating carrier. When the timer for the first call is timed out, Kamailio cancels the branch with a CANCEL so it does not exist any more Now the c-destination picks up the phone, and reply with 200 OK with TT2 The carrier then sends a BYE with TT1 to us, but Kamailio doesn't know anything about it, so it does not reply and the carrier sends a timeout and terminates the complete call. So the question is, how handle such a situation. Is it possible to store the first branch, or fake a OK reply to the BYE request of the carrier as for my understanding this behaviour is not against RFC. To clarify the situation here comes a call-flow diagram A Kamailio B C |---INV Bob@P1->| | | | |--INV Bob@B--->| | | |<-100----------| | |<-100----------| | | | |<-180 TT1------| | |<-180 TT1------| | | Forward after 10 seconds | |--CANCEL------>| | | |<-487----------| | | |<-200 OK-------| | | |--ACK--------->| | |<-181 TT1------| | | | |--INV Carol@C----------------->| | |<-100--------------------------| | |<-180 TT2----------------------| |<-180 TT2------| | | | |<-200 OK TT2-------------------| |<-200 OK TT2---| | | |--BYE TT1----->| | | |--BYE TT1----->| | | |--BYE TT1----->| | | |--408--------->| | | |--ACK--------->| | | | |--ACK------------------------->| | |<-BYE TT2----------------------| |<-BYE TT2------| | |--481--------->| | | |--481------------------------->| To get around the the situation with the second to-tag, we passing every call through SEMS now, because its B2B function rewrites the to-tag, so there is always only one to-tag towards the carrier A. But it's not the gold solution. I hope somebody can help me. BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

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Kamailio v5.3.6 Released
by Daniel-Constantin Mierla 10 Sep '20

10 Sep '20

Hello, Kamailio SIP Server v5.3.6 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.5. 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 v5.3.6. Note that 5.3 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.4, with v5.4.1 being release out of it. For more details about version 5.3.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/09/kamailio-v5-3-6-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

Releasing v5.3.6
by Daniel-Constantin Mierla 10 Sep '20

10 Sep '20

Hello, I am considering to release v5.3.6 from the branch 5.3 in the next days, aiming from Thursday (Sep 10, 2020) if everything goes as planned at this moment, or on Monday (Sep 14, 2020) if a little bit of time is needed. Should you be aware of any issue that needs to be addressed in the branch 5.3 (e.g., a bug, or a missing backport), inform us as soon as possible to try to sort it out in time. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

Dispatcher / ds_select_routes
by Peter Gradwell 09 Sep '20

09 Sep '20

Hello I'm doing a project where I need to route using dispatcher across multiple groups. i.e. when the customer makes a call, the "core" server makes a request to cgrates to get back a list of carriers that support that, and then want to work through them in order of preference. e.g. the string we get back from cgrates via evapi might be "2,1,3". i.e. try all Group 2 destinations, then group 1, then group 3. Each of these is then a dispatcher group (with multiple dispatcher routes in each group). When the call then gets to the "Group 2" server that in turn is also using dispatcher to handle multiple destinations, and if all of those destinations fail, it returns a 503 to the "core", which is then meant to try "Group 1", and so on. The documentation on this is a bit sparse, but what I think I am meant to do (at the core level) is use ds_select_routes to load up the list of groups (https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…) if(ds_select_routes("2=4;1=4;3=4", "d", "0")) { t_on_failure("REROUTE"); t_relay(); exit; } and then in my failure_route, if the error code is 503 I call ds_next_domain, e.g. failure_route[REROUTE] { if(t_check_status("503")) { if(ds_next_domain()) { t_on_failure("REROUTE"); t_relay(); exit; } } } My problem is, that in applying theory to practice, I have ended up with a bunch of looping and it's not working right! Does anyone have an example code for such a scenario? many thanks peter

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testing SIP address
by Nicholas Saunders 09 Sep '20

09 Sep '20

How do I test connectivity of a SIP address? Is there a "time" or "echo" type SIP address to call or message? I'm just using a free address from onsip.com thanks, Nick

4 3

how to reload kemi scripts? kamailio 5.3.x or 5.4.x
by Karsten Horsmann 09 Sep '20

09 Sep '20

Hello List, i read Daniels Presentation https://www.kamailio.org/events/2018-KamailioWorld/Day0/W07-Daniel-Constant… and there is an hint about python(2/3) - that kemi script reloading is not supported with python kemi scripts. Is that correct for the actual versions (5.3.x and 5.4.x)? And if i use for example lua as kemi script language, how i restart that script only? Drop in a new file and thats it (like FreeSWITCH did that)? Thanks in advance Cheers Karsten -- *Karsten Horsmann*

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$HN(key) for IPv6
by Sergey Safarov 05 Sep '20

05 Sep '20

I want to deploy dual-stack kamailio and config file in need use in one place IPv4 kamailio IP address and in another place I want to use IPv6 address. Could I get something like $HN(i4) and $HN(i6)?

2 1

Change FROM to hide topology
by Andre Pires dos Santos 05 Sep '20

05 Sep '20

Hi guys, We have SIP Server > Kamailio > webrtc client. The only call flow we have is SIP server calling kamailio tha sends the call to webrtc client. We set up TOPOH to hide the sip server info but once the sip server initiates the call we still have the FROM details identifying SIP Server IP address. We couldn't find a way to mask this ip using TOPOH. What would be the best way to hide the sip server IP address in this case? Thanks a lot! Andre

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