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42 participants
37517 discussions

Kamailio LCR rule matching
by Ali Taher 02 Jun '15

02 Jun '15

Hi, I'm using LCR module in Kamailio , where matching of the rule (From-URI in LCR-rule table) is currently based on user part of sip address in From field (highlighted below in blue). While we need to change the identification to be on the otg value sent in the sip address in From field (highlighted below in yellow). INVITE sip:86532917453592@172.16.5.108:5060 <http://sip:86532917453592@172.16.5.108:5060> SIP/2.0 Via: SIP/2.0/UDP 172.16.5.112:5060;branch=z9hG4bK1sansay9906915rdb2243 Via: SIP/2.0/UDP 216.53.4.5:5060;branch=z9hG4bK_1852370774_4741_1 Record-Route: <sip:sansay9906915rdb2243@172.16.5.112:5060;lr;transport=udp> To: <sip:86532917453592@69.41.186.186:5060 <http://sip:86532917453592@69.41.186.186:5060> > From: <sip: <http://sip:+249964642906@216.53.4.5:5060> +249964642906@216.53.4.5:5060;otg=024050>;tag=1852370774_C Call-ID: sbcsipuac.2_169.132.137.51_b55sb12_1_1_2015052909455960_1852370774_487698 CSeq: 1 INVITE Contact: <sip:+249964642906@216.53.4.5:5060 <http://sip:+249964642906@216.53.4.5:5060> > P-Asserted-Identity: <sip:+249964642906@216.53.4.5:5060 <http://sip:+249964642906@216.53.4.5:5060> > Privacy: none User-Agent: VCS 5.9.2.42-01 Allow: INVITE, ACK, CANCEL, BYE Max-Forwards: 69 Content-Length: 0 How can we achieve this ? Any help will be appreciated. Regards, Ali

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RTPengine+Kamailio, 200 OK without DTLS fingerprint
by Vasiliy Ganchev 02 Jun '15

02 Jun '15

Hello list! Have some difficulties with RTPengine: My test scheme is: UA(WS) -> Kamailio (WS proxy) -> Kamailio (SIP REGISTRAR, proxy) -> ASTERISK (media server) -> UA (SIP) Kamailio (WS) use RTPengine. Kamailio (WS proxy) - 4.2.3 Kamailio (SIP REGISTRAR, proxy) - 4.1.4 RTPengine - 3.3.0.0+0~mr3.8.0.0 I make a test call and have troubles in negotiation between UA(WS) -> Kamailio (WS proxy): UA(WS) -> INVITE -> Kamailio WS ...... rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force"); ...... this INVITE consists in SDP section a=fingerprint:sha-256...... When Kamailio (WS) receives 200 OK, it is also handled by RTPengine ....... rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force"); ....... but UA (WS) receives 200OK without fingerprint, and log an error: Failed to set remote answer sdp: Called with SDP without DTLS fingerprint. Currently spend a lot of time reading Kamailio/RTPengine documentation/mail-list - but without success If someone have some hints, I would appreciate any help. -- View this message in context: http://sip-router.1086192.n5.nabble.com/RTPengine-Kamailio-200-OK-without-D… Sent from the Users mailing list archive at Nabble.com.

3 8

RFC 6140
by Al S 02 Jun '15

02 Jun '15

Hi, I was wondering if Kamailio support RFC 6140 to register Addresses of Record (AORs) in bulk map to a unique set of contacts. REGISTER sip:kamailio.example.com SIP/2.0 Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7 Max-Forwards: 70 To: <sip:pbx@kamailio.example.com> From: <sip:pbx@kamailio.example.com>;tag=a23589 Call-ID: 843817637684230@998sdasdh09 CSeq: 1826 REGISTER Proxy-Require: gin Require: gin Supported: path Contact: <sip:198.51.100.3:5060;bnc> Expires: 7200 Content-Length: 0 Thanks,Al

2 3

next via branch value
by Juha Heinanen 01 Jun '15

01 Jun '15

when kamailio forwards invite request, it adds to it a new via header. for example, incoming invite: SIP/2.0/TCP 192.168.43.192:48089;branch=z9hG4bK4e1ab219cbde6190;rport outgoing invite: SIP/2.0/TCP 192.98.102.10;branch=z9hG4bK27b9.597e97c43da7ddd6be8520b662d616b9.0;i=5 SIP/2.0/TCP 192.168.43.192:48089;received=192.98.102.10;branch=z9hG4bK4e1ab219cbde6190;rport=49582 is it somehow possible to find out in the config script, what will the branch value of the new via header be? if that value would be known, it could be assigned to rtpengine extra_id_pv pseudo variable before making the offer and could then be used in rtpengine_delete to delete the right branch. if the new via branch value cannot be known, any suggestions on how to uniquely identify the outgoing branched of the invite? -- juha

4 14

Request of advice about TAS for IMS
by dfretes＠ing.una.py 01 Jun '15

01 Jun '15

I am trying to setup a really simple (I hope so) IMS platform, with some basic services (like voicemail, IVR, videoconference), but I'm kind of stuck. I have installed Kamailio IMS on debian and a OpenIMS VM also. But my little problem begins when I try to configure a application server (in this case, a TAS). I have tried with Elastix 3.0 (kamailio+asterisk), asterisk alone, and read some about Restcomm. For Elastix and asterisk, I get stuck trying to setup the trigger point. For Restcomm, I only find documentation for integration with Clearwater IMS. So, I need your help. Somebody has something like that working? What TAS (or components for a TAS) do you recommend? Tutorials, examples, manuals that you know can help me? Thanks for reading, and many many thanks for any advice. David

2 1

Select a network
by Nelson Migliaro 01 Jun '15

01 Jun '15

Hello everybody, I need to apply a specific configuration to softphones registering from a specific network range. Is there something like:

2 2

Invalid SDP line (no '=' delimiter) error !
by Badri Ranganathan 01 Jun '15

01 Jun '15

Hi all, I am answering a SIP call made to my softphone. The softphone rings and also sends a 200 OK correctly. But the other end that made the call is seeing a failure in the SDP of the 200 OK packet. This error is seen in SDP header => Invalid SDP line (no '=' delimiter) error w.r.t w_frame4_public_ip.pcap It looks like the failure as seen in wireshark is because line "audio 5000 RTP/AVP 8 101" should really be "m= audio 5000 RTP/AVP 8 101" so it doesn't see the '=' delimiter. But if you look at w_frame52_local_ip.pcap, the packets sent from the local IP has the contents correct. It is only after passing the public IP and at the network premises, that this failure is reported by the Session Border controller. I am not able to understand why a correctly formed packet sent out is seen "corrupted" at the network end. Any ideas really appreciated. Thanks, Badri.

2 1

Re: [SR-Users] Regarding presence conference
by Daniel-Constantin Mierla 01 Jun '15

01 Jun '15

Hello, the conference mixer is supposed to send PUBLISH requests with information about who is in the conference room, who is speaking at that moment and so. That is not the job of the presence_conference module. Eventually, that could be a pua_conference module that has to be implemented. The presence_conference module was implemented as part of GSoC and the PUA was supposed to be inside SEMS, but the other project failed. If you want to get it all working, you have to implement the PUA either in your version of rtpproxy or part of kamailio as well. When all in kamailio, you can see the pairs of modules such as presence_dialoginfo and pua_dialoginfo. You already have presence_confgerece. Cheers, Daniel On 28/05/15 14:28, Surendra Pullaiah wrote: > > Hello Daniel, > > > > I am implementing conference bridge using kamalio and > RTP porxy without media mixing. Since for now we are ok with this. > My question is as per the flow of RFC 4579, UA may send SUBSCRIBE to > the conference URI with event as conference. In kamalio modules > presence_conference where they mentioned such way that it will support > the same what I am looking for. But I simulated using sipP by sending > subscriber with conference event. I am not seeing any notify XML which > will have the conference users list. One more doubt where it will pick > the conference users, since I am not seeing any specific data base > table for the same. > > > > Can you please clarify what exactly for > presence_conference module. > > > > Note: I have loaded three modules presence, presence_xml, > presence_conference, in route config I am calling handle_subscribe > when SUBCRIBE method receives. > > > > Regards > > Surendra > > India: +91-8124625001 > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

Forward register to multiple Asterisk
by Nelson Migliaro 01 Jun '15

01 Jun '15

Hello, I am trying to forward the register to multiple Asterisk using Kamailio. The basic Idea is: Softphone A---->Kamailio-----> Asterisk 1 ------> Asterisk 2. I follow Asipto howto http://lylix.net/kamailio, but the problem I see in my case, is that registration IP in Asterisk is Kamailio´s IP. This causes calls going from kamailio to asterisk and then back to kamailio. In my scenario, I need that softphone A IP appeard in Asterisk realtime contact IP. The call path would be: Softphone B -> kamailio -> Asterisk 1 or Asterisk 2 (depending on dispatcher) -> Softphone A Following Asipto how to the call goes Softphone A -> kamailio -> asterisk -> kamailio Softphone B Hope I explain myself clearly. Thank you in advance.

2 1

Kamailio , RTPProxy and NAT
by Mir Jee 01 Jun '15

01 Jun '15

HI! I need help to configure RTPProxy. I have working Kamailio on Centos 6.5, On LAN everything is working fine. Now I want my other users to connect using 3G or any other network. Please help me with that. Statis Public IP (DSL Modem) ----- Proxy Box with 2 Lan cards --- 192.168.2.0 and 192.168.0.0 (Lan Side)

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