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20 participants
37491 discussions

Kamailio + RTPEngine + RTPEngine Recording daemon forward-to option
by Balázs Radák 08 May '18

08 May '18

Dear all, I would need some help setting up Kamailio with RTP engine. I'm trying to make use of RTPEngine recording daemon's forward-to option, to have the calls streams sent to 3rd party application. Have someone managed to have this forward-to option work? If I understood well, when RTPEngine's forward-to option is turned on, the call metainfo + streams can be sent to a socket server? Thanks in advance! Regards, Balazs

2 1

Copy $xavp
by Duarte Rocha 08 May '18

08 May '18

Greetings, I'm getting a query result in a $xavp. I'm trying to copy the entire structure to a new $xavp like this : $xavp(Valid_Acd_Results) = $xavp(ACD_Query_Result[$var(iterator)]); However, i can't retrieve the value with : xnotice(" Mode = $xavp(Valid_Acd_Results=>TBK_AcdMode)"); since it returns NULL. Am i doing something wrong? Thanks

1 0

Record-route and proxy_challenge
by Евгений Голей 08 May '18

08 May '18

Hi! Tell me why when using the save ('') function in the response generated by the server, there are the Record-route headers that were in the request. And when using the following functions: - proxy_challenge (), - www_challenge (), - send_reply (), - sl_send_reply there is no Record-route header. How can you add them? -- Evgeniy

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Garbage in to field
by Kjeld Flarup 08 May '18

08 May '18

I'm having problems with my To field. I'm doing an async_route, and afterwards sends the call with auth to a SIP server. My problem is that in the first invite I see the expected To field sip:004520202020@isp.com But when the Auto is send the To field is changed to the, To field for the PBX sip:1234@pbx.local I tried to fix this with uac_replace_to("$ru"), but that ends up in this To field: sip:004520202020@isp.comsip:004520202020@isp.com It seems like uac_replace_to sometimes appends rather that replaces. If I run it twice, then it for sure appends. Is this expected behaviour. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

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How to change Calling Line ID in cr_route
by KamDev Essa 08 May '18

08 May '18

I have cr_route working well. Only problem is that the username is sent off to the carrier and the 3 or 4 digit username appears as the callerid. Is there a module or scripts that can replace the username with a db table defined callerid. Say 202 maps to a 10 digit TN. KD

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STUN
by Wilkins, Steve 08 May '18

08 May '18

Hello all, Does anyone know the current status of how to enable STUN. I am using Kamailio Ver 5.x. Thank you, -Steve

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Kamailio v5.1.3 Released
by Daniel-Constantin Mierla 07 May '18

07 May '18

Hello, Kamailio SIP Server v5.1.3 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.3. For more details about version 5.1.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/04/kamailio-v5-1-3-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.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

3 3

Determining next hop for any SIP message
by George Diamantopoulos 07 May '18

07 May '18

Hello all, I've been trying to figure out a cleaner way to determine the next hop for a SIP message, mainly for use within the NATMANAGE route for multi-homed kamailio instances (with three or more network interfaces on the kamailio host). So far I have achieved this with a series of nested ifs, depending on whether the message is a request or a response and by calculating the next hop based on the various headers (R-URI, Route) and variables ($T_req($Ri), $dd, ) involved in SIP routing. A simpler way to do it, of course, would be to use the onsend_route, but that would most likely introduce an unnecessary overhead for all routed messages. I recently noticed there a pseudovariable called $nh(key), and I believe I can use $nh(d) to the same effect. I understand however, that this works for requests only. Also, the description of this PV in the documentation reads as follows: Return attributes of next hop for the SIP request. Address is taken from > dst_uri, if set, if not from new r-uri or original r-uri. > What is not clear to me is if this covers in-dialog requests with the Route header set as well. Does the inclusion of a route header set the dst_uri PV? And if yes, is it safe to rely on dst_uri during request processing or is it set only after completion of script processing? Lastly, is there an analogue for SIP responses? If not, is it safe to rely on first Via header to determine next hop for responses, or are there any other corner cases I need to heed? Thanks in advance, BR, George

3 6

Server-side NAT traversal with Kamailio
by Alex Balashov 07 May '18

07 May '18

Recent repeated questions on this topic have inspired a blog post: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defi… This is a rough cut, and may be edited or expanded later. -- 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/

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Route header in local reply
by Евгений Голей 07 May '18

07 May '18

Hi. Colleagues write how to configure the local responses to requests so that the response added the header of the Route in accordance with the record-route headers that were in the request ? Example SBC --> Kamailio REGISTER sip:domain.com SIP/2.0 Record-Route: <sip:AAA.AAA.AAA.AAA:6060;lr> Via: SIP/2.0/UDP CCC.CCC.CCC.CCC;branch=z9hG4bKae7f.474bd538e793b8bc912ce7d0c8e0bae8.0 Via: SIP/2.0/UDP S.s.s.s;branch=z9hG4bKae7f.d81efe54000000000000000000000000.0 To: <sip:000143949@domain.com> From: <sip:000143949@domain.com>;tag=d55866b28e60059575f1fd1facc53fdd-516b CSeq: 10 REGISTER Call-ID: f74322050fd19c19f5d0ce763cc7dbdb Max-Forwards: 70 Content-Length: 0 User-Agent: SBC Expires: 0 Kamailio --> SBC SIP/2.0 200 OK Record-Route: <sip:AAA.AAA.AAA.AAA:6060;lr> Via: SIP/2.0/UDP CCC.CCC.CCC.CCC;branch=z9hG4bKae7f.474bd538e793b8bc912ce7d0c8e0bae8.0 Via: SIP/2.0/UDP S.s.s.s;branch=z9hG4bKae7f.d81efe54000000000000000000000000.0 To: <sip:000143949@domain.com>;tag=550bc1454fc5070bdadb084c686d99a2.ac18 From: <sip:000143949@domain.com>;tag=d55866b28e60059575f1fd1facc53fdd-516b CSeq: 10 REGISTER Call-ID: f74322050fd19c19f5d0ce763cc7dbdb Server: kamailio (5.1.0 (x86_64/linux)) Content-Length: 0 -- Evgeniy

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