sr-users September 2014

sr-users@lists.kamailio.org

96 participants
171 discussions

Re: [SR-Users] Sip Event Filter
by Raúl Alexis Betancor Santana 24 Sep '14

24 Sep '14

Hi, you are facing the problem the wrong way. You could do that easy on the UAC side. For example, with Asterisk as UAC, you only need to pass a 10s timeout as paran to the dial app. A sip-proxy its not for what you are trying to acomplish. What you are trying to do it's a UAC or B2BUA task. Best regards. El 24/09/2014 09:00, Delifisek Tux <delifisek.tux(a)gmail.com> escribió: > > Hello, > > Very newbie here. Even I may not know what I'm talking about. Please be patience, thank you :) > > I had auto dialer and I have problem with unreachable numbers. > > So I wish if I call busy, unavailable, ringing but not answering numbers, system will hangup before the voice mail active. > > After some googling I found a sip proxy may solve my problem. > > I setup kamailio and manage to work with an example edge proxy setup. > > Then what ? > > My wish is, after the number dialed, start to filter every sip message, then let say if I do not get 183 session in progress in 10 seconds, or get a 10 trying in 10 seconds hangup line. > > Which module should I look after. > > My Best Regards. > >

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Sip Event Filter
by Delifisek Tux 24 Sep '14

24 Sep '14

Hello, Very newbie here. Even I may not know what I'm talking about. Please be patience, thank you :) I had auto dialer and I have problem with unreachable numbers. So I wish if I call busy, unavailable, ringing but not answering numbers, system will hangup before the voice mail active. After some googling I found a sip proxy may solve my problem. I setup kamailio and manage to work with an example edge proxy setup. Then what ? My wish is, after the number dialed, start to filter every sip message, then let say if I do not get 183 session in progress in 10 seconds, or get a 10 trying in 10 seconds hangup line. Which module should I look after. My Best Regards.

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upgrade question to Kamailio 4, how to handle ruid in location/aliases
by Klaus Darilion 24 Sep '14

24 Sep '14

Hi! I have to upgrade a production server from 3.x to 4.x. 4.x added a unique constraint on ruid. I have plenty of entries in aliases and usrloc table. How do I update these entries to get the ruid field populated? Can I do an upgrade reusing the entries stored in the usrloc table or do I have to truncate the table and then wait until every client reREGISTERs? Thanks Klaus

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R: Re: RTPPROXY & BRANCH
by marino.mileti＠alice.it 24 Sep '14

24 Sep '14

Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive video media before they answer) So at the moment this scenario is not possible? Is there any possible turnaround? ----Messaggio originale---- Da: rfuchs(a)sipwise.com Data: 24-set-2014 1.15 A: <sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] RTPPROXY &amp; BRANCH On 23/09/14 04:35 AM, marino.mileti(a)alice.it wrote: > I've a problem with rtpproxy during a parallel ring scenario. > I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I > try to call them in parallel mode (ringall) rtpproxy module sends in to > the INVITE the same RTP ports. > > Is it possible to manage rtpproxy in order to generate a couple of new > port for each branch? Not at the moment. The logic is that one endpoint advertised by the client is mapped to one endpoint advertised by rtpengine. Why would you like a separate set of ports? cheers _______________________________________________ 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

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RTPPROXY & BRANCH
by marino.mileti＠alice.it 23 Sep '14

23 Sep '14

Hi I've a problem with rtpproxy during a parallel ring scenario. I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I try to call them in parallel mode (ringall) rtpproxy module sends in to the INVITE the same RTP ports. Is it possible to manage rtpproxy in order to generate a couple of new port for each branch? My rtpproxy string in the kamailio conf (in manage branch section): rtpproxy_manage("bFAIE","192.168.10.10"); Brief extract of 2 SIP INVITE message send by kamailio to natted clients: 192.168.10.10 --> 192.168.10.20 User Datagram Protocol, Src Port: sip (5060), Dst Port: 6090 (6090) Session Initiation Protocol (INVITE) Request-Line: INVITE sip:1001@192.168.10.20:6090 SIP/2.0 Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 1004 1476 1657 IN IP4 192.168.10.10 Session Name (s): Talk Connection Information (c): IN IP4 192.168.10.10 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 62910 RTP/AVP 111 110 0 8 101 Media Description, name and address (m): video 47160 RTP/AVP 102 Media Attribute (a): nortpproxy:yes 192.168.10.10 --> 192.168.10.50 User Datagram Protocol, Src Port: sip (5060), Dst Port: 6090 (6090) Session Initiation Protocol (INVITE) Request-Line: INVITE sip:1002@192.168.10.50:6090 SIP/2.0 Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 1004 1476 1657 IN IP4 192.168.10.10 Session Name (s): Talk Connection Information (c): IN IP4 192.168.10.10 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 62910 RTP/AVP 111 110 0 8 101 Media Description, name and address (m): video 47160 RTP/AVP 102 Media Attribute (a): nortpproxy:yes

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Wrong from/to header compose way in print_to and print_from
by dongwf 23 Sep '14

23 Sep '14

Hi kamailio dev: I am writing a module to implement some features, my version is 4.1.2, I am not sure if it is a known issue, I found: 1. When I get a INVITE request with from/to like from:"display"<aaa(a)sip.com>, then I call tm's new_dlg_uas with the sip req, now the dlg's rem_uri set as value: "display"<aaa(a)sip.com>; 2. After that, when I need use the dlg to send BYE with tm's req_within with the above saved dlg, the print_to function will check the rem_uri's first value if "<" or not, now my uri has display name, so the outgoing BYE's to header becomes to:<"display"<aaa(a)sip.com>>, is it wrong? Should we check the last char in the rem_uri as ">"? The detail code is : static inline char* print_to(char* w, dlg_t* dialog, struct cell* t) { t->to.s = w; t->to.len = TO_LEN + dialog->rem_uri.len + CRLF_LEN + ((dialog->rem_uri.s[0]!='<')?2:0); memapp(w, TO, TO_LEN); if(dialog->rem_uri.s[0]!='<') memapp(w, "<", 1); memapp(w, dialog->rem_uri.s, dialog->rem_uri.len); if(dialog->rem_uri.s[0]!='<') memapp(w, ">", 1); ....

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Kamailio fronting Freeswitch authentication
by Ashwin Jain 23 Sep '14

23 Sep '14

Hi all, I am using Kamailio to front my freeswitch servers and it is working great. I am getting some problems with authentication, sometimes users which are not registered with kamailio are also able to make calls. I have a feeling that I am doing something really silly with kamailio routing. Following is the AUTH route that I am using, please let me know I am missing something very obvious. route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } I have set "WITH_AUTH" and "WITH_FREESWITCH". Let me know, if any more details are required. System configuration: Ubuntu: 12.04 LTS Kamailio: 4.1.5 (x86_64/linux) Freeswitch: 1.4.7 stable release -- Thanks and Regards, Ashwin Jain

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Siremis v4.1.0 Questions.
by Mahmoud Ramadan Ali 23 Sep '14

23 Sep '14

Hello everyone, Please i want more clarification about what i can do with the Siremis v4.1.0 Can i configure Kamailio to load balance between multiple Asterisk servers and RTP proxy using the Siresmis web interface or i have to interface with the Kamailio configuration file and do that manually...? Thanks in advance.

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User definable Ring Duration before forwarding
by Tim Chubb 23 Sep '14

23 Sep '14

Hi I was wondering if the following scenario is possible and if it is how I would go about implementing it: I would like to be able read from a database a list of destinations associated with a given dialled number which are tried in order of priority with each destination having a user definable ring duration before the next destination is tried. i.e. A Call is received from the PSTN, the number the caller dialled is looked up, it has 3 destinations attached to it, the users deskphone which is configured to ring for 10 seconds before the call is forwarded to their Mobile phone with a ring duration of 15 seconds, which if unanswered would be redirected to the main switchboard IVR. I think im looking for the equivalent of the Asterisk Dial app, where you can specify a timeout as an argument, is this possible to do with kamailio? All examples I have seen use a global timer value, which is unsuitable for this requirement, as each user could define n failover routes each with a distinct timeout before the next route is tried. Currently I have got as far as querying the database to retrieve the data but im a bit stumped as to how to handle the timeout variable. Any ideas? Tim Chubb

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Kamailio - Separated Servers
by Bruno Emer 22 Sep '14

22 Sep '14

Hello! I am relatively new to Kamailio and I'm trying to create a new enviroment using it in my company. I am thinking about use Amazon to host the servers and use OpsWorks to automatically escalate then if necessary. To accomplish this, my idea is to separate the servers, using one dedicated server to run as WebSocket, one to run as proxy and one as a Registrar. I'll be using just one database to store informations to all of my servers. With this, if I need more resources later, I can just create new servers with the specific roles (WebSockets, Proxy, Registrar). By now, the idea is clear, but the point is that I don't know how to separate the WebSockets server from the proxy server. Actually, I can do this, but when I have one agent using a regular softphone and one agent using WebSockets (with JSSIP) they are not able to establish a session if the softphone user starts it. Now, I want to know if is there a way to use two websockets servers, register users using both of then and start sessions between then, with a separated proxy and registrar. Has anyone done this before? Is possible to use kamailio like this? Thanks. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

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sr-users September 2014