sr-users

sr-users@lists.kamailio.org

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37563 discussions

old mailing lists serusers
by Pavel Kasparek 28 Apr '10

28 Apr '10

Hello all, the email address for old iptel.org mailing list serusers(a)lists.iptel.org (and serusers(a)iptel.org) is now forwarded to the new sip-router project mailing list sr-users(a)lists.sip-router.org. Pavel iptel.org

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old mailing lists serusers
by Pavel Kasparek 28 Apr '10

28 Apr '10

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old SER bugtracker turned off
by Pavel Kasparek 27 Apr '10

27 Apr '10

Hello all, the old SER bug tracking system http://tracker.iptel.org/ was shut down. There is now a small page at http://tracker.iptel.org/ with links to: - the sip-router project bugtracker http://sip-router.org/tracker/ - the new iptel.org bugtracker http://bugtracker.iptel.org/, where Sems and Serweb projects are now and which is used also for reporting problems with iptel.org public SIP service There is also a read-only archive of old SER bugs (exported from the old bugtracker), just for reference: http://tracker.iptel.org/jira-ser/ser-oldbugs.html Pavel

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[Serusers] old SER bugtracker turned off
by Pavel Kasparek 27 Apr '10

27 Apr '10

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termination with user/pwd
by Anders 27 Apr '10

27 Apr '10

I'm sure this has been asked before, but I couldn't find the answer anywhere when I googled it. I have a temporary situation where I need to terminate calls to a carrier with user/pwd authentication, and not through the IP authentication. As I understand it this is not a normal setup situation for a gw in Kamailio - but is there a way I can do this? Thanks, Anders

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Sintax errors can stop the machine?
by Bruno Machado 26 Apr '10

26 Apr '10

We are facing a periodically problem with a server that is running Kamailio 1.5.2. I think that it is not the case, but I will ask anyway. Some clients, sometimes, send us messages with sintax errors. This machine needs to be rebooted every week. It stops working completely. I don't think that SIP messages with this kind of errors can lead the machine to its 'death'. This proxy is very simple: it just receives and forwards the calls to pstn carriers. The routing rules are defined in the script, so we dont use modules to take the IPs and all other information about the carriers. The database is used just for ACC. Nothing more. Also, it is not receiving a flood atack. We have several servers running at this time with a huge number of clients and we never had this problem. This machine is the only one that does not want to work without a weekly break :) The causes, in my opnion, will be with the operational system or something physical. But, I would like to know if somebody already saw this behavior with SIP messages. tks a lot bruno machado

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B2BUA removing a=nortpproxy:yes (rtpproxy-1 <---RTP---> rtpproxy-2)
by Iñaki Baz Castillo 26 Apr '10

26 Apr '10

Hi, I must deal with a B2BUA which keeps the original SDP unchanged except the fact that it removes the "a=nortpproxy:yes" line added by RtpProxy. The B2BUA intercommunicates two Kamailio, both forcing its own RtpProxy server. I've not tested it again but expect there will be no RTP as each RtpProxy will wait for RTP coming from the other. Am I right? Of course I could force RtpProxy just in NAT cases and so, but I prefer to force it always so the clients of each Kamailio can configure QoS for the media addresses (different in each server). Is there some workaround for it? In the other hand: is it possible two different Kamailio's to share the same RtpProxy server? I've never tryied it. If it's valid, would it solve the B2BUA issue? (I don't think so as there are still different media sessions created by different kamailio instances). Any thought? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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htable size
by Alex Balashov 26 Apr '10

26 Apr '10

The htable size is static; it doesn't dynamically resize itself up to the next power of 2 after a certain utilisation threshold, right? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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Re-invite problems with t.38?
by Espen Berg 26 Apr '10

26 Apr '10

Hello all! I finally have something that I could call a working config (almost that is), the proxy works fine with voice and fax sent over g.711 (but I guess re-invites will cause problems in this scenario also). The problem occurs when I try to switch to T.38. Since it works with voice my guess is that the problem is caused by the re-invite to t.38. I guess it has to be solved by a if (loose_route()) section, but I am little bit clueless right now, I have tried multiple variants but with no success. RTP proxy is started with the -l flag to distinguish between local and public traffic. -l PUBLIC.IP.ETH0/LOCAL.IP.ETH1 The setup: Sonus UAC/Trunk <-private ip-> (eth1)Kamailio/RTPProxy(eth0) <-public ip-> Asterisk IP.TO.TRUNK = Sonus IP, public address range, but only reachable by private net (from the range of LOCAL.IP.ETH1). LOCAL.IP.ETH1 = Local net, static route to IP.TO.TRUNK. PUBLIC.IP.ETH0 = Public routable address. IP.TO.ASTERISK = Public routable address. The purpose of Kamailio/RTPProxy are to proxy everything between the routable and the private net. Does anyone here have any idea of how I'm going to solve my issue? It would also be nice if anyone here have better solutions to my approach, and there could also be some other errors in my config that causes this behaviour. ####### Routing ######## route { if (msg:len >= 2048) { sl_send_reply("513", "Message too big"); exit; } if (method != "INVITE" && method != "ACK" && method != "BYE" && method != "CANCEL") { sl_send_reply("503", "Service Unavailable"); exit; } if (!has_totag()) { record_route(); route("1"); } if (has_totag()) { if (!loose_route()) { if (t_check_trans()) { t_relay(); exit; } exit; } } if (has_body("application/sdp")) { if (method == "INVITE") { route("2"); t_on_reply("2"); } else if (method == "ACK") { rtpproxy_answer(); } } else if (method == "INVITE") { t_on_reply("2"); } if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); } t_relay(); } #RELAY BY SOURCE ROUTE route[1] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") { force_send_socket("PUBLIC.IP.ETH0"); rewritehost("IP.TO.ASTERISK"); } else { force_send_socket("LOCAL.IP.ETH1"); rewritehost("IP.TO.TRUNK"); } } # RTP OFFER ROUTE route[2] { if (src_ip == "IP.TO.TRUNK" || dst_ip == "LOCAL.IP.ETH1") force_rtp_proxy("eir"); else force_rtp_proxy("ier"); } # RTP OFFER ONREPLY onreply_route[1] { if (has_body("application/sdp")) route("2"); } # RTP ANSWER ONREPLY onreply_route[2] { if (has_body("application/sdp")) route("2"); } ## END OF CONFIG Sorry for the long post, but I hope someone outthere are able to help med with this issue. -- Espen C. Berg, Driftstekniker - Monsternett AS (www.monsternett.no) Violgata 3A, N-1776 HALDEN, NORWAY Sentralbord: 69701800 - Direkte: 69701810 - Faks: 69701801

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meeting in Berlin, June 7-8
by Daniel-Constantin Mierla 25 Apr '10

25 Apr '10

Hello, several developers will be around Berlin in June 7-8 time frame and we think to organize a face to face meeting. It is a good time frame if you want to attend Amoocon in Rostock (June 4-6, aprox 230km north of Berlin) or LinuxTag (June 9-12, in Berlin). http://www.linuxtag.org/2010/ http://www.amoocon.de/2010 I will have two presentations at Amoocon, about Asynchronous processing in SIP Router / Kamailio and extending your routing logic with Lua (via the new app_lua module). Hennig has one talk at LinuxTag about Geographical redundant VoIP systems with Kamailio. So far, interest in event and participation was shown by: - Andrei Pelinescu-Onciul - Jiri Kuthan - Inaki Baz Castillo - Jesus Rodriguez - Elena-Ramona Modroiu - Daniel-Constantin Mierla - Henning Westerholt - Marcus Hunger A good number to move it into public space and see if others from community are interested in such event. Based on that we will decide the type of event and look for a proper place to host it. Among topics could be: hands on hacking or docs update sessions, quick start guide to start writing your own extensions as module, open debates about current devel state and future directions... Please drop an email if you want to participate to the event - there is no participation fee - however, late registrations may be rejected if the capacity of location is filled - priority is first come first served. You can email me at miconda(a)gmail.com or to registration(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

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