sr-users May 2018

sr-users@lists.kamailio.org

92 participants
143 discussions

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

5 6

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

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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

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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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How long time does a push take?
by Kjeld Flarup 07 May '18

07 May '18

Hello I have a question to those of You who uses push via Google or Apple to initiate calls. How long time does it usually take, before the phone gets the push and wakes up? Our experience is, that it can take from 2 to 10 seconds, and that not impressing when we are talking about telephony. -- -------------------- 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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Pleasing all kinds of UACs for RTP/SRTP with rtpengine
by Sergiu Pojoga 07 May '18

07 May '18

In continuation of previous <https://marc.info/?l=sr-users&m=150730146302404&w=2>thread about none/optional/compulsory SRTP handling... I used the suggested approach of *t_reuse_branch()* in a *tm:branch-failure* detected by t_check_status("(415)|(488)") replies, with the only difference of me doing it vice-versa from the suggested mantra of offer SRTP first then fallback to RTP. It works... well, for some UACs, not so with others. The 'good' UACs happily accept an updated branch, like for e.g. Yealink, will reply with "*488 Not Acceptable Here*" at first followed by a *200 OK*. The 'bad' UACs, like for e.g. Bria || Zoiper, will issue a "*415 Unsupported Media Type*" at first, and then a "*482 Merged Request*" after receiving and updated branch with a different RTP profile. Rightfully so, they decided that the updated branch is a retransmission, since both branches have the same CSeq, Call-ID and From tag. The branch ID however differs, so does the RTP profile obviously. *Any suggestion on how to convince all UACs that the 2nd attempt is different than the 1st one?* According to RFC 3261, a request is a merged request if ... "8.2.2.2 Merged Requests If the request has no tag in the To header field, the UAS core MUST check the request against ongoing transactions. If the From tag, Call-ID, and CSeq exactly match those associated with an ongoing transaction, but the request does not match that transaction (based on the matching rules in Section 17.2.3), the UAS core SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction."

2 2

TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 07 May '18

07 May '18

Enabling topos breaks PRACK in my simple (near default) proxy setup (attached kamailio.conf). asterisk 13 (10.0.3.147) <-> kamailio (10.0.3.87) <-> asterisk 13 (10.0.3.154) With the event route topos:msg-outgoing dropping all. The flow of messages is as expected. INVITE -> <- Ringing PRACK -> <- 200 OK With topos enabled, the PRACK doesn't get past route[WITHINGDLG] and is answered with the sl_send_reply("404","Not here"); INVITE -> <- Ringing PRACK -> <- 404 Not here These specific tests were done on kamailio 5.1.2, but with 5.1.3 I see the exact same messages.

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Need help with NAT Traversal - Changing INVITE OK contact to IP port to Contact in Location
by KamDev Essa 06 May '18

06 May '18

The stock cfg states definng WITH_NAT and alls good. I did exactly that. However boing with stock cfg and WITH_NAT, the Registration info in location table is perfect with the UAs external IP address and port. This is not the case with the OK that in response to an INVITE. In the Contact of the OK the private IP shows up and Kamailio just sends it off to the Carrier who obviously cannot send a ack to a private IP. Is there a step by step guide for NAT Traversal? H

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sr-users May 2018