- sr-users - kamailio.org

Re: [SR-Users] [sr-dev] Hello, please unsubscribe my email from the mail list, thank you.
by Iñaki Baz Castillo 30 Apr '10

30 Apr '10

2010/4/30 Lee Jacky <jacky.lee888(a)yahoo.com.cn>: > Hello,please unsubscribe my email:jacky.lee888@yahoo.com.cn from the mail list,thank you. List-Unsubscribe: <http://lists.sip-router.org/cgi-bin/mailman/options/sr-dev>, <mailto:sr-dev-request@lists.sip-router.org?subject=unsubscribe> -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Exotic case (but real) in which RtpProxy doesn't work
by Iñaki Baz Castillo 28 Apr '10

28 Apr '10

Hi, I've a problem with an Alcatel PBX in the way it performs transference: - Alcatel (behind NAT) sends INVITE to 111 and Kamailio forces RtpProxy. Let's assume the selected UDP port for RtpProxy is 1000. - Alcatel sends INVITE to 222 and Kamailio forces RtpProxy. Let's assume the selected UDP port for RtpProxy is 2000. Then Alcatel performs the transference as follows: - It sends a re-INVITE for 111 to Kamailio by setting the SDP to the IP or RtpProxy and port 2000. - It also sends a re-INVITE for 222 to Kamailio by setting the SDP to the IP or RtpProxy and port 1000. This is, Alcatel wants that the provider (me) sends the RTP to itself, while mantaining the original SIP dialogs established (so it's ok at signalling level, but at RTP level it cannot work as RtpProxy shoud send RTP to itself). I'm thinking on how to solve it but find no solution. Any suggestion? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[permissions] 'address' table and mask priorirty
by Iñaki Baz Castillo 28 Apr '10

28 Apr '10

Hi, some time ago I asked if allow_source_address_group() gives preference to the entries in 'address' table with lowest mask. I was replied that such case is not analyzed. But the fact is that in my tests it just works: Entry 1: - grp = 1 - ip_addr = 9.9.9.9 - mask = 32 Entry 2: - grp = 2 - ip_addr = 9.9.9.0 - mask = 24 I send a request from IP 9.9.9.9 and the function allow_source_address_group() always returns "1", it doesn't matter if the entry 1 appears before or after the entry 2, neither their "id" value. So it works as I expected (even if it's not explained in the documentation). However after checking the module code it seems that there are just two cases: - IP addresses (mask = 32). - Subnets (mask != 32). So first the source address is always matched against he address hash, and if it doesn't match then it is matched against the subnet hash, but in this last case it is not taken into account any kind of priority based on mask value, am I right? Could somebody confirm it please? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[LCR openser 1.2] Strange behavior in "lcr" table
by Iñaki Baz Castillo 28 Apr '10

28 Apr '10

Hi, I have configured LCR in an existing OpenSER 1.2 installation by filling the "lcr" table as follows: id prefix from_uri grp_id priority 1 NULL NULL 1 1 2 NULL NULL 2 2 In "gw" table there are some entries with "grp_id=1" and some others with "grp_id=2". Then in the script I just call "load_gws()" as usual, expecting that first is ALWAYS selected a gateway with grp_id=1. In fact I have ths *same* configuration working as expected in Kamailio 1.5. But for some reason, in OpenSer 1.2 for some calls "load_gws()" selects gateways with "grp_id=2", which IMHO doesn't make sense at all as grp_id 2 has priority 2 and its gateways would be selected always after those belonging to grp_id 1. Is there any bug or different behavior in LCR in openser 1.2? or do I miss something? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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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

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 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

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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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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[OT] Hitler should use Kamailio
by Iñaki Baz Castillo 24 Apr '10

24 Apr '10

http://www.youtube.com/watch?v=4nKh8bg3eHo -- Iñaki Baz Castillo <ibc(a)aliax.net>

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LCR in openser 1.2 - is it possible to disable a gateway?
by Iñaki Baz Castillo 24 Apr '10

24 Apr '10

Hi, I need to add LCR support to an existing OpenSer 1.2 server (not possible to upgrade it for now). I've realized that 1.2 doesn't include gateways monitorize feature. Ok, I can do the check with an external tool, but I would like some way to dissable such gw from the LCR module, something like upgrading the gw table and performing the MI reload function. But I don't see how to set in the gw table that such gateway is disabled. Is it possible? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Solutions to missing BYEs, accounting for them
by Alex Balashov 23 Apr '10

23 Apr '10

Hi all, Please forgive the slightly long post, but if you have anything to contribute on this topic, please consider giving it a read as I could really use your input. :-) As I'm sure many others of you running proxy-based service delivery platforms of some description also, I am faced with the problem of trying to account for calls with missing BYEs in a realistic way. There is no shortage of mailing list posts over the years on this topic. Inevitably, in a platform with sufficient call volume, with some NAT'd and/or endpoint diversity and other technical causes, there will be some calls that are never officially terminated from the point of view of a proxy. The ability of the 'dialog' module to spoof bidirectional BYEs on timeout[1] goes a long way toward addressing this problem theoretically. However, there are practical obstacles to relying on it solely as a solution, mainly because there is not an acceptable timeout value to use as a trade-off. If the timeout period is set to a very low value, users will obviously complain, and in any case, depending on the destination, the worst-case scenario for maximum call billing may still be far too high. If the timeout period is set high--perhaps something like 5-8 hours--then all calls that fail to end in the normal way will be billed some excessively large amount that certainly will not sit well with users either. If either the core delivery element of the platform or the user agent is tightly controlled by the operator of the proxy from an administrative point of view, it is indeed probably possible to rely on RTP timeouts or SIP Session Timers (SSTs) on one of the endpoints. That doesn't create a satisfying resolution for those of us dealing with indeterminate call completion scenarios with a great deal of user and vendor diversity, though. For instance, I route to about 15 ITSPs and carriers; I think maybe one of them does 15-minute SSTs, and the rest are certainly not going to turn them on just for me, even if their SBCs/switches/things have the capability. The user endpoints are mostly Asterisk and do RTP timeout, of course, and in most cases I do get the resulting BYE. However, this discussion is about the minute but nontrivial percentage of cases in which I do not get the BYE, whether because of NAT statekeeping problems or network reachability or whatever underlying causes--in truth, I cannot accurately characterise these. So, it seems to me that from a theoretical point of view, there are basically two directions someone in this position can go from here: 1) Inline B2BUA in the signaling path of all calls; 1a) Make it do SSTs; or 1b) Make it relay media, too, and hang up the call (bidirectional BYE) on RTP receive timeout; 2) Couple the proxy to an RTP relay and provide some mechanism by which the proxy can be made aware, in an asynchronous fashion, that an RTP timeout was detected by the relay. It seems to me from a brief and informal survey of prior mailing list literature that #1 is the usually recommended option here. If #1 is pursued, what is the best tool to use in the Kamailio/SIP-Router-oriented ecosystem? My default instinct would say SEMS; I really like SEMS, and use it a lot for various related chores. The problem is that the pre-built modules and examples for SEMS mostly center on application-level functionality, while low-level documentation of its powerful C++ API is a bit impoverished, so this would take a lot of work. Needless to say, I am interested in the option that requires the least work but still solves the problem in an elegant way from a technical and--dare I say--aesthetic perspective. For instance, it seems clear from looking at the SEMS-1.1.1 sources that SSTs are supported in principle in core/plug-in/session_timer. But unless I am missing something, I cannot find anywhere in the sources or examples where it is actually used. So, I suppose one option is to figure out how to make this stuff work in SEMS, and make it work. But for some reason who is not attune to the universe of its C++ API, it is a rather formidable chore. I think the same would hold true of making it observe bidirectional RTP timeout. Turning attention to option #2, I have looked at rtpproxy (my preferred default), iptrtpproxy, and mediaproxy modules but have not found any evidence that the control protocols Kamailio/SR uses to engage them support any notion of backward asynchronous feedback in case of RTP timeout. It would be really nice if one of these stream control protocols was augmented to kick back a packet to Kamailio that can be caught in a special event_route, like event_route[nathelper:rtp-stream-timeout], but that is clearly not the case today. To be honest, I would not use MediaProxy even if it had this feature, because, well, let's be bluntly honest and acknowledge what the more politically aware presumably already conjecture: in light of AG Projects' zealous OpenSIPS partnership, it's difficult to muster confidence in future compatibility of MediaProxy with Kamailio. The module is there, it works, and I'm sure its maintainers are dedicated to doing whatever it takes to reverse engineer and keep it working, lift patches from OpenSIPS as necessary, etc., but who wants to be on the wrong side of the project ecosystem fence? Not I. That leaves iptrtpproxy, whose 'switchboard' concept I do not fully comprehend due to lack of experience with it, but which holds a potentially viable, if slightly kludgy/Rube Goldbergian answer. Of the three RTP proxies, it is the only one that provides a ready means of exporting a list of media streams it is currently tracking, together with statistics on how many packets have been received, etc. It is not inconceivable to cook up an external process that will frequently check this 'switchboard', as it were, and incite Kamailio/SR to do dlg_bye() via MI if it appears that the media stream has disappeared from either side; the dialog module helpfully exports the MI command dlg_end_dlg. Still, this does not seem nearly as parsimonious and reliable a solution as simply building some kind of RTP stream leg timeout notification into the control socket. After all, the control socket is open persistently, right, not on-demand? The various RTP proxies all seem to have some kind of dead peer detection internally in order to have some means of gracefully expiring resources allocated to media streams that have gone away, so it would just be a matter of passing a control frame up the socket to Kamailio/SR and wiring that to a custom event_route or a more static callback in the code. By the way, I should mention that I am aware of and historically very sympathetic to the perspective that this kind of call control is alien to the nature of a proxy, and an appropriate job for UAs and not proxies at all. However, we all have to make pragmatic concessions to the realities of real-world operation, which I assume is the motivation for dialog timeouts, dlg_bye(), and other perversions from the point of view of a purist. :-) I welcome your thoughts and suggestions about the easiest and most technically meritorious approach. Thanks, -- Alex [1] Enabled via $dlg_ctx(timeout_bye) = 1 -- 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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MI functions
by Hector.Ortiz＠swisscom.com 23 Apr '10

23 Apr '10

Hi everyone, In the page where a module is described, there is a list of dependencies that have to be met in order to use the module, the list of exported parameters that might need to be set in the configuration file and a list of exported functions which can be used in the configuration script. I've noticed as well, that there are some "Exported MI functions" but I was wondering if any of you know what the purpose of those functions is, where and how such functions could be used? Thanks Héctor

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Error with Presence: no shmem
by Tristan Mahé 23 Apr '10

23 Apr '10

Hi List, I'm writing a little message because I encountered today a strange error on my Kamailio 3.0 proxy/presence server ( and can't ask on IRC there's nobody atm ;p ) I get these errors in the logs since a few hours: Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: tm [t_msgbuilder.c:1371]: build_uac_req(): no shmem Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: tm [uac.c:327]: t_uac: Error while building message Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: presence [notify.c:1577]: in function tmb.t_request_within Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: presence [notify.c:1673]: sending Notify not successful Apr 23 14:31:41 proxy-b /proxy/bin/kamailio/sbin/kamailio[29008]: ERROR: presence [notify.c:1348]: Could not send notify for dialog That prevents NOTIFYs from being sent, but I can also see sometimes NOTIFYs get sent without theses errors. The strange thing is that the error indicate no shared memory, but on the server, I'm far from being memless (16G total RAM, 4G free not even used for cache ). I have presence, presence_xml, presence_dialoginfo, pua, pua_dialoginfo and pua_bla loaded. Anyone got an idea on where to look ? Regards, Gled.

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[Kamailio-Users] How can i install and configuration CDRTool + Freeradius + mediaproxy + kamailio
by RAJNIKANT VANZA 23 Apr '10

23 Apr '10

Hi All, I would like to use CDRTool for accounting and integrate with kamailio + mediaproxy + freeradius. so, please let me know about configuration or installation steps for CDRToolon centos-5.X os. I have followed this link for CDRTool configuration http://cdrtool .ag-projects.com/wiki/Install but not success and confused. I have seen above url for CDRTool that provides information for deban os. but i want to install on centos. I am installed and run kamailio and mediaproxy successful. I am confused about compile or install about CDRTool and i want to help about Freeradius integration with kamailo and CDRTool. Thanks in advance. -- Best Regards, Rajnikant

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possible bug with dialog module
by Kelvin Chua 23 Apr '10

23 Apr '10

currently using kamailio 3.0 with around 1000 users. Apr 21 10:19:57 kamvm-1 /usr/local/sbin/kamailio[28511]: ERROR: dialog [dlg_handlers.c:218]: bad sip message or missing Contact hdr Apr 21 10:19:57 kamvm-1 /usr/local/sbin/kamailio[28511]: ERROR: dialog [dlg_handlers.c:350]: could not add further info to the dialog Apr 21 10:19:57 kamvm-1 kernel: kamailio[28511]: segfault at 0000000000000088 rip 00002ab61b62779a rsp 00007fffec157440 error 4 Apr 21 10:19:57 kamvm-1 /usr/local/sbin/kamailio[28507]: ALERT: <core> [main.c:722]: child process 28511 exited by a signal 11 Apr 21 10:19:57 kamvm-1 /usr/local/sbin/kamailio[28507]: ALERT: <core> [main.c:725]: core was generated Kelvin Chua

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new features: write access to branch attributes
by Daniel-Constantin Mierla 22 Apr '10

22 Apr '10

Hello, yesterday I committed to git new feature that allow write access to branch attributes. It is about branches kept in core before TM takes over and creates the branches for transaction. These branches are filled when you do location lookup, alias db lookup or append_branch in route block. You can update the attributes: - r-uri - outbound proxy address (dst uri) - path - q value - local socket to be used for sending - branch flags More at: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#branch_attribut… The access can be done with index, 0 is the first. Negative value of index starts counting from the last, therefore index -1 is the last branch. If the index is missing, then first branch is used. Adding a new branch can be done with km_append_branch(...) of kex module or append_branch() functions. Dropping a branch is by assigning $null to uri attribute of branch. Among benefits: - drop invalid branches before creating the transaction - you can lookup the aor from branches in location table, avoiding a loopback of a branch (e.g., for alias db case) Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

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Catching internally generated BYE event & other
by Alex Balashov 21 Apr '10

21 Apr '10

Hello, I am using the dialog module to time out excessively long calls: modparam("dialog", "default_timeout", 21600) # 6 hours modparam("dialog", "dlg_flag", 2) ... route { ... ... setflag(2); $dlg_ctx(timeout_bye) = 2; ... ... if(!t_relay()) sl_reply_error(); } I have two unrelated questions: 1) Why does $dlg_ctx(timeout_bye) work here? My understanding is that $dlg_ctx(...) is a PV namespace container that exposes some aspect of the allocated dialog tracking structure. The dialog structure is created via a TM->dialog callback triggered by t_relay(); why is it possible to access it prior to that point? 2) The spoofed BYEs that are generated by the proxy with this setting do not show up as sequential requests, nor are catchable in script at all. As a result, BYE events are not written to CDRs with 'acc' and I have no way of knowing when the call actually closed. What is the most efficient and canonical solution to this problem? I can think of two: (a) Define a $dlg_ctx(timeout_route) and have it write a BYE event artificially using acc_db_request(). (b) Loop the BYE request back to the proxy with some sort of differentiating URI parameter or similar that causes it to be dropped instead of forwarded. But this is really nasty. What is your suggestion? Thanks, -- 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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Appended headers are not added in serial forking
by Iñaki Baz Castillo 21 Apr '10

21 Apr '10

Hi, I receive a request, append some custom headers and use "load_gws()" (from LCR module). If the first gateway fails I do "next_gw()" and this second request to the second gateway doesn't contain the previously appended custom headers. Is it the expected behavior? -- Iñaki Baz Castillo <ibc(a)aliax.net>

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SQLOps question
by Maciej Bylica 21 Apr '10

21 Apr '10

Hello, I am trying to use SQLOps Module in my kamailio 1.5 configuration. According to your explanations every result of sql_query should be freed just after using that data. So it means that there will be better to fetch for THE SAME data three times instead of just one. Am I right? But nevertheless I've been trying to query just one and free the result of the sql_query at the end of the file. Could sb tell me if the result of sql query executed in the main part of the script is remembered and could be used on failure route as well? Thanks, Maciej.

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Re: [SR-Users] Unregister when tcp connection timeout
by Daniel-Constantin Mierla 20 Apr '10

20 Apr '10

please keep the mailing list cc-ed, others may benefit of this discussion, participate with ideas or solutions. On 4/20/10 9:18 PM, Bruno Bresciani wrote: > Thank you for your help Daniel... I was reading the source code to > understand the parameter "connection_timeout" in the TLS module 3.0.x > version and check if it help me too. When timer "connection_timeout" > expire, the function that die idle tls connection cold generate a user > unregister regarding this connection. That event happens in a different place than location service and a location record does not keep reference to the connection, therefore you need to do a bit of coding. Cheers, Daniel > > > 2010/4/20 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > > > On 4/20/10 8:29 PM, Daniel-Constantin Mierla wrote: > > > > On 4/20/10 7:30 PM, Iñaki Baz Castillo wrote: > > 2010/4/20 Bruno Bresciani<bruno.bresciani(a)gmail.com > <mailto:bruno.bresciani@gmail.com>>: > > Hi people, this is my first time that I writing to > this list and I nedd some > help... When kamailio lost connection with a user > registred over TLS > (timeout for examlple), I want this user be unregister > in kamailio too. How > can I make this unregister? > > It's a commendable feature request, but it's not > implemented right now. > > > it might now be easy to catch connection lost (forward or wait > state case). > > > ^^^ I meant: it might NOT be easy ... > > E.g., for forward case, detection is after config processing, when > trying to relay. If there are many branches, it is not even > reported if one branch succeeds or sends another negative reply code. > > > > Also, unregistering a contact is not really following the rfc, > anyhow, this is not something complex to add if you have the > contact details. > > What you can do right now is to avoid trying to send if the > connection is not longer open, see: > http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#set_forward_n… > > > Contact is in memory, but no penalty for performances. Very > useful for NAT cases or when UA is supposed to keep the > connection open. > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla * http://www.asipto.com/ * > http://twitter.com/miconda * > http://www.linkedin.com/in/danielconstantinmierla > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

Unregister when tcp connection timeout
by Bruno Bresciani 20 Apr '10

20 Apr '10

Hi people, this is my first time that I writing to this list and I nedd some help... When kamailio lost connection with a user registred over TLS (timeout for examlple), I want this user be unregister in kamailio too. How can I make this unregister? Thank you...

3 3

PVs in Regular Expression Transformations
by jh＠tutpro.com 20 Apr '10

20 Apr '10

i tried to embed a PV in regex transformation, but didn't get desired result. is it possible or must regex only contain strings. here is an example, where the idea is to strip string "456" from the end of string "123456": $var(test) = "123456"; $var(rest) = "456"; $var(test) = $(var(test){re.subst,/^(.*)$var(rest)$/\1/}); xlog("result is $var(test)\n"); unfortunately, the result is 0 rather than "123". i also tried $var(test) = $(var(test){re.subst,/^(.*)\$var(rest)$/\1/}); and $var(test) = $(var(test){re.subst,/^(.*)$(var(rest))$/\1/}); but still 0. -- juha

2 16

[Kamailio-Users] new module: embedded lua
by Daniel-Constantin Mierla 20 Apr '10

20 Apr '10

Hello, few weeks ago I uploaded a new module named app_lua, providing functionality to execute Lua (http://www.lua.org) scripts in your configuration file. Lua is small and fast, therefore fits very well in a sip proxy environment. It has a lot of extensions, so when you cannot do something with existing cfg functions, check this one as well. The readme and the api are available at: http://sip-router.org/wiki/api/lua/devel http://sip-router.org/docbook/sip-router/branch/master/modules/app_lua/app_… The API will be enhanced during this development cycle, your feedback will help a lot in this direction. By now you get access to pseudo-variables, sip headers, logging functions and sl module. For new comers in community, if you are a Perl fan, then you can use the perl module in a similar way. Also, another module was uploaded recently, app_python, allowing to write python extensions. Module and API for Python are not yet documented, but I think this is not a barrier for a true Python fan, actually I hope will be a motivation to contribute the docs :-) Cheers, Daniel -- Daniel-Constantin Mierla SIP Server Professional Solutions * http://www.asipto.com/

1 1

Striping Character from $rU
by dotnetdub 19 Apr '10

19 Apr '10

Hi, Running 1.53 and also 3.01 I know there are operators to strip() and strip_tail() to remove characters from the front and end of a uri - what would people recommend to make sure that the uri only ever contains numerical value? If I wanted to for example strip the - character what would be the best way? TIA. Stephen.

3 3

[Fwd: Re: [Kamailio-Users] Possible timing/latency relatedproblem ?]
by Jeff Brower 19 Apr '10

19 Apr '10

Juha- Vikram had posted a while back on the thread where he explained a bit more about his resolution and the fact the issue was occurring with VoipSwitch but not asterisk. Let us know if you want a link to that. Thanks for following up. -Jeff -------------------- Original Message ----------------- Subject: Re: [SR-Users] [Kamailio-Users] Possible timing/latency relatedproblem ? From: "Vikram Ragukumar" <vragukumar(a)signalogic.com> Date: Mon, April 19, 2010 11:15 am To: "Juha Heinanen" <jh(a)tutpro.com> Cc: users(a)lists.kamailio.org ------------------------------------------------------- Juha, > what was the conclusion regarding this? did the problem go > away when you called nat traversal functions both on 180 > ringing and 183 session progress? Yes. The problem did go away once we started forcing rtpproxy on 180 Ringing also. Thanks and Regards, Vikram.

1 0

[Kamailio-Users] Possible timing/latency related problem ?
by Vikram Ragukumar 19 Apr '10

19 Apr '10

Hello, Scenario 1 ---------- --------- -------- -------- -------- -------- |Cellphone|--|Wireless|--|Internet|--|Kamailio|--|Asterisk| | | |Router | | | |rtpproxy| | | --------- -------- -------- -------- -------- |<-Lowlatency->| Scenario 2 ---------- --------- -------- -------- -------- -------- |Cellphone|--|Wireless|---|Internet|--|Kamailio|--|Asterisk| | | |Router | | | |rtpproxy| | | --------- -------- -------- -------- -------- |<-Highlatency->| Both scenario 1 and scenario 2 are similar in setup, excepting that in scenario 2 there is a greater latency involved in SIP and RTP transmission between the wireless router and Kamailio+rtpproxy server. In scenario 1 calls from the cellphone using a SIP softphone app go through 100% of the time with both endpoints of the call being audible. However in scenario 2, calls from the cellphone go through with both endpoints of the call being audible only sometimes, and during all other attempts, call goes through with no voice from the cellphone. In scenario 2 the SIP softphone can always register, the SDP connection address is always rewritten to use rtpproxy so it does not appear to be a NAT issue, rtpproxy's logs show a session being set up and torn down, with RTP stats for caller being 0. In both scenarios, the equipment and their configuration are identical. (Both cellphone's are Nokia e51, router's are Netgear wgr614) What might be probable causes for the behavior seen in scenario 2 ? Could there be a timing/latency related issue ? Any pointers on how to approach this problem will be appreciated. Thanks and Regards, Vikram.

5 10

loadbalancer
by Hector Muñoz 19 Apr '10

19 Apr '10

Hi all, I have a kamailio sip server with two trunk connections to two asterisk servers. The register server is kamailio. I'd like to be able to configure a load balancing with kamailio between two asterisk servers. How can i do that? Do you know any good tutorial for configure load balancing with dispatcher module in kamalio? Thanks in advance. Regards. -- jabber: triptik(a)12jabber.com blog: http://impresionesdeunloco.wordpress.com

1 0

sr-trunk with mysql support
by Meftah Tayeb 17 Apr '10

17 Apr '10

hi dear all, i'm trying to use latest trunk with mysql i have down compiling kamailio now i don't know ghow do i enable mysql for kamailio (not for kamdbctl) the db is created successfully but kamailio is unable to connect to the db default user/pass, i don't change anything how do i use this define stufs? thanks

3 3

load balancing redirect
by Benjamin Lawetz 17 Apr '10

17 Apr '10

Hi, I'm trying to setup Kamailio to act as a load-balancer in front of asterisk servers. I want it to do the following: - The UA sends an invite to Kamailio - Kamailio sends a 30x redirect to the UA (dispatcher) - The UA sends it's invite to the asterisk server I've seen a couple of examples, but doesn't seem to work for me. Here's the relevant config: route{ if (method=="REGISTER") { [...clip...] }; if (method=="INVITE") { ds_select_domain("2","8"); #ds_select_dst("2","8"); sl_send_reply("300","Redirect"); exit; }; } What seems to happen is that Kamailio does send the redirect, but with no indication of where to go. ds_select_domain (or ds_select_dst ... I having trouble understanding the difference like many people it seems) I'm guessing selects an appropriate uri, but doesn't put it the contact for the redirect. Is there anyway to see what was set by the ds_select_dst? what pv would have this value? What do I need to set prior to sending a redirect reply for it to be properly forwarded? BTW: using Kamailio 3.0.1 Thank you, -- Benjamin

4 6

wiki: upgrade from 3.0.x to 3.1.x
by Daniel-Constantin Mierla 17 Apr '10

17 Apr '10

Hello, I started the page that should collect notes about how to upgrade from version 3.0.x to 3.1.0: http://sip-router.org/wiki/install/3.0.x-to-3.1.x 3.1.0 is not yet released, target date is September 2010, right now is the code on master branch in GIT repository. However, this page will help people that want to test it or use some modules from it, making as well the life easier before release. Please add to the document as you change (or discover) something breaking compatibility with 3.0.x - the wiki is open. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

reply when drop() in branch route
by jh＠tutpro.com 17 Apr '10

17 Apr '10

i call drop() in branch route and it is the only branch left. the branch gets correctly dropped, i.e., the request is not send out, but the reply to UAC is strange: U 2010/04/14 21:03:43.750712 192.98.102.10:5060 -> 192.98.103.2:5074 SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM). Via: SIP/2.0/UDP 192.98.103.2:5074;rport=5074;branch=z9hG4bKaoxzewgj. To: <sip:0407058055@192.98.102.10>;tag=c02670ad1171fe45d9ff9a27d6c2cb82-e03b. From: <sip:+35816234567@foo.bar>;tag=bclsw. Call-ID: doibilvynrtqxpk@localhost. CSeq: 134 INVITE. Server: SIP Proxy (3.0.99-dev1 (i386/linux)). Content-Length: 0. why this kind of reply, because the proxy didn't even try to send the request to next ho? is it possible to somehow choose in the script, which reply to send? in this example, i'm using lcr to choose gws and none of them was suitable, so something like '503 service not available' would be more appropriate. -- juha

3 15

tm response issue
by Klaus Darilion 16 Apr '10

16 Apr '10

Hi! Kamailio does not forward the received 503 response but instead generates a new 500 response. Failure routes are not activated. U 2010/04/14 15:08:26.201865 83.136.22.111:5060 -> 83.136.32.159:5060 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 83.136.32.159;branch=z9hG4bK2a5a.0d5ea987.0;received=83.136.32.159 Via: SIP/2.0/UDP 10.10.0.51:5086;received=83.136.33.3;rport=5086;branch=z9hG4bK2901533893 From: <sip:klaus.darilion@labs.nic.at>;tag=1939498352 To: <sip:01888801505641636@labs.nic.at>;tag=as1b74528e Call-ID: 3387977514 CSeq: 21 INVITE Server: Asterisk PBX 1.6.2.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 U 2010/04/14 15:08:26.201865 83.136.32.159:5060 -> 83.136.22.111:5060 ACK sip:01888801505641636@labs.nic.at SIP/2.0 Via: SIP/2.0/UDP 83.136.32.159;branch=z9hG4bK2a5a.0d5ea987.0 From: <sip:klaus.darilion@labs.nic.at>;tag=1939498352 To: <sip:01888801505641636@labs.nic.at>;tag=as1b74528e Call-ID: 3387977514 CSeq: 21 ACK Max-Forwards: 69 Content-Length: 0 # U 2010/04/14 15:08:26.201865 83.136.32.159:5060 -> 83.136.33.3:5086 SIP/2.0 500 Service Unavailable Via: SIP/2.0/UDP 10.10.0.51:5086;rport=5086;branch=z9hG4bK2901533893;received=83.136.33.3 From: <sip:klaus.darilion@labs.nic.at>;tag=1939498352 To: <sip:01888801505641636@labs.nic.at>;tag=c3829fdbb3bfc7416671e56cae8e633d-a9b5 Call-ID: 3387977514 CSeq: 21 INVITE X-Info: (83.136.32.159:1035) Enforcing NAT traversal for local caller X-Info: (83.136.32.159:1035) Enforcing symmetric response routing Server: kamailio (3.0.1 (i386/linux)) Content-Length: 0 Is this a feature or a bug? thanks klaus

2 1

Re: [SR-Users] load balancing redirect
by Iñaki Baz Castillo 16 Apr '10

16 Apr '10

2010/4/16 Benjamin Lawetz <benjaminlawetz(a)gmail.com>: >> I strongly suggest not to do that. Instead: >> >> - The UA sends an invite to Kamailio >> - Kamailio uses LCR module to randomly select asterik-1 or asterisk-2. >> - Kamailio sends the request to the selected asterisk. >> - If there is a failure (to inspect in failure_route) Kamailio uses >> next_ge() (LCR module) and routes the INVITE to the other asterisk. >> >> This will work really better. > > Good idea, hadn't thought of a retry for the invite (was thinking of > removing the server from the dispatch if it failed, but your idea is > better). > I'm guessing this doesn't add too much load to Kamailio. > For learning/understanding the dispatcher... was what I wanted to do possible? Please, keep the mail within the maillist thread. No, it doesn't add too much load to Kamailio. This is one of the major purposes of a transaction stateful proxy (the serial forking in order to implement failover). Take a look to the LCR module. It gives you load balancing (with priorities if you want) and failover features. It's all you need in your scenario. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

call not being accounted when rejected by global blacklist
by Panagiotis Skoulikaritis 15 Apr '10

15 Apr '10

Dear list members I'm using kamailio 3 and the problem that I'm facing is that when a call is rejected by global blacklist, kamailio does not generate an accounting record. Is this normal behavior or I need to configure something somewhere ? Below are parts of the config file # ----- acc params ----- modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 0) modparam("acc", "failed_transaction_flag", 3) #!ifdef WITH_ACCDB modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "mysql://username:password@localhost/openser") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif ---------------------- if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } ------------------------- if (is_method("INVITE")) { setflag(1); # do accounting setflag(2); # log missed calls setflag(3); # log failed calls setflag(4); # do siptrace setflag(5); # log dialog ------------- route[name] { # --- Remove # or %23 from rU --- bm_start_timer("hash_remove"); if($rU =~ "^.*%+") { xlog("------- The number contains %23 ---- "); $rU = $(rU{re.subst,/^(.*)%23(.*)/\1\2/}); xlog(" ------- The perl $rU ------- "); } if($rU =~ "^.*#+") { xlog("------- The number contains # ----"); $rU = $(rU{re.subst,/^(.*)#(.*)/\1\2/}); xlog("------- The perl $rU ------- "); } bm_log_timer("hash_remove"); # --- PDT section ------ get service prefix to be appended before is send out bm_start_timer("pdt_lookup"); if(!prefix2domain("2", "0")) { sl_send_reply("404", "Prefix not found"); exit; } $var(dstgrp) = 0; $var(dstgrp) = $(rd{s.int}); if($var(dstgrp)==0) { sl_send_reply("404", "Service not found"); exit; } $avp(s:service_prefix) = $var(dstgrp); xlog("--$rU----------- $avp(s:service_prefix) --- PDT ---------\n"); bm_log_timer("pdt_lookup"); # --- DialPlan section --- get the routing group id xlog("----- The rU is $rU before DP -----\n"); bm_start_timer("dialplan_lookup"); if(dp_translate("1", "$rU/$rU")) { $avp(s:dstgrp) = $(var(dpattrs){s.int}); xlog("----- The destination Drouting GroupID is $avp(s:dstgrp) -----\n "); } bm_log_timer("dialplan_lookup"); xlog("----- The rU is $rU after DP -----\n"); # --- Global Blacklist section --- bm_start_timer("blacklist_lookup"); if(!check_blacklist("globalblacklist")){ xlog("----- The number is blacklisted globaly! -----"); sl_send_reply("403", "Forbidden"); exit; } bm_log_timer("blacklist_lookup"); # --- Dynamic routing section --- route the call based on the routing group id bm_start_timer("drouting_lookup"); if (!do_routing("$avp(s:dstgrp)")) { xlog("----- DRouting don't match dstgrp $avp(s:dstgrp) -----\n"); sl_send_reply("404","Route Not Found -$rU-"); exit; } bm_log_timer("drouting_lookup"); xlog("-----DRouting with $rU and $avp(s:dstgrp) -and $avp(s:service_prefix)----\n"); # Append service prefix in front of number $ru = "sip:" + $avp(s:service_prefix) + $rU + "@" + $rd; xlog(" ------ $ru ----$rU- $avp(s:user)---\n"); t_on_failure("FAIL_ONE"); route(RELAY); exit; return; } Regards Panagiotis

3 6

[Kamailio-Users] PKG memory issues
by Iñaki Baz Castillo 15 Apr '10

15 Apr '10

Hi, I've experimented PKG out of memory in a production server. Of course I'm already increasing such value but I would like to know if the current settings and traffic could run into PKG memory issues: - Dual core INTEL XEON 3.00 GHz server. - Kamailio PKG memory = 4 MB. - Kamailio Shared memory = 64 MB. - Kamailio just listens in a single UDP port (8 childrens). - Just INVITE method is handled (no registration, no subscription). The script does the following for each request: - 'permissions' module to match source IP (just ~20 entries in 'address' table). - 2 custom SQL queries (returning a simple value). - 'dialog' memory (just in memory). - 'uac' module to change From header. - There are 10 AVP's set per transaction. - 'lcr' module for routing to two gateways (just 2 entries in 'lcr' table). - 'rtpproxy' is forced for every call. The server has been working properly for months but these days the traffic has been duplicated, having ~400 simultaneous calls in peak hours. Also note that such calls come from callcenters so they are ver "fast". With this environment, is it normal to get into PKG memory issues (4 MB)? I understand that it makes sense, but I would like to hear some opinions. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 14

minutes IRC meeting 2010-04-14
by Henning Westerholt 15 Apr '10

15 Apr '10

Hello, the minutes of the developer IRC meeting from yesterday are now available in our wiki: http://sip-router.org/wiki/devel/irc-meetings/2010-04-14-minutes Best regards, Henning -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

Hostname in VIA on Forward
by Eric Hiller 15 Apr '10

15 Apr '10

Hello all, I need to have kamailio put its hostname on its via append instead of its IP address. How can I do this or is it possible? Thank you,Eric _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:…

3 5

Re: [SR-Users] Failure to load module db_mysql.so due to undefined symbol: log
by Pratab Ali 14 Apr '10

14 Apr '10

Hello Marius, Yes, I can confirm that MySQL-devel-community-5.1.43-1.sles10.i586.rpm was also installed. I think the log_10_int is included by something that's linked in and has nothing to do with MySQL. A colleague and friend has installed on a Debian system with success. I am about to install Debian over my SuSE installation. Thanks Pratab. -----Original Message----- From: marius zbihlei [mailto:marius.zbihlei@1and1.ro] Sent: 14 April 2010 14:34 To: Pratab Ali Subject: Re: [SR-Users] Failure to load module db_mysql.so due to undefined symbol: log Pratab Ali wrote: > Hello > > My version of libmysqlclient was installed from this file > MySQL-client-community-5.1.43-1.sles10.i586.rpm. So, that'd make it > version 5.1.43.1 I'd guess. > > Hello, I presume you have also installed MySQL-devel-community-5.1.43-1.sles10.i586.rpm <http://rpm.pbone.net/index.php3/stat/4/idpl/13531085/dir/other/com/MySQ L-devel-community-5.1.43-1.sles10.i586.rpm.html> for the development headers and libraries ... Can you check if this is also installed . I have tested on Debian with libmysqlclient-dev 5.1.45 and it works. I still wonder about the log_10_int symbol... Marius > Unfortunately, it is the config file parser that is detecting the > unresolved symbol and so refusing to run kamailio at all. > > Thanks > Pratab > >

3 7

reminder: irc devel meeting, apr 14, 15:00utc
by Daniel-Constantin Mierla 14 Apr '10

14 Apr '10

Hello, short reminders about tomorrow's IRC devel meeting on #sip-router / irc.freenode.net. Add your topics to wiki if you want something to be discussed: http://sip-router.org/wiki/devel/irc-meetings/next Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 2

syntax error with kamailio
by alexis heron 14 Apr '10

14 Apr '10

Hi, I have a problem to routing with kamailio. When I restart kamailio I have this error message : 0(710) : <core> [cfg.y:3328]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 640, column 2-3: syntax error 0(710) : <core> [cfg.y:3328]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 640, column 2-3: ERROR: bad config file (2 errors) here is my line 640 : if (uri=~"sip:9[0-9][0-9]@.*") { //line 640 log(1, "Matched Cisco Call Manager"); route(4); }; Can you help me please thank you

7 22

[NAT] BYE not received to caller
by Pascal Maugeri 14 Apr '10

14 Apr '10

Hi I need some "guidelines" to troubleshoot the following issue: a) A is behind NAT b) B is not behind NAT c) A calls B, SIP INVITE is sent over TCP d) A's firewall does NAT and changes the source port to let's say p1 e) B releases the call and sends BYE over UDP f) Kamailio sends the BYE to A, over UDP, to the NATed source port p1 ?! 2 comments: - This scenario works perfectly when A is the one who disconnects the call. and of course when no NAT is involved everything works ok - if works when I comment the line "fix_nated_contact()" in the route[NAT] block: route[NAT]{ #!ifdef WITH_NAT force_rport(); #if (nat_uac_test("19")) { if (nat_uac_test("3")) { if (method=="REGISTER") { setbflag(10); fix_nated_register(); } else { #fix_nated_contact(); [...] Any hint is very welcome. Cheers Pascal

3 7

Failure to load module db_mysql.so due to undefined symbol: log
by Pratab Ali 14 Apr '10

14 Apr '10

Hello, I have installed from source version 3.0.1 of Kamailio on an OpenSuSE 10 Linux system. Currently I am attempting to enable persistence in my Kamailio with the help of MySQL. In the config file kamailio.cfg I have enabled MySQL as follows #!define WITH_MYSQL In the config file db_mysql.so is the first module to be loaded by calling loadmodule, but the module fails to load. By turning, Kamailio, debugging on I get the following explanation. 0(9858) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/local/lib/kamailio/modules/db_mysql.so> 0(9858) ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/db_mysql.so>: /usr/local/lib/kamailio/modules/db_mysql.so: undefined symbol: log Using the unix/linux tool 'nm' I see the following. mindy:~/tmp # nm /usr/local/lib/kamailio/modules/db_mysql.so | grep log U getlogin@@GLIBC_2.0 U log 001a6e40 R log_10 001e5d00 D log_10_int U log_level_info U log_stderr U syslog@@GLIBC_2.0 And for sure the capital 'U' means undefined. I have also searched the source code that came with the kamailio package, for definitions of 'log'. And it also looks like I'll be in trouble later for log_level_info, log_level_stderr and syslog are undefined too. However, log_level_info I have just found in source files in the kamailio package. So, why isn't it linking in? Unfortunately, searching on the internet I have not yet come across a forum post that tackles the above issue. As, I suspect everyone else has got the build right. Have I built db_mysql.so incorrectly? Though I built it using the make files that came with the package. Is it a case of linking in the appropriate libraries to resolve these undefined symbols? Finally, I have a colleague who has used the same version of kamailio but installed on a Debian lenny distro. He has been successful in getting db_mysql loaded and persistence enabled. However, he has an older version of mysql (by only one minor-version number) though. Having looked that the changes' history for the two mysql versions, I see no obvious reasons that could be the cause of my failure. By comparing notes, he did nothing too different than I did. We both followed the instructions in the INSTALL and README files. There is no particular reason for me using OpenSuSE 10 other than it was already installed and the computer was not being used by anyone. I am considering installing Debian Lenny on my computer to match my colleague, would this be prudent or a waste of time? i.e I may face my same problem because I'm missing a stage? Regards and thank you, in advance, for any help given. Pratab

3 5

[Kamailio-Users] Relaying NOTIFY UDP messages over TCP
by Pascal Maugeri 14 Apr '10

14 Apr '10

Hi We need to force some NOTIFY messages going through our kamailio instance over UDP to be sent over TCP. For that purpose we inserted the following relaying rule at the beginning of the routing script: route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(msg:len>1300 && is_method("NOTIFY") && proto==UDP) { $du = "sip:" + $(ruri{uri.host}) + ":" + $(ruri{uri.port}) + ";transport=tcp"; t_relay(); exit; } [...] This works well if the destination of the NOTIFY message is not behind NAT. We're looking for a solution for the NAT case. We did some tests using lookup("location") before changing $du but we did not get any success. Do you have any idea on how to implement this relaying properly ? Is it acceptable to change the $du and then execute the t_relay as we did ? Cheers Pascal

3 7

hox to a SIPtrunk with kamailio 3.0.0
by alexis heron 14 Apr '10

14 Apr '10

Hi, I would like to make a SIPTrunk with kamailio to redirect to a 9xx numbers in Cisco Call manager, can you help me please. Here's what I did but it does not work. if (uri=~"sip:9[0-9][0-9]@ccm-ipv6.crihan.fr") { log(1, "Matched Cisco Call Manager"); route(4); }; route(4) { log(1, "In route[4]"); rewritehostport("195.221.29.145.fr:5060"); append_hf("P-hint: Forwarded to CCM\r\n"); xlog("L_INFO", "$rm from $fu to $tu"); t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; }

4 6

when mySQL is unreachable ...
by Alex rsm 14 Apr '10

14 Apr '10

Hi, My Kamailio server uses an external mysql database with perl script to proxy traffic. When mysql server is unreachable, Kamailio do not respond with 100 Trying and instead responses with 484 (Address Incomplete) to the INVITE message. Is this a normal behavior? I need to response with 503 when mysql is unreachable. Can I do it in kamailio.cfg or I need to implement it in the perl script? Thanks, A R _________________________________________________________________ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccount&ocid=PID28…

3 2

RTP destination address uncorrect
by xiangfeng_006 14 Apr '10

14 Apr '10

  Hi all: I am trying to setup follow IMS test environment：                          |----------------|                          |   kamailo      |-----------SIP--------->SIP server （10.100.1.12） 8700                  |                   |                                            |                                            7700 UA1-SIP/RTP-->|                   |                                            |--------------SIP/RTP---------->UA （10.100.1.51） 10.100.1.41        |                   |                                            |                          | RTP proxy   |------------RTP------->Media Gateway（10.100.1.15）                          |----------------|                             PC 10.100.1.10 SIP signaling： 1. UA1 send INVITE to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 session name: A conversation connection information: IN IP4 10.100.1.41 2. PC send INVITE to SIP server message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 connection information: IN IP4 10.100.1.10 .............100/183-------------------- 3. SIP server send 200 OK to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1 message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.10 after ACK, UA1 and UA2 are in calling, but no any voice I found when PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15 How to set PC (RTP proxy) transfer RTP data packages to 10.100.1.15( Media Gateway address) ? thanks !

1 0

set kamailo with RTP proxy as redirect server
by xiangfeng_006 14 Apr '10

14 Apr '10

Hi all: I am trying to setup follow IMS test environment：                          |----------------|                          |   kamailo      |-----------SIP--------->SIP server （10.100.1.12） 8700                  |                   |                                            |                                            7700 UA1-SIP/RTP-->|                   |                                            |--------------SIP/RTP---------->UA （10.100.1.51） 10.100.1.41        |                   |                                            |                          | RTP proxy   |------------RTP------->Media Gateway（10.100.1.15）                          |----------------|                             PC 10.100.1.10 SIP signaling： 1. UA1 send INVITE to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 session name: A conversation connection information: IN IP4 10.100.1.41 2. PC send INVITE to SIP server message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 connection information: IN IP4 10.100.1.10 .............100/183-------------------- 3. SIP server send 200 OK to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1 message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.10 after ACK, UA1 and UA2 are in calling, but no any voice I found when PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15 How to set PC (RTP proxy) transfer RTP data packages to 10.100.1.15( Media Gateway address) ? thanks !

1 0

set kamailo with RTP proxy as redirect server
by xiangfeng_006 14 Apr '10

14 Apr '10

Hi all: I am trying to setup follow IMS test environment：                          |----------------|                          |   kamailo      |-----------SIP--------->SIP server （10.100.1.12） 8700                  |                   |                                            |                                            7700 UA1-SIP/RTP-->|                   |                                            |--------------SIP/RTP---------->UA （10.100.1.51） 10.100.1.41        |                   |                                            |                          | RTP proxy   |------------RTP------->Media Gateway（10.100.1.15）                          |----------------|                             PC 10.100.1.10 SIP signaling： 1. UA1 send INVITE to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 session name: A conversation connection information: IN IP4 10.100.1.41 2. PC send INVITE to SIP server message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 connection information: IN IP4 10.100.1.10 .............100/183-------------------- 3. SIP server send 200 OK to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1 message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.10 after ACK, UA1 and UA2 are in calling, but no any voice I found when PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15 How to set PC (RTP proxy) transfer RTP data packages to 10.100.1.15( Media Gateway address) ? thanks !

1 0

set kamailo with RTP proxy as redirect server
by xiangfeng_006 14 Apr '10

14 Apr '10

Hi all: I am trying to setup follow IMS test environment： |----------------| | kamailo |-----------SIP--------->SIP server （10.100.1.12） 8700 | | | | 7700 UA1----SIP/RTP----> | | | |--------------SIP/RTP---------->UA （10.100.1.51） 10.100.1.41 | | | | | RTP proxy |------------RTP------->Media Gateway（10.100.1.15） |----------------| PC 10.100.1.10 SIP signaling： 1. UA1 send INVITE to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 session name: A conversation connection information: IN IP4 10.100.1.41 2. PC send INVITE to SIP server message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 connection information: IN IP4 10.100.1.10 .............100/183-------------------- 3. SIP server send 200 OK to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1 message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.10 after ACK, UA1 and UA2 are in calling, but no any voice I found when PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15 How to set PC (RTP proxy) transfer RTP data packages to 10.100.1.15( Media Gateway address) ? thanks !

1 0

set kamailo with RTP proxy as redirect server
by xiangfeng_006 14 Apr '10

14 Apr '10

 Hi all:   I am trying to setup follow IMS test environment：                                     |----------------|                                     |   kamailo      |-----------SIP--------->SIP server （10.100.1.12）    8700                          |        |           |                                       |                                           7700     UA1----SIP/RTP----> |        |           |                                       |--------------SIP/RTP---------->UA （10.100.1.51）     10.100.1.41               |        |           |                                       |                                     |   RTP proxy |------------RTP------->Media Gateway（10.100.1.15）                                     |----------------|                                                                                PC 10.100.1.10                                        SIP signaling：  1.  UA1 send INVITE  to PC          message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41                          session name: A conversation                          connection  information: IN IP4 10.100.1.41  2.  PC send INVITE to SIP server          message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41                           connection  information: IN IP4 10.100.1.10 .............100/183-------------------- 3.   SIP server  send  200 OK  to PC                  message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15                                                  connection  information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1                  message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15                          connection  information: IN IP4 10.100.1.10   after ACK,  UA1 and UA2 are in calling, but no any voice   I found  when  PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15            How to set PC (RTP proxy)  transfer RTP data packages  to 10.100.1.15( Media Gateway address) ?    thanks !

1 0

set kamailo with RTP proxy as redirect server
by xiangfeng_006 14 Apr '10

14 Apr '10

 Hi all:   I am trying to setup follow IMS test environment：                                     |----------------|                                     |   kamailo      |-----------SIP--------->SIP server （10.100.1.12）    8700                          |        |           |                                       |                                           7700     UA1----SIP/RTP----> |        |           |                                       |--------------SIP/RTP---------->UA （10.100.1.51）     10.100.1.41               |        |           |                                       |                                     |   RTP proxy |------------RTP------->Media Gateway（10.100.1.15）                                     |----------------|                                                                                PC 10.100.1.10                                        SIP signaling：  1.  UA1 send INVITE  to PC          message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41                          session name: A conversation                          connection  information: IN IP4 10.100.1.41  2.  PC send INVITE to SIP server          message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41                           connection  information: IN IP4 10.100.1.10 .............100/183-------------------- 3.   SIP server  send  200 OK  to PC                  message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15                                                  connection  information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1                  message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15                          connection  information: IN IP4 10.100.1.10   after ACK,  UA1 and UA2 are in calling, but no any voice   I found  when  PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15            How to set PC (RTP proxy)  transfer RTP data packages  to 10.100.1.15( Media Gateway address) ?    thanks !

1 0

when mySQL is unreachable ...
by Robert R 13 Apr '10

13 Apr '10

Hi, My Kamailio server uses an external mysql database with perl script to proxy traffic. When mysql server is unreachable, Kamailio do not respond with 100 Trying and instead responses with 484 (Address Incomplete) to the INVITE message. Is this a normal behavior? I need to response with 503 when mysql is unreachable. Can I do it in kamailio.cfg or I need to implement it in the perl script? Thanks, R

1 0

Compiling and installing sip-router with some SER modules
by Steven C. Blair 13 Apr '10

13 Apr '10

I'd like to build the latest version of sip-router from source but include server modules from the modules_s directory. I thought the command 'make include_modules="modules_s/permissions modules_s/pike modules_s/ratelimit modules_s/sms modules_s/sanity" ' would do the trick but it doesn't. Instead all modules from the modules_s are built. Is there a way to compile and install the core + modules_s/(permissions,pike,ratelimit,sms,sanity) ? Thanks --- ISC Networking & Telecommunications 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 215-573-8396 215-898-9348 (fax)

3 2

[OT] Nagios SIP Plugin
by Iñaki Baz Castillo 13 Apr '10

13 Apr '10

Hi, I've coded a SIP plugin for Nagios. There is some other similar plugin but it's less customizable and less powerful. Homepage: http://dev.sipdoc.net/projects/sip-stuff/wiki/NagiosSIPplugin Features: - SIP UDP and TCP transport protocols. - Customizable parameters (From URI, Request URI, server address, local UDP port, timeout internal). - The replied status code can be matched against an expected value to be considered as valid. Regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

Calling number Routing possibility in 1.5rel.
by Maciej Bylica 13 Apr '10

13 Apr '10

Dear ALL, I am playing around with kamailio 1.5. Could somebody explain me what module best fit to route the call on the basis of number A (calling number). What is more all rules should be able to be modified directly from mysql db. I've run through 3.0 modules and figured out that there is drouting module which is really great, but is it present in 1.5 rel? If negative, what other modules can be used. Thx in advance, Maciej.

2 7

[Serusers] Problems Using iptel accounts
by Nahum Nir 12 Apr '10

12 Apr '10

Hi all, I have made two accounts at iptel: nahum1 and nahum2. I have two computers running X-Lite, both of them registered succesfully. But when I try to invite I get 404 Not Found. I'm dialing nahum2(a)iptel.org. What an I doing worng? Thanks, Nahum

2 1

My wish
by Alex Balashov 12 Apr '10

12 Apr '10

Allow cfg_get parameters to be used as modparams, too, so I can define one set of global database properties and pass them as db_urls to every module that uses one and also to 'sqlops' as an 'sqlcon', and similar things. -- 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/

3 16

Close TLS connection and Registrar Server (Kamailio V1.4.x)
by edson.gomes.leme 11 Apr '10

11 Apr '10

Hello, We are using Kamailio V1.4 with TCP/TLS. The Kamailio is configured the TCP lifetime to the expire value of the REGISTER by using the tcp_persistent_flag parameter of Kamailio's registrar module. Note: The SIP proxy should not close the TCP connection as it is not capable of opening a new one. Kamailio tries to open a TLS connection to the client. The Kamailio should use the existing TLS/TCP connection that was established by the client during registration. We are applying NAT traversal techniques to instruct Kamailio to reuse the established TCP/TLS connection. Example: for each register call fix_nated_register(). For every other request and response call fix_nated_contact(). I have got one SIP phone registered via TLS to kamailio-1.4.X version with Expires = 3600. phone1: connected via ip1:port1 and Expires = 3600. Example: After 15 minutes TLS connection is closed. The SIP Phone opens a new TLS connection and registers via TLS to Kamailio. Thus, we have the following records on the Registrar Server (Kamailio): phone1: connected via ip1:port1 and Expires = 2700 [3600 – (15 min * 60 s) = 2700]. phone1: connected via ip1:port2 and Expires = 3600. Is this the desirable behavior? Shouldn’t the Kamailio (Registrar Server) delete the below registry? phone1: connected via ip1:port1 and Expires = 2700 [3600 – (15 min * 60 s) = 2700]. Best regards! Edson G. Leme

2 1

Close TLS connection and Registrar Server (Kamailio V1.4.x)
by edson.gomes.leme 11 Apr '10

11 Apr '10

1 0

users@kamailio and sr-users@sip-router merged
by Daniel-Constantin Mierla 10 Apr '10

10 Apr '10

Hello, the process of merging users(a)list.kamailio.org and sr-users(a)lists.sip-router.org is completed, offering a single place for community discussions via email related to stable versions and generic topics, also removing confusion for new comers which to use for best feedback. Because sr-users was a rather new mailing list, I imported the archive from users. It will help to continue the topics and have the entire knowledge base built over years in the same place, for web searches. The old Users mailing list archive still exists separately, link at: http://lists.kamailio.org/pipermail/users/ No new posts will get there, subscription is disabled, messages to users list are directed to sr-users (therefore replying to existing threads on either of the mailing lists work just fine). No matter you were a Users or SR-Users subscriber, updates to your profile can be done at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users I case your password is no longer valid, you can retrieve it via password reminder at: http://lists.sip-router.org/cgi-bin/mailman/options/sr-users Thanks, Daniel PS. Some of you may have received an unintentional "unsubscribed from sr-users-new mailing list" message, just ignore it - happened when I simulated synchronization of members on temporary name sr-users-new, no unsubscribe action was actually done -- sorry for inconvenience. -- Daniel-Constantin Mierla

1 0

[Kamailio-Users] merging users mailing list
by Daniel-Constantin Mierla 10 Apr '10

10 Apr '10

Hello, I am starting the merging process of users mailing list. Please ignore any notifications you get, if you get them is by mistake in my config -- none of subscribed users will be affected, the work is done for testing on a temporary new list, therefore everyone is safe. Sorry for any inconvenience! I will announce when it is finished. Daniel

1 0

[Kamailio-Users] dispatcher has problems
by Vic Jolin 09 Apr '10

09 Apr '10

I just noticed that if I have 4 servers set like this # group sip addresses of your * units 1 sip:XXX.XXX.XXX.XXX:5060 1 sip:XXX.XXX.XXX.XXX:5060 1 sip:XXX.XXX.XXX.XXX:5060 1 sip:XXX.XXX.XXX.XXX:5060 the registration does not work and I get 401 unauthorized. My setup is like this Openser as proxy sends all SIP messages to load balanced asterisk servers So sip phones should be able to register to the asterisk server The routing part in kamailio.cfg looks like this if ( method=="REGISTER" || method=="NOTIFY" || method=="OPTIONS" || method=="ACK" || method=="MESSAGE") { fix_nated_contact(); fix_nated_register(); ds_select_dst("1","4"); forward();#uri:host, uri:port); exit(); } if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } else if (method=="INVITE"){ log("VCTOR: Got an invite\n"); fix_nated_contact(); force_rtp_proxy(); ds_select_dst("1","4"); forward();#uri:host, uri:port); t_on_failure("1"); }; search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); if (!is_method("INVITE|REGISTER|MESSAGE|BYE|NOTIFY|ACK|CANCEL|REFER|SUBSCRIBE|OPTIONS|INFO")) { xlog("L_INFO", "=== [$rm] Method Not Implement ===========\n"); sl_send_reply("501", "Not implemented here"); return; }; any thoughts?

2 2

[Kamailio-Users] lcr module creating extra branch
by Jon Bonilla 09 Apr '10

09 Apr '10

Hi I am using Kamailio 3.0.1 and experiencing a strange behaviour of the lcr module. In short: Using next_gw() function makes my server to send two INVITEs to different destinations! The INVITE Kamailio receives is like: R=sip:4644235465675757@77.77.77.77 F=sip:manwe2@77.77.77.77 T=sip:4644235465675757@77.77.77.77 To try this I have set this in my gw table: sercmd lcr.dump_gws { lcr_id: 1 grp_id: 17 ip_addr: 91.121.117.27 hostname: sindominio.net port: 5080 scheme: sip transport: UDP strip: 0 tag: weight: 1 flags: 37 defunct_until: 268313248 } sercmd lcr.dump_lcrs { lcr_id: 1 prefix: from_uri: grp_id: 17 priority: 1 } Hostname and ip_addr don't match (sindominio.net points to 82.144.4.26). What I capture leaving Kamailio is: U 2010/04/08 17:51:09.614055 77.77.77.77:5060 -> 91.121.117.27:5080 INVITE sip:+4644235465675757@77.77.77.77 SIP/2.0' Record-Route: <sip:77.77.77.77;lr=on;ftag=hgtcn;nat=yes;vsf=czFwOgs5JCAEJEYoJGk3ajxNYmBrPBYC>' Via: SIP/2.0/UDP 77.77.77.77;branch=z9hG4bK90e8.cd3135e1.0' Via: SIP/2.0/UDP 10.0.0.92;received=91.115.172.34;rport=53658;branch=z9hG4bKiwibmejv' Max-Forwards: 69' To: <sip:4644235465675757@77.77.77.77>' From: "1102" <sip:1102@77.77.77.77>;tag=hgtcn' Call-ID: qxvyddwlnmurqvi@multivac' CSeq: 571 INVITE' Contact: <sip:manwe2@91.115.172.34:53658>' Content-Type: application/sdp' Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE' Supported: replaces,norefersub,100rel' User-Agent: Twinkle/1.4.2' Content-Length: 226' ' v=0' o=twinkle 1296734594 70729915 IN IP4 77.77.77.77' s=-' c=IN IP4 77.77.77.77' t=0 0' m=audio 38232 RTP/AVP 8 101' a=rtpmap:8 PCMA/8000' a=rtpmap:101 telephone-event/8000' a=fmtp:101 0-15' a=ptime:20' a=nortpproxy:yes' # U 2010/04/08 17:51:09.614074 77.77.77.77:5060 -> 82.144.4.26:5060 INVITE sip:+4644235465675757@sindominio.net SIP/2.0' Record-Route: <sip:77.77.77.77;lr=on;ftag=hgtcn;nat=yes;vsf=czFwOgs5JCAEJEYoJGk3ajxNYmBrPBYC>' Via: SIP/2.0/UDP 77.77.77.77;branch=z9hG4bK90e8.cd3135e1.1' Via: SIP/2.0/UDP 10.0.0.92;received=91.115.172.34;rport=53658;branch=z9hG4bKiwibmejv' Max-Forwards: 69' To: <sip:4644235465675757@77.77.77.77>' From: "1102" <sip:1102@77.77.77.77>;tag=hgtcn' Call-ID: qxvyddwlnmurqvi@multivac' CSeq: 571 INVITE' Contact: <sip:manwe2@91.115.172.34:53658>' Content-Type: application/sdp' Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE' Supported: replaces,norefersub,100rel' User-Agent: Twinkle/1.4.2' Content-Length: 226' ' v=0' o=twinkle 1296734594 70729915 IN IP4 77.77.77.77' s=-' c=IN IP4 77.77.77.77' t=0 0' m=audio 38232 RTP/AVP 8 101' a=rtpmap:8 PCMA/8000' a=rtpmap:101 telephone-event/8000' a=fmtp:101 0-15' a=ptime:20' a=nortpproxy:yes' My kamailio.cfg script does something like this: if(!load_gws("1", "$ru")) ... if(!next_gw()) ... t_on_branch("BRANCH_ROUTE_CLI_RTP"); if(!t_relay_to("0x01")) ... Branch route_cli_rtp: do rtpproxy stuff No append_branch is executed at all. I have two INVITES IMHO because next_gw() function does create the second one. Any idea about this behaviour?

3 10

[Kamailio-Users] sr-docs mailing list
by Daniel-Constantin Mierla 09 Apr '10

09 Apr '10

Hello, I renamed kamailio-docs(a)lists.kamailio.org to sr-docs(a)lists.sip-router.org. Info about it at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-docs Lately, several discussions touched the documentation aspects, in a need for a better structuring and updates for the new context of sip router project. The old mailing list was active for a while, but with the start of sip router project was ignored because of more focus on integration work. I hope some of you will join, help to build new documentation and spot mistakes in existing versions. Volunteers that want to lead various documentation efforts are more than welcome! Just say what you need... Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

[Kamailio-Users] Kamailio Blocking
by dotnetdub 09 Apr '10

09 Apr '10

Hello List, I have been running a very stable Kamailio 1.4 install for over a year now with no downtime. From time to time I get a message from the OS telling me that task kamailio:xxxx blocked for more than 120 seconds and a dump into syslog. I recently setup Kamailio 3.01 and it's been running for around 3 weeks now. Purely testing at the moment, there is only 1 SIP registration and almost no activity most of the time. I was checking dmesg and: [269844.208011] INFO: task kamailio:28504 blocked for more than 120 seconds. [269844.208093] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [269844.208178] kamailio D f6d77bf8 0 28504 28484 [269844.208180] f771e140 00000082 00000002 f6d77bf8 f7091390 f771e2cc c4019020 00000000 [269844.208184] 00000000 011615c9 f7091398 00000282 00000000 00000000 00000000 ffffffff [269844.208188] f5cba384 f5cba38c f5cba388 f771e140 c02c91ec f6969d68 f5cba38c f771e140 [269844.208191] Call Trace: [269844.208203] [<c02c91ec>] __mutex_lock_slowpath+0x50/0x7b [269844.208208] [<c02c9082>] mutex_lock+0xa/0xb [269844.208210] [<c02b7d4d>] unix_dgram_recvmsg+0x3e/0x231 [269844.208214] [<c0161beb>] get_page_from_freelist+0xc1/0x3e9 [269844.208218] [<c0161045>] __rmqueue_smallest+0x83/0xe3 [269844.208221] [<c025f771>] sock_recvmsg+0xde/0xf9 [269844.208226] [<c01387b0>] autoremove_wake_function+0x0/0x2d [269844.208231] [<c01620e9>] __alloc_pages_internal+0xb5/0x34e [269844.208236] [<c0260595>] sys_recvfrom+0xb4/0x116 [269844.208239] [<c011af13>] do_page_fault+0x0/0x8f9 [269844.208242] [<c0108540>] do_notify_resume+0x5f8/0x649 [269844.208247] [<c011b3c5>] do_page_fault+0x4b2/0x8f9 [269844.208250] [<c0260a48>] sys_socketcall+0x135/0x19e [269844.208254] [<c0108853>] sysenter_past_esp+0x78/0xb1 [269844.208257] [<c02c0000>] xenfb_probe+0xd1/0x35b [269844.208261] ======================= [269844.208262] INFO: task kamailio:28506 blocked for more than 120 seconds. [269844.208314] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [269844.208395] kamailio D d0e00947 0 28506 28484 [269844.208397] f77f8e60 00000082 00000002 d0e00947 00004292 f77f8fec c4024020 00000001 [269844.208401] 00000000 00000001 00000002 00000000 f7091398 00000001 00000001 c01211f2 [269844.208404] f5cba384 f5cba38c f5cba388 f77f8e60 c02c91ec f5cba38c f60add68 f77f8e60 [269844.208408] Call Trace: [269844.208410] [<c01211f2>] __wake_up_sync+0x2a/0x3e [269844.208413] [<c02c91ec>] __mutex_lock_slowpath+0x50/0x7b [269844.208416] [<c02c9082>] mutex_lock+0xa/0xb [269844.208418] [<c02b7d4d>] unix_dgram_recvmsg+0x3e/0x231 [269844.208421] [<c0161beb>] get_page_from_freelist+0xc1/0x3e9 [269844.208424] [<c0161045>] __rmqueue_smallest+0x83/0xe3 [269844.208427] [<c025f771>] sock_recvmsg+0xde/0xf9 [269844.208431] [<c01387b0>] autoremove_wake_function+0x0/0x2d [269844.208435] [<c01620e9>] __alloc_pages_internal+0xb5/0x34e [269844.208440] [<c0260595>] sys_recvfrom+0xb4/0x116 [269844.208444] [<c0180d0d>] cp_new_stat64+0xfc/0x10e [269844.208449] [<c011b3c5>] do_page_fault+0x4b2/0x8f9 [269844.208452] [<c0260a48>] sys_socketcall+0x135/0x19e [269844.208456] [<c0108853>] sysenter_past_esp+0x78/0xb1 [269844.208459] [<c02c0000>] xenfb_probe+0xd1/0x35b [269844.208462] ======================= [274198.229285] INFO: task kamailio:28504 blocked for more than 120 seconds. [274198.229340] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [274198.229422] kamailio D f6d77bf8 0 28504 28484 [274198.229425] f771e140 00000082 00000002 f6d77bf8 f7091390 f771e2cc c4019020 00000000 [274198.229429] 00000000 011615c9 f7091398 00000282 00000000 00000000 00000000 ffffffff [274198.229432] f5cba384 f5cba38c f5cba388 f771e140 c02c91ec f6969d68 f5cba38c f771e140 [274198.229436] Call Trace: [274198.229447] [<c02c91ec>] __mutex_lock_slowpath+0x50/0x7b [274198.229453] [<c02c9082>] mutex_lock+0xa/0xb [274198.229455] [<c02b7d4d>] unix_dgram_recvmsg+0x3e/0x231 [274198.229459] [<c0161beb>] get_page_from_freelist+0xc1/0x3e9 [274198.229463] [<c0161045>] __rmqueue_smallest+0x83/0xe3 [274198.229466] [<c025f771>] sock_recvmsg+0xde/0xf9 [274198.229471] [<c01387b0>] autoremove_wake_function+0x0/0x2d [274198.229476] [<c01620e9>] __alloc_pages_internal+0xb5/0x34e [274198.229481] [<c0260595>] sys_recvfrom+0xb4/0x116 [274198.229484] [<c011af13>] do_page_fault+0x0/0x8f9 [274198.229487] [<c0108540>] do_notify_resume+0x5f8/0x649 [274198.229492] [<c011b3c5>] do_page_fault+0x4b2/0x8f9 [274198.229495] [<c0260a48>] sys_socketcall+0x135/0x19e [274198.229499] [<c0108853>] sysenter_past_esp+0x78/0xb1 [274198.229502] [<c02c0000>] xenfb_probe+0xd1/0x35b [274198.229506] ======================= [274198.229507] INFO: task kamailio:28506 blocked for more than 120 seconds. [274198.229559] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [274198.229641] kamailio D d0e00947 0 28506 28484 [274198.229643] f77f8e60 00000082 00000002 d0e00947 00004292 f77f8fec c4024020 00000001 [274198.229646] 00000000 00000001 00000002 00000000 f7091398 00000001 00000001 c01211f2 [274198.229650] f5cba384 f5cba38c f5cba388 f77f8e60 c02c91ec f5cba38c f60add68 f77f8e60 [274198.229653] Call Trace: [274198.229656] [<c01211f2>] __wake_up_sync+0x2a/0x3e [274198.229659] [<c02c91ec>] __mutex_lock_slowpath+0x50/0x7b [274198.229662] [<c02c9082>] mutex_lock+0xa/0xb [274198.229664] [<c02b7d4d>] unix_dgram_recvmsg+0x3e/0x231 [274198.229667] [<c0161beb>] get_page_from_freelist+0xc1/0x3e9 [274198.229670] [<c0161045>] __rmqueue_smallest+0x83/0xe3 [274198.229673] [<c025f771>] sock_recvmsg+0xde/0xf9 [274198.229677] [<c01387b0>] autoremove_wake_function+0x0/0x2d [274198.229681] [<c01620e9>] __alloc_pages_internal+0xb5/0x34e [274198.229686] [<c0260595>] sys_recvfrom+0xb4/0x116 [274198.229690] [<c0180d0d>] cp_new_stat64+0xfc/0x10e [274198.229696] [<c011b3c5>] do_page_fault+0x4b2/0x8f9 [274198.229699] [<c0260a48>] sys_socketcall+0x135/0x19e [274198.229703] [<c0108853>] sysenter_past_esp+0x78/0xb1 [274198.229706] [<c02c0000>] xenfb_probe+0xd1/0x35b There was no activity when this happened. ON the 1.4 box there could be about 10 sessions setup when it happens. kamctl ps: Process:: ID=14 PID=28504 Type=MI DATAGRAM Process:: ID=15 PID=28505 Type=MI DATAGRAM Process:: ID=16 PID=28506 Type=MI DATAGRAM Even on the old install this doesn't seem to cause any problem and same here on 3.01 but would like to try and solve it. Any idea? Regards, Stephen

4 5

[Kamailio-Users] fr_inv_timer in kam-3.0
by Andreas Granig 09 Apr '10

09 Apr '10

Hi, In the docs at http://kamailio.org/docs/modules/3.0.x/modules/tm.html#fr_inv_timer it says that fr_inv_timer_avp should be defined like this, without $ or $avp: modparam("tm", "fr_inv_timer_avp", "my_fr_inv_timer") In kam <= 1.5 I did it like that: modparam("tm", "fr_inv_timer_avp", "$avp(s:callee_fr_inv_timer)") which doesn't give me an error on startup with kam-3.0 either, but the timer doesn't get fired (I use seconds for that as noted in the docs). If I change it to modparam("tm", "fr_inv_timer_avp", "callee_fr_inv_timer") then I get the error "malformed or non AVP callee_fr_inv_timer AVP definition", same with setting it to "s:callee_fr_inv_timer". Anyhow, it seems to be deprecated anyways, so I'm looking to get t_set_fr() working. I'm just curious how I can use a var or AVP loaded from DB to set the value on-the-fly? t_set_fr(...) seems to allow only constants to be set. Couldn't find anything in the docs regarding that one. Thanks, Andreas

5 14

[Kamailio-Users] drop in branch/onsend-route in kam-3.0
by Andreas Granig 08 Apr '10

08 Apr '10

Hi, Looks like behavior of drop has changed in Kamailio 3.0. Before, I was able to drop() a specific branch in branch_route depending on various checks. http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#drop still says this is possible, http://kamailio.org/docs/modules/3.0.x/modules/tm.html#t_on_branch says it's not possible. So calling drop() in branch_route has no effect, and I have no idea how to mark a branch in branch_route to be dropped in onsend_route, because checking a bflag doesn't work there, and neither does checking $rU. Any ideas on how to accomplish dropping a specific branch? Andreas

5 7

[Kamailio-Users] Understanding memory-leaks by inspecting PKG/SHM memory status at shutdown
by Iñaki Baz Castillo 08 Apr '10

08 Apr '10

Hi, according to the wiki page: http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory In order to detect a possible memory leak (in PGK or SHM) a good way is enabling memory debugging and shuting down the server, so: "the memory manager will dump the memory status. Normally most of the memory is cleaned during shutdown. If there is memory leak, it should be visible as not-freed memory." But I would like to know how exactly to detect if there is a memoty leak by inspecting the SHM/PKG status, something like this: ----------------------------------------------------------------------------------------------------------------------------- [32715]: Memory status (pkg): [32715]: fm_status (0x701a40): [32715]: heap size= 16777216 [32715]: used= 189696, used+overhead=247392, free=16529824 [32715]: max used (+overhead)= 247392 [32715]: dumping free list: [32715]: hash = 1 fragments no.: 1, unused: 0#012#011#011 bucket size: 8 - 8 (first 8) [32715]: hash = 2058 fragments no.: 1, unused: 0#012#011#011 bucket size: 8388608 - 16777216 (first 16530136) [32715]: TOTAL: 2 free fragments = 16530144 free bytes [32678]: Memory status (shm): [32678]: fm_status (0x7fa57921c000): [32678]: heap size= 67108864 [32678]: used= 16, used+overhead=35872, free=67072992 [32678]: max used (+overhead)= 2821064 [32678]: dumping free list: [32678]: hash = 1 fragments no.: 45, unused: 0#012#011#011 bucket size: 8 - 8 (first 8) [32678]: hash = 2 fragments no.: 4, unused: 0#012#011#011 bucket size: 16 - 16 (first 16) [32678]: hash = 3 fragments no.: 16, unused: 0#012#011#011 bucket size: 24 - 24 (first 24) [32678]: hash = 4 fragments no.: 2, unused: 0#012#011#011 bucket size: 32 - 32 (first 32) [32678]: hash = 5 fragments no.: 1, unused: 0#012#011#011 bucket size: 40 - 40 (first 40) [32678]: hash = 6 fragments no.: 1, unused: 0#012#011#011 bucket size: 48 - 48 (first 48) [32678]: hash = 7 fragments no.: 39, unused: 0#012#011#011 bucket size: 56 - 56 (first 56) [32678]: hash = 8 fragments no.: 1, unused: 0#012#011#011 bucket size: 64 - 64 (first 64) [32678]: hash = 9 fragments no.: 1, unused: 0#012#011#011 bucket size: 72 - 72 (first 72) [32678]: hash = 10 fragments no.: 2, unused: 0#012#011#011 bucket size: 80 - 80 (first 80) [32678]: hash = 15 fragments no.: 1, unused: 0#012#011#011 bucket size: 120 - 120 (first 120) [32678]: hash = 25 fragments no.: 1, unused: 0#012#011#011 bucket size: 200 - 200 (first 200) [32678]: hash = 40 fragments no.: 5, unused: 0#012#011#011 bucket size: 320 - 320 (first 320) [32678]: hash = 112 fragments no.: 1, unused: 0#012#011#011 bucket size: 896 - 896 (first 896) [32678]: hash = 128 fragments no.: 4, unused: 0#012#011#011 bucket size: 1024 - 1024 (first 1024) [32678]: hash = 129 fragments no.: 2, unused: 0#012#011#011 bucket size: 1032 - 1032 (first 1032) [32678]: hash = 224 fragments no.: 1, unused: 0#012#011#011 bucket size: 1792 - 1792 (first 1792) [32678]: hash = 258 fragments no.: 3, unused: 0#012#011#011 bucket size: 2064 - 2064 (first 2064) [32678]: hash = 473 fragments no.: 1, unused: 0#012#011#011 bucket size: 3784 - 3784 (first 3784) [32678]: hash = 1026 fragments no.: 1, unused: 0#012#011#011 bucket size: 8208 - 8208 (first 8208) [32678]: hash = 2049 fragments no.: 2, unused: 0#012#011#011 bucket size: 16384 - 32768 (first 16512) [32678]: hash = 2051 fragments no.: 1, unused: 0#012#011#011 bucket size: 65536 - 131072 (first 98336) [32678]: hash = 2056 fragments no.: 1, unused: 0#012#011#011 bucket size: 2097152 - 4194304 (first 2621440) [32678]: hash = 2060 fragments no.: 1, unused: 0#012#011#011 bucket size: 33554432 - 67108864 (first 64287800) [32678]: TOTAL: 137 free fragments = 67072992 free bytes ----------------------------------------------------------------------------------------------------------------------------- This is, how should look this SHM/PKG status example (after shutdown) in case there is a memory leak? how to detect not-freed memory? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 5

[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 08 Apr '10

08 Apr '10

Hello, I know some of devels are still in Easter vacation, but I hope we can get together soon on irc to sketch the plan for 3.1 release. I propose next week, Wednesday, April 14, 15:00UTC, on #sip-router channel hosted by irc.freenode.net If you are a developer and want to participate, please announce. If it is not possible to attend for major contributors, we can look for another date (eventually start a pool to see the day were most of us are available). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

4 4

[Serusers] ReTransmission of CANCELs on 0.9.7
by rupert.organ＠bt.com 08 Apr '10

08 Apr '10

Hi All, Has anyone come up against an issue on 0.9.6 / 0.9.7 whereby a SIP App Server, delivers a CANCEL to the SER. The SER proxies the CANCELs to all the forked endpoints. At the same time it sends "200 Cancelling" back to the SIP App server...(this suppresses CANCEL retransmission from the SIP App server). If one of the CANCELs from the SER to the SIP UA gets lost....the phone keeps ringing. There is no CANCEL retransmission on the SER !!! There is CANCEL retransmission on the SIP App, but when it sends SER replies "no pending branches" Ideally want the SER do CANCEL retransmissions. I understand the timers were rewritten in 0.10.x but this is not hardened version like 0.9.6....any suggestions? Thanks in advance Rupert

4 3

[Kamailio-Users] merging users mailing lists
by Daniel-Constantin Mierla 08 Apr '10

08 Apr '10

Hello, I propose to merge the users mailing lists, most of the traffic these days is about 3.0 and even there are 2 stables branches now, they are sync'ed, so same code more or less. For 3.1 will be one stable branch, falvor selection will be only a matter of make command. Lately common useful topics are discussed on those different mailing lists, notifications and knowledge base building require cross-posting, lot of overhead imo. Any other opinion? Like with devel mailing lists, existing email addresses for users ML can still be used, just that end on same ML. Natural choice will be to have users@kamailio and serusers@iptel to be directed to sr-users(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

6 8

[Kamailio-Users] mc (Midnight Commander) syntax highlighting scheme
by Константин 07 Apr '10

07 Apr '10

Hi all! For easy editing kamailio configuration files I developed a syntax highlighting scheme for mc. Preview The scheme includes basic syntactic structures of Kamailio 1.5, all constants, variables, operators and standart core functions. To install it, download the file kamailio.syntax from attachment and copy it to /usr/share/mc/syntax (for Debian Etch) Next, add to the file /usr/share/mc/syntax/Syntax including: file ..\*\\.cfg$ Kamailio\sconfig include kamailio.syntax Thats all, I hope its helps someone ;) -- Shpinev Konstantin mailto:kks_mail@inbox.ru

3 2

[Kamailio-Users] Version 3
by dotnetdub 07 Apr '10

07 Apr '10

Hi List, Anybody using this in production yet? If so what kind of volume and how is reliability? Looking to move to this platform, looks very good, interested to hear some experiences. Thanks, Stephen

5 6

Re: [Kamailio-Users] Odd Behaviour with Kamailio + TLS
by Hemanshu Patel 02 Apr '10

02 Apr '10

Yes, i had installed self signed certificates via IE. Bria shows it and validates them during registration process, its the call initiation where it creates problem. Kamailio starts without fix_nated_register. With this function it gives configuration file error. and i had load nathelper.so and that is been confirmed via logs -- Regards, Hemanshu Patel SIS,Ahmedabad. M: 09601295238 > Just checking... > > Some time ago, I made some tests with Bria and self signed certs... It was a little tricky to find that I had to install the cert through IE. Did you install them this way? Can you check Bria's log to ensure that the cert is accepted? > > Edson. > > Henning Westerholt escreveu: >> On Thursday 01 April 2010, Hemanshu Patel wrote: >>> yes i did, and log shows that nathelper.so load without any problem. kamailio loads and work perfectly with grandstream devices with TLS and >>> SRTp. it gives problem with Bria softphones only. >>> I am using self signed certificates which works perfectly with grandstream, can there be any chances that due to self sign ness of the >>> certificates, it doesnt work well with bria devices. >> Hi Hemanshu, >> ok, so the error you reported in your previous message is solved. It could be >> possible that the Bria softphone implement a more strict policy with regards >> to the check of the certificates. But you should be able to find something in >> the UA documentation about this, you could ask the vendor or other users. As >> the last resort just test with certificates that are not self signed. ;-) >> Cheers, >> Henning >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >

2 2

[Kamailio-Users] Odd Behaviour with Kamailio + TLS
by Hemanshu Patel 01 Apr '10

01 Apr '10

Hi all, I am not sure it is strange behavior or its the desirable one. i have got two phones registered via TLS to kamailio-1.5.X version. phone1: connected via ip1:port1 and having contact value ip1:port2 phone2: connected via ip2:port3 and having contact value ip2:port4 when i have just this two phones i can see 2 TCP connected to my kamailio from both phones from ip1:port1 and ip2:port3. now when i call from phone1 to phone2, rather then using the already established and TLS authenticated connection kamailio creates two new connection from kamailio to ip1:port2 and ip2:port4. mean kamailio creates new connection to the contact value stored in location table of openser DB. Is this the desirable behavior? shouldn't kamailio use the alreay established connection? -- Regards, Hemanshu Patel SIS,Ahmedabad. M: 09601295238

4 6

SEMS 1.2.0 released
by Stefan Sayer 01 Apr '10

01 Apr '10

Hello, it is my pleasure to announce that SEMS 1.2.0 has been released. This release again brings many improvements to this fine, free SIP Media and Application server. Get the source at http://ftp.iptel.org/pub/sems/sems-1.2.0.tar.gz Some debian packages at http://ftp.iptel.org/pub/sems/1.2/1.2.0/packages/ Changelog: http://ftp.iptel.org/pub/sems/doc/current/changelog.html Documentation got some new getting started tutorials: http://ftp.iptel.org/pub/sems/doc/current/index.html And there is now an experimental lenny32 vmware image available, configured as webconference server (but usable for all applications with minimal reconfiguration): https://developer.berlios.de/project/shownotes.php?release_id=17393 For more information see the home page at http://iptel.org/sems . Among the most interesting news I would like to emphasize that its powerful state machine scripting engine DSM has matured a lot between 1.1 and 1.2 versions. Thanks to its growing number of modules (including MySQL DB, conferencing, Python and AWS), live script reloading and monitoring, its interfacing to external applications, and its extended script language it has become a good service development and deployment platform. Other interesting additions is a CMake based build system, an in-memory DB 'monitoring' for administrative and applications use, a twitter app, and numerous fixes and improvements all around. Respect and many Thanks go to all contributors and sponsors of the SEMS project. With Best Regards Stefan Sayer P.S. Apologies if you have received multiple copies of this message. -- Stefan Sayer VoIP Services Consulting and Development Warschauer Str. 24 10243 Berlin tel:+491621366449 sip:sayer@iptel.org email/xmpp:stefan.sayer@gmail.com

1 0

[Serusers] Registration process : 401 then 403, is it possible ?
by inge 01 Apr '10

01 Apr '10

Hi, I'm looking for a workarround to avoid a buggy CPE. The CPE doesn't stop to try to register after he got a 401 Unauthorized. So this produce a flooding. That's why I'm looking for something to send a first 401 and then a 403 if the registration keep failing. Thanks in advance for your idea. Regards, Adrien

1 0

Re: [Kamailio-Users] Users Digest, Vol 58, Issue 76
by Rothe, Marcus, D22-WHV 31 Mar '10

31 Mar '10

"users-request(a)lists.kamailio.org" <users-request(a)lists.kamailio.org> schrieb: Send Users mailing list submissions to users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.kamailio.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request(a)lists.kamailio.org You can reach the person managing the list at users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: Odd Behaviour with Kamailio + TLS (Klaus Darilion) 2. Re: Registrar: Save modified contact / set the q value (Daniel-Constantin Mierla) 3. Re: more type conversion wonders (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Wed, 31 Mar 2010 11:34:45 +0200 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> Subject: Re: [Kamailio-Users] Odd Behaviour with Kamailio + TLS To: Hemanshu Patel <hemanshu.patel(a)saicare.com> Cc: users(a)lists.kamailio.org Message-ID: <4BB31735.6030500(a)pernau.at> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Am 31.03.2010 08:56, schrieb Hemanshu Patel: > > Hi all, > > I am not sure it is strange behavior or its the desirable one. > > i have got two phones registered via TLS to kamailio-1.5.X version. > > phone1: connected via ip1:port1 and having contact value ip1:port2 > phone2: connected via ip2:port3 and having contact value ip2:port4 > > when i have just this two phones i can see 2 TCP connected to my kamailio > from both phones from ip1:port1 and ip2:port3. > > now when i call from phone1 to phone2, rather then using the already > established and TLS authenticated connection kamailio creates two new > connection from kamailio to ip1:port2 and ip2:port4. > mean kamailio creates new connection to the contact value stored in > location table of openser DB. > > Is this the desirable behavior? shouldn't kamailio use the alreay > established connection? This is the standard Kamailio behavior. Thus, you should apply NAT traversal techniques to instruct Kamailio to reuse the established connection. In short: for each register call fix_nated_register(). For every other request and response call fix_nated_contact(). Thus, if you use the default configuration and apply NAT traversal it should work. regards klaus ------------------------------ Message: 2 Date: Wed, 31 Mar 2010 11:37:27 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [Kamailio-Users] Registrar: Save modified contact / set the q value To: Ovidiu Sas <osas(a)voipembedded.com> Cc: NeoTel Lists <MailingLists(a)neotel.at>, users(a)lists.kamailio.org Message-ID: <4BB317D7.4030102(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 3/30/10 8:29 PM, Ovidiu Sas wrote: > If you need this in 1.5, try to fix the contact header and then > forward the REGISTER back to yourself. > The loopback REGISTER should have the fixed Contact header and now you > can call save(). > You will need to tweak a a little bit the config to get everything > right into the usrloc (the loopback REGISTER will come from the server > IP) and it might work. > it works for no nat scenario, otherwise make sure you will get the looped message on the same socket and you add before looping a header with source ip and port (corresponding nat box pinhole) that you store in received avp. Cheers, Daniel > Regards, > Ovidiu Sas > > On Tue, Mar 30, 2010 at 11:36 AM, NeoTel Lists<MailingLists(a)neotel.at> wrote: > >> Hello Everybody! >> >> Is there any way to save(domain, 0x02) the contact after it has been changed >> somehow in Kamailio<= 1.5? >> Do I have to upgrade to 3.0 and use msg_apply_changes() before save()? >> >> Or: Can I set the to be saved q value somehow? >> >> br >> Walter >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ------------------------------ Message: 3 Date: Wed, 31 Mar 2010 11:41:29 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [Kamailio-Users] more type conversion wonders To: Juha Heinanen <jh(a)tutpro.com> Cc: users(a)lists.sip-router.org Message-ID: <4BB318C9.1080302(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 3/31/10 8:40 AM, Juha Heinanen wrote: > i played a bit more with selects and found that statement > > if ($sht(auth=>foo::count)> @cfg_get.local.gw_timeout) { > xlog("foo"); > } > I guess selects have types, while pseudo-variables are kind of type agnostic. Any PV has a string representation of the value, comparison is done as integer if both PVs are integers, otherwise is done as string. The safe side is to use PV representation of selects when comparing with another PV. PV and selects is another point for the future coherence. Cheers, Daniel > produces error > > 0(9213) :<core> [cfg.y:3379]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 507, column 28-55: bad expression: type mismatch: str instead of int at (507,55) > > but the error goes away if i either make explicit conversion > > if ($sht(auth=>foo::count)> (int)(a)cfg_get.local.gw_timeout) { > xlog("foo"); > } > > or use $sel > > if ($sht(auth=>foo::count)> $sel(cfg_get.local.gw_timeout)) { > xlog("foo"); > } > > this is thus exactly opposite than in my t_set_fr ordeal where i got > conversion error when i used $sel instead of @. > > in my opinion, no (int) conversion should be needed in the firs case, > because the value of the cfg variable is int. > > in summary, writing statements that include selects is very difficult > and error prone. > > -- juha > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ------------------------------ _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 58, Issue 76 *************************************

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Re: [Kamailio-Users] Users Digest, Vol 58, Issue 76
by Rothe, Marcus, D22-WHV 31 Mar '10

31 Mar '10

"users-request(a)lists.kamailio.org" <users-request(a)lists.kamailio.org> schrieb: Send Users mailing list submissions to users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.kamailio.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request(a)lists.kamailio.org You can reach the person managing the list at users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: Odd Behaviour with Kamailio + TLS (Klaus Darilion) 2. Re: Registrar: Save modified contact / set the q value (Daniel-Constantin Mierla) 3. Re: more type conversion wonders (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Wed, 31 Mar 2010 11:34:45 +0200 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> Subject: Re: [Kamailio-Users] Odd Behaviour with Kamailio + TLS To: Hemanshu Patel <hemanshu.patel(a)saicare.com> Cc: users(a)lists.kamailio.org Message-ID: <4BB31735.6030500(a)pernau.at> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Am 31.03.2010 08:56, schrieb Hemanshu Patel: > > Hi all, > > I am not sure it is strange behavior or its the desirable one. > > i have got two phones registered via TLS to kamailio-1.5.X version. > > phone1: connected via ip1:port1 and having contact value ip1:port2 > phone2: connected via ip2:port3 and having contact value ip2:port4 > > when i have just this two phones i can see 2 TCP connected to my kamailio > from both phones from ip1:port1 and ip2:port3. > > now when i call from phone1 to phone2, rather then using the already > established and TLS authenticated connection kamailio creates two new > connection from kamailio to ip1:port2 and ip2:port4. > mean kamailio creates new connection to the contact value stored in > location table of openser DB. > > Is this the desirable behavior? shouldn't kamailio use the alreay > established connection? This is the standard Kamailio behavior. Thus, you should apply NAT traversal techniques to instruct Kamailio to reuse the established connection. In short: for each register call fix_nated_register(). For every other request and response call fix_nated_contact(). Thus, if you use the default configuration and apply NAT traversal it should work. regards klaus ------------------------------ Message: 2 Date: Wed, 31 Mar 2010 11:37:27 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [Kamailio-Users] Registrar: Save modified contact / set the q value To: Ovidiu Sas <osas(a)voipembedded.com> Cc: NeoTel Lists <MailingLists(a)neotel.at>, users(a)lists.kamailio.org Message-ID: <4BB317D7.4030102(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 3/30/10 8:29 PM, Ovidiu Sas wrote: > If you need this in 1.5, try to fix the contact header and then > forward the REGISTER back to yourself. > The loopback REGISTER should have the fixed Contact header and now you > can call save(). > You will need to tweak a a little bit the config to get everything > right into the usrloc (the loopback REGISTER will come from the server > IP) and it might work. > it works for no nat scenario, otherwise make sure you will get the looped message on the same socket and you add before looping a header with source ip and port (corresponding nat box pinhole) that you store in received avp. Cheers, Daniel > Regards, > Ovidiu Sas > > On Tue, Mar 30, 2010 at 11:36 AM, NeoTel Lists<MailingLists(a)neotel.at> wrote: > >> Hello Everybody! >> >> Is there any way to save(domain, 0x02) the contact after it has been changed >> somehow in Kamailio<= 1.5? >> Do I have to upgrade to 3.0 and use msg_apply_changes() before save()? >> >> Or: Can I set the to be saved q value somehow? >> >> br >> Walter >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ------------------------------ Message: 3 Date: Wed, 31 Mar 2010 11:41:29 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [Kamailio-Users] more type conversion wonders To: Juha Heinanen <jh(a)tutpro.com> Cc: users(a)lists.sip-router.org Message-ID: <4BB318C9.1080302(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 3/31/10 8:40 AM, Juha Heinanen wrote: > i played a bit more with selects and found that statement > > if ($sht(auth=>foo::count)> @cfg_get.local.gw_timeout) { > xlog("foo"); > } > I guess selects have types, while pseudo-variables are kind of type agnostic. Any PV has a string representation of the value, comparison is done as integer if both PVs are integers, otherwise is done as string. The safe side is to use PV representation of selects when comparing with another PV. PV and selects is another point for the future coherence. Cheers, Daniel > produces error > > 0(9213) :<core> [cfg.y:3379]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 507, column 28-55: bad expression: type mismatch: str instead of int at (507,55) > > but the error goes away if i either make explicit conversion > > if ($sht(auth=>foo::count)> (int)(a)cfg_get.local.gw_timeout) { > xlog("foo"); > } > > or use $sel > > if ($sht(auth=>foo::count)> $sel(cfg_get.local.gw_timeout)) { > xlog("foo"); > } > > this is thus exactly opposite than in my t_set_fr ordeal where i got > conversion error when i used $sel instead of @. > > in my opinion, no (int) conversion should be needed in the firs case, > because the value of the cfg variable is int. > > in summary, writing statements that include selects is very difficult > and error prone. > > -- juha > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ------------------------------ _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 58, Issue 76 *************************************

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[Kamailio-Users] more type conversion wonders
by jh＠tutpro.com 31 Mar '10

31 Mar '10

i played a bit more with selects and found that statement if ($sht(auth=>foo::count) > @cfg_get.local.gw_timeout) { xlog("foo"); } produces error 0(9213) : <core> [cfg.y:3379]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 507, column 28-55: bad expression: type mismatch: str instead of int at (507,55) but the error goes away if i either make explicit conversion if ($sht(auth=>foo::count) > (int)(a)cfg_get.local.gw_timeout) { xlog("foo"); } or use $sel if ($sht(auth=>foo::count) > $sel(cfg_get.local.gw_timeout)) { xlog("foo"); } this is thus exactly opposite than in my t_set_fr ordeal where i got conversion error when i used $sel instead of @. in my opinion, no (int) conversion should be needed in the firs case, because the value of the cfg variable is int. in summary, writing statements that include selects is very difficult and error prone. -- juha

2 1

[Kamailio-Users] Registrar: Save modified contact / set the q value
by NeoTel Lists 31 Mar '10

31 Mar '10

Hello Everybody! Is there any way to save(domain, 0x02) the contact after it has been changed somehow in Kamailio <= 1.5? Do I have to upgrade to 3.0 and use msg_apply_changes() before save()? Or: Can I set the to be saved q value somehow? br Walter

4 8

Re: [Kamailio-Users] Registrar: Save modified contact / set the qvalue
by NeoTel Lists 30 Mar '10

30 Mar '10

Again sorry, too much trial&error makes a lot of noise ... if (avp_subst("$(avp(contact))", "/;phone-context=q([01]\.[0-9]+)(.*>)?([^>]*)/\2\3;q=\1/i")) { # just to have it handy int that var(Q) $var(Q) = $(avp(contact){s.select,1,>}{param.value,q}); remove_hf("Contact"); append_hf("Contact: $(avp(contact))\r\n"); # must not add q= twice } -> Same result: q=-1 in Database. -----Ursprüngliche Nachricht----- Von: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] Im Auftrag von NeoTel Lists Gesendet: Dienstag, 30. März 2010 18:56 An: Alex Balashov; users(a)lists.kamailio.org Betreff: Re: [Kamailio-Users] Registrar: Save modified contact / set the qvalue Sorry for being unclear: I meant: Modify the contact in the script, then do save(). E.g. $(avp(contact)) = $ct; # Those Patton can't send the q value ... Misuse their ;phone-context param if (avp_subst("$(avp(contact))", "/;phone-context=q([01]\.[0-9])(.*>)?([^>]*)/\2\3;q=\1/i")) { # just to have it handy int that var(Q) $var(Q) = $(avp(contact){s.select,1,>}{param.value,q}); remove_hf("Contact"); append_hf("Contact: $(avp(contact));q=$var(Q)r\n"); # subst() does the same, also not save()ed } ... setbflag(1); fix_nated_register(); save("contactimpl_nat", "0x02"); Br Walter -----Ursprüngliche Nachricht----- Von: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] Im Auftrag von Alex Balashov Gesendet: Dienstag, 30. März 2010 17:58 An: users(a)lists.kamailio.org Betreff: Re: [Kamailio-Users] Registrar: Save modified contact / set the q value On 03/30/2010 11:36 AM, NeoTel Lists wrote: > Hello Everybody! > Is there any way to save(domain, 0x02) the contact after it has been > changed somehow in Kamailio <= 1.5? If it were not possible, how would nathelper:fix_nated_contact() work? -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/ _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

2 2

Re: [Kamailio-Users] Registrar: Save modified contact / set the q value
by NeoTel Lists 30 Mar '10

30 Mar '10

Sorry for being unclear: I meant: Modify the contact in the script, then do save(). E.g. $(avp(contact)) = $ct; # Those Patton can't send the q value ... Misuse their ;phone-context param if (avp_subst("$(avp(contact))", "/;phone-context=q([01]\.[0-9])(.*>)?([^>]*)/\2\3;q=\1/i")) { # just to have it handy int that var(Q) $var(Q) = $(avp(contact){s.select,1,>}{param.value,q}); remove_hf("Contact"); append_hf("Contact: $(avp(contact));q=$var(Q)r\n"); # subst() does the same, also not save()ed } ... setbflag(1); fix_nated_register(); save("contactimpl_nat", "0x02"); Br Walter -----Ursprüngliche Nachricht----- Von: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] Im Auftrag von Alex Balashov Gesendet: Dienstag, 30. März 2010 17:58 An: users(a)lists.kamailio.org Betreff: Re: [Kamailio-Users] Registrar: Save modified contact / set the q value On 03/30/2010 11:36 AM, NeoTel Lists wrote: > Hello Everybody! > Is there any way to save(domain, 0x02) the contact after it has been > changed somehow in Kamailio <= 1.5? If it were not possible, how would nathelper:fix_nated_contact() work? -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/ _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

1 0

[Kamailio-Users] Quick REGEX to match Private IP
by Uriel Rozenbaum 30 Mar '10

30 Mar '10

Hi guys, Does anyone have a REGEX syntax to match a private IP on the 192.168.x.x range? I'm trying with: if($avp(s:ip_origen)=~"192.168(\.([1]?\d{1,2}|2[0-4]{1}\d{1}|25[0-5]{1})){2}" ) But all IPs pass as private, even public ones. Thanks! Uriel

4 6

[Kamailio-Users] Fast lock loop with arm
by Zappasodi Daniele 30 Mar '10

30 Mar '10

Hi, I have a big problem with fast lock mutex on arm CPU: sometimes I find one or more children that go in an infinite loop, in the while loop of get_lock function. They remain in Run for long time (sometimes hours) and openser keeps 100% CPU. Now I have changed the functions get_lock and tsl in order to obtain more info and I observe that (if and) when the process finish the cycle, it have done a big amount of cycles. This the log with my added info: Mar 26 20:29:08 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle) Mar 26 20:29:51 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle) Mar 26 20:29:55 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle) Mar 26 20:29:55 SAM-IP openser[1645]: NOTICE:tm:t_retransmit_reply: MYTM lock [process in loop] Mar 26 20:33:46 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 59374917 (cycle) [after 4 minutes and 59374917 cycles, this is an example with a "short" loop] Mar 26 20:33:46 SAM-IP openser[1645]: NOTICE:tm:t_retransmit_reply: MYTM lock done I'm not able to recognize the call flow that generates the loop (I know only that it always happens when t_retransmit_reply calls LOCK_REPLIES), but I urgently need a work around to escape from the loop. This is the get_lock function with my counter j: inline static int get_lock(fl_lock_t* lock) { #ifdef ADAPTIVE_WAIT int i=ADAPTIVE_WAIT_LOOPS; int j=1; /***** my change ****/ #endif while(tsl(lock)){ #ifdef BUSY_WAIT #elif defined ADAPTIVE_WAIT j++; /***** my change ****/ if (i>0) i--; else sched_yield(); #else sched_yield(); #endif } return(j); } Can I break the lock when my counter reaches a big value? Should I call the unlock function after the break? which value can be considered too big? Thanks, Daniele ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

2 7

[Kamailio-Users] Kammailio as proxy and clustered asterisk as registrar
by Vic Jolin 30 Mar '10

30 Mar '10

Hi, I would like to seek help from the list regarding my setup. I have kamailio setup to send any sip messages to load balance asterisk clusters. The registration part is successful. The calling part is not, calls get dropped, I believe it's rtp issue or maybe nat. I'm very new to kamailio and would like to ask for help on this. Tia!

3 2

[Kamailio-Users] kam 1.5 sqlops sql_query timeout value
by Robert McGilvray 30 Mar '10

30 Mar '10

Hello, I'm using the sqlops module for some custom queries against mysql. Sometimes during a cluster failure the mysqld nodes will hang there waiting for the backend to finish up whatever it's doing, it still accepts the connection and the query but doesn't return results. I have a pair of F5 load balancers in front of the two sql nodes, so there is redundancy as long as the cluster is operational. I'd prefer to keep the cross-site failover in kamailio. I looked through the docs on sqlops and I can't find any reference to a timeout value. I'd like to implement a failover in the script to my other database cluster but if kam waits for a long time before returning a negative it may not work very well. Consider this code in my script for 911 services in my US offices: (I rewrite the rpid/pai and ruri based on IP address then send it to my provider(s)) modparam("sqlops","sqlcon","gokam=>mysql://*****:*****@172.20.180.21/sip _gokam") if (!sql_query("gokam", "select location,cidname,cidnum,ciddomain,e911number,inet_ntoa(network) as network,inet_ntoa(netmask) \ as netmask from e911 where (inet_aton('$si') & netmask) = network", "result")) { sl_send_reply("500", "Database error"); exit; } What I'd like to do is: modparam("sqlops","sqlcon","gokam=>mysql://****:****@172.20.180.21/sip_g okam") modparam("sqlops","sqlcon","gokam_site2=>mysql://*****:*****@172.23.180. 21/sip_gokam") if (!sql_query("gokam", "select location,cidname,cidnum,ciddomain,e911number,inet_ntoa(network) as network,inet_ntoa(netmask) \ as netmask from e911 where (inet_aton('$si') & netmask) = network", "result")) { xlog("L_CRIT", "Primary database failure, using alternate\n"); if (!sql_query("gokam_site2", "select location,cidname,cidnum,ciddomain,e911number,inet_ntoa(network) as network,inet_ntoa(netmask) \ as netmask from e911 where (inet_aton('$si') & netmask) = network", "result")) { sl_send_reply("500", Database error"); exit; } What is the default timeout for sql_query before it returns a negative, is it configurable? Thanks! Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

3 2

[Kamailio-Users] get warning when sip client is not online
by s.maertens＠telenet.be 30 Mar '10

30 Mar '10

Hello, I have a kamailio setup running for several SIP devices. I would like to be warned whenever one of those devices is not reacheable anymore. That means when they go offline or are not anymore registered to the kamailio server. (timeout, unregister, ...). I am using the location table so we would be able to write a script that just at specific times checks the difference between the subscriber table and the location table but maybe there is a much better and more realtime solution. So basically the question is : how to execute a script whenever a host gets removed fom the location table. I am running Kamailio 3.0.1 All hints are very much appreciated :)

3 2

[Kamailio-Users] GSoC 2010 Student Registration is Open
by Daniel-Constantin Mierla 30 Mar '10

30 Mar '10

Hello, if you are a student (or know one) interested in working with SIP and Presence over the summer, earn some money and experience, you have the choice of Kamailio - SIP Router project within SIP Communicator organization: http://www.kamailio.org/w/gsoc-2010/ Registration is now open, up to April 9, more details in above link. If you have further questions, drop an email. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

ser avp question
by jh＠tutpro.com 29 Mar '10

29 Mar '10

modules_s/domain domain_lookup() function can be used to assign domain attributes to avps. for example, lookup_domain("$fd", "foo.bar"), assigns attributes of domain foo.bar to avps $fd.x, where x is name of attribute. my question is, how can i index these avps if there is more than one with the same name? statement $var(attr) = $fd.attr[]; does not give syntax errors, but $var(attr) = $fd.attr[$var(i)]; produces 0(7368) : <core> [cfg.y:3379]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 3174, column 24-30: syntax error 0(7368) : <core> [cfg.y:3379]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 3174, column 24-30: bad command 0(7368) : <core> [cfg.y:3382]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 3174, column 31: bad command 0(7368) : <core> [cfg.y:3382]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 3174, column 31: bad command -- juha

3 5

weird dp_translate() behavior
by jh＠tutpro.com 29 Mar '10

29 Mar '10

while playing with dp_translate(), i noticed the following weird behavior. i have a rule with dpid=11, where match_exp='^\+' and subst_exp and repl_exp are empty. my test goes like this: $rU = "+35892345670"; if (dp_translate("11", $rU/$rU")) { xlog("ruri user is '$rU'\n"); }; $var(test) = "+35892345670"; if (dp_translate("11", "$var(test)/$var(test)")) { xlog("var(test) is '$var(test)'\n"); }; and this is produced to syslog: Mar 29 20:01:39 localhost /usr/sbin/sip-proxy[20533]: ERROR: ruri user is '+35892345670' Mar 29 20:01:39 localhost /usr/sbin/sip-proxy[20533]: ERROR: var(test) is '' that is, $rU is not translated to empty string, but $var(test) is. can someone provide an explanation for this? what should the expected behavior be, when subst_exp and repl_exp are empty? original author of dialplan module is Andreea-Ancuta Onofrei, but i don't know if he is still around. -- juha

1 1

@msg.%s syntax
by jh＠tutpro.com 29 Mar '10

29 Mar '10

first section of page http://sip-router.org/wiki/cookbooks/selects/devel had example @msg.["P-anyheader-youwant"]) that, however, produced syntax error. i removed the dot and got rid of the error. i then went and fixed the wiki page too. however, later on the page there still is @msg.%s and many more like that. are they generated automatically? if so, then fixing them on the wiki page would not help. also, is there somewhere a summary of pv syntax of s and k flavors? -- juha

2 5

Re: [Kamailio-Users] Problem with secure TLS call
by Hemanshu Patel 29 Mar '10

29 Mar '10

With TLS it still is not working Mar 27 11:39:16 [4421] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Mar 27 11:39:16 [4425] WARNING:usrloc:dbrow2info: non-local socket <tcp:172.16.16.218:5091>...ignoring Mar 27 11:39:18 [4431] INFO:core:tls_accept: client did not present a certificate Mar 27 11:39:55 [4432] INFO:core:tls_accept: client did not present a certificate Mar 27 11:40:01 [4431] ERROR:rr:w_record_route: Double attempt to record-route Mar 27 11:40:01 [4426] ERROR:core:tls_connect: SSL_ERROR_SYSCALL err=Connection reset by peer(104) Mar 27 11:40:01 [4426] ERROR:core:tls_connect: something wrong in SSL: 5 (ret=-1) err=Connection reset by peer(104) Mar 27 11:40:01 [4426] ERROR:core:tcp_send: failed to send Mar 27 11:40:01 [4426] ERROR:core:msg_send: tcp_send failed Mar 27 11:40:01 [4425] WARNING:core:run_actions: null action list (rec_level=1) my doubt is that if somethings wrong with SSL certificates that i created myself (i.e my own root CA, and self signed certificates), then how come registration is working like charm? if there is some problem with Certificates then registration should also not work. am i right? ---- Regards, Hemanshu Patel Sr. Software Engg SIS,Ahmedabad Mo:09601295238 On Sat 27/03/10 9:50 AM , "Hemanshu Patel" <hemanshu.patel(a)saicare.com> wrote: > I havent tested over TCP, let me check it > but hardphone, i mean hardware based phones from grandstream gvx3140 > works > fine with same implementation on TLS. > -- > Regards, > Hemanshu Patel > M: 09601295238 > > Does eyebeam with SIP over TCP is working? > > > > Am 26.03.2010 13:43, schrieb Hemanshu Patel: > >> > >> i am still having this problem. > >> when i use two grandstream phone everything works fine, > >> i can make calls on TLS and users can talk to each other. > >> > >> But when i use two eyebream phone, it doesnt work, gives > following error > >> > >> :33 [2875] WARNING:core:init_ssl_ctx_behavior: server > verification NOT > >> activated. Weaker security. > >> [ panreg-tls]$ > >> [ panreg-tls]$ > >> [ panreg-tls]$ Mar 26 18:11:59 [2889] > >> ERROR:rr:w_record_route: Double attempt to record-route > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:09 [2884] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:09 [2884] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:10 [2883] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:10 [2883] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:10 [2883] WARNING:core:run_actions: null action list > >> (rec_level=1) > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:11 [2881] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:11 [2881] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:13 [2882] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:13 [2882] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:19 [2884] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:19 [2884] ERROR:core:msg_send: tcp_send failed > >> > >> > >> > >> Any idea? Same configuration works with grandstream, while they > cant in > >> softphone? and gives 477 errot code. > >> > >> my asterisk log is as below: > >> > >> 26 18:08:09] WARNING[2833]: chan_sip.c:1648 setup_crypto: > Hemanshu: > >> local_key641 jhbAJ7jFE1p/Ngn4kVxy7qTTmkJpeBiN6W98+gmM len 40 > >> -- Called kamailio/1003 > >> -- Got SIP response 477 "Send failed (477/SL)" back from > >> 172.16.16.218 > >> -- No one is available to answer at this time (1:0/0/0) > >> -- Executing [_default:4] Hangup("SIP/5091-8c001430", "") > >> in > >> new stack > >> == Spawn extension (hemu_default, 1003, 4) exited non-zero on > >> 'SIP/5091-8c001430' > >> > ::Disconnected form Oracle, trying to connect again.. > >> > ::Tried a lot, not getting connected.. > >> -- Got SIP response 477 "Send failed (477/SL)" back from > >> 172.16.16.218 > >> > >> > >> Any idea what could be the problem? > >> > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > ---- SIS Blade Server, Desktop & Thin Client is now available on DGS & D rate contract. For more detail kindly visit our website http://www.saicare.com

2 4

Re: [Kamailio-Users] Problem with secure TLS call
by Hemanshu Patel 29 Mar '10

29 Mar '10

Here is the complete call log for One call from two clients eyebream and bria running on single PC. Mar 27 12:36:36 [5492] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:36 [5492] DBG:core:receive_msg: cleaning up Mar 27 12:36:42 [5492] DBG:core:tcp_receive_timeout: 0x7fd335f04798 expired (50, 51) lt=0 Mar 27 12:36:42 [5492] DBG:core:io_watch_del: io_watch_del (0x74efe0, 18, -1, 0x10) fd_no=2 called Mar 27 12:36:42 [5492] DBG:core:release_tcpconn: releasing con 0x7fd335f04798, state 0, fd=18, id=3 Mar 27 12:36:42 [5492] DBG:core:release_tcpconn: extra_data 0x7fd335f017b8 Mar 27 12:36:42 [5499] DBG:core:handle_tcp_child: reader response= 7fd335f04798, 0 from 0 Mar 27 12:36:42 [5499] DBG:core:io_watch_add: io_watch_add(0x74ee80, 24, 2, 0x7fd335f04798), fd_no=17 Mar 27 12:36:42 [5499] DBG:core:handle_tcp_child: cmd CONN_RELEASE 0x7fd335f04798 refcnt= 0 Mar 27 12:36:45 [5499] DBG:core:handle_tcpconn_ev: data available on 0x7fd335eddc18 23 Mar 27 12:36:45 [5499] DBG:core:io_watch_del: io_watch_del (0x74ee80, 23, -1, 0x0) fd_no=18 called Mar 27 12:36:45 [5499] DBG:core:send2child: to tcp child 0 0(5492), 0x7fd335eddc18 Mar 27 12:36:45 [5492] DBG:core:handle_io: received n=8 con=0x7fd335eddc18, fd=18 Mar 27 12:36:45 [5492] DBG:core:io_watch_add: io_watch_add(0x74efe0, 18, 2, 0x7fd335eddc18), fd_no=1 Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 18 Mar 27 12:36:45 [5492] DBG:core:_tls_read: 1109 bytes read Mar 27 12:36:45 [5492] DBG:core:tcp_read_req: content-length= 512 Mar 27 12:36:45 [5492] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5492] DBG:core:parse_msg: method: <INVITE> Mar 27 12:36:45 [5492] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5091;transport=tls> Mar 27 12:36:45 [5492] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-e333132001727d6d-1---d8754z->; state=6 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5492] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5492] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5492] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5492] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=100 Mar 27 12:36:45 [5492] DBG:maxfwd:is_maxfwd_present: value = 70 Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=10 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: <To> [31]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091> ] Mar 27 12:36:45 [5492] DBG:siputils:has_totag: no totag Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=78 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: cseq <CSeq>: <1> <INVITE> Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: start searching: hash=44320, isACK=0 Mar 27 12:36:45 [5492] DBG:tm:matching_3261: RFC3261 transaction matching failed Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: no transaction found Mar 27 12:36:45 [5492] DBG:core:parse_to_param: tag=9f574228 Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={"1003"}, ruri={sip:1003@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5090 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5092 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5091 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: match found for: [172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:check_self: host == me Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=10000 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: content_length=512 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: found end of header Mar 27 12:36:45 [5492] DBG:auth:pre_auth: credentials with given realm not found Mar 27 12:36:45 [5492] DBG:auth:reserve_nonce_index: second= 22, sec_monit= -1, index= 3 Mar 27 12:36:45 [5492] DBG:auth:build_auth_hf: nonce index= 3 Mar 27 12:36:45 [5492] DBG:auth:build_auth_hf: 'Proxy-Authenticate: Digest realm="172.16.16.218", nonce="4badaea30000000316e854bb0194116d66618a39dcd9340c" ' Mar 27 12:36:45 [5492] DBG:sl:send_reply: reply in stateless mode (sl) Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=ffffffffffffffff Mar 27 12:36:45 [5492] DBG:core:check_via_address: params 172.16.17.85, 172.16.17.85, 0 Mar 27 12:36:45 [5492] DBG:core:tcp_send: tcp connection found (0x7fd335eddc18), acquiring fd Mar 27 12:36:45 [5492] DBG:core:tcp_send: c= 0x7fd335eddc18, n=16 Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335eddc18, 1, fd -1 from 7 (5492) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after receive_fd: c= 0x7fd335eddc18 n=8 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: sending... Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 19 Mar 27 12:36:45 [5492] DBG:core:tls_write: write was successful (519 bytes) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after write: c= 0x7fd335eddc18 n=519 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: buf= SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 172.16.17.85:4403;branch=z9hG4bK-d8754z-e333132001727d6d-1---d8754z-;rport=1301 To: <sip:1002@172.16.16.218:5091>;tag=2cf7f9a22b6374f465182a893bba7dec.5636 From: "1003"<sip:1003@172.16.16.218:5091>;tag=9f574228 Call-ID: M2Y2OThmYmYwZjM4ODhhYzk4Mjc0MGFhMDQyNDUzZWU. CSeq: 1 INVITE Proxy-Authenticate: Digest realm="172.16.16.218", nonce="4badaea30000000316e854bb0194116d66618a39dcd9340c" Server: Kamailio (1.5.4-tls (x86_64/linux)) Content-Length: 0 Mar 27 12:36:45 [5492] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5492] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 18 Mar 27 12:36:45 [5492] DBG:core:_tls_read: 370 bytes read Mar 27 12:36:45 [5492] DBG:core:tcp_read_req: content-length= 0 Mar 27 12:36:45 [5492] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5492] DBG:core:parse_msg: method: <ACK> Mar 27 12:36:45 [5492] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5091;transport=tls> Mar 27 12:36:45 [5492] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-e333132001727d6d-1---d8754z->; state=6 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5492] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5492] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5492] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5492] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5492] DBG:core:parse_to_param: tag=2cf7f9a22b6374f465182a893bba7dec.5636 Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: <To> [73]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091>] Mar 27 12:36:45 [5492] DBG:sl:sl_filter_ACK: local ACK found -> dropping it! Mar 27 12:36:45 [5492] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5492] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 18 Mar 27 12:36:45 [5492] DBG:core:_tls_read: 1339 bytes read Mar 27 12:36:45 [5492] DBG:core:tcp_read_req: content-length= 512 Mar 27 12:36:45 [5492] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5492] DBG:core:parse_msg: method: <INVITE> Mar 27 12:36:45 [5492] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5091;transport=tls> Mar 27 12:36:45 [5492] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-863b295f961d253f-1---d8754z->; state=6 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5492] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5492] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5492] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5492] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=100 Mar 27 12:36:45 [5492] DBG:maxfwd:is_maxfwd_present: value = 70 Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=10 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: <To> [31]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091> ] Mar 27 12:36:45 [5492] DBG:siputils:has_totag: no totag Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=78 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: cseq <CSeq>: <2> <INVITE> Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: start searching: hash=44317, isACK=0 Mar 27 12:36:45 [5492] DBG:tm:matching_3261: RFC3261 transaction matching failed Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: no transaction found Mar 27 12:36:45 [5492] DBG:core:parse_to_param: tag=9f574228 Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={"1003"}, ruri={sip:1003@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5090 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5092 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5091 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: match found for: [172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:check_self: host == me Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=10000 Mar 27 12:36:45 [5492] DBG:auth:check_nonce: comparing [4badaea30000000316e854bb0194116d66618a39dcd9340c] and [4badaea30000000316e854bb0194116d66618a39dcd9340c] Mar 27 12:36:45 [5492] DBG:core:db_new_result: allocate 48 bytes for result set at 0x78d6c8 Mar 27 12:36:45 [5492] DBG:db_mysql:db_mysql_get_columns: 1 columns returned from the query Mar 27 12:36:45 [5492] DBG:core:db_allocate_columns: allocate 8 bytes for result names at 0x78aab8 Mar 27 12:36:45 [5492] DBG:core:db_allocate_columns: allocate 4 bytes for result types at 0x79ea58 Mar 27 12:36:45 [5492] DBG:db_mysql:db_mysql_get_columns: allocate 16 bytes for RES_NAMES[0] at 0x78d010 Mar 27 12:36:45 [5492] DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x78d010)[0]=[password] Mar 27 12:36:45 [5492] DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Mar 27 12:36:45 [5492] DBG:core:db_allocate_rows: allocate 16 bytes for rows at 0x79ea38 Mar 27 12:36:45 [5492] DBG:core:db_allocate_row: allocate 32 bytes for row values at 0x79ea78 Mar 27 12:36:45 [5492] DBG:core:db_str2val: converting STRING [1234] Mar 27 12:36:45 [5492] DBG:auth_db:get_ha1: HA1 string calculated: 6650a5aab1d658b465ee304e21b8db36 Mar 27 12:36:45 [5492] DBG:auth:check_response: our result = '19406efa15546eb60e105c164a1feb3d' Mar 27 12:36:45 [5492] DBG:auth:check_response: authorization is OK Mar 27 12:36:45 [5492] DBG:auth:post_auth: nonce index= 3 Mar 27 12:36:45 [5492] DBG:core:db_free_columns: freeing 1 columns Mar 27 12:36:45 [5492] DBG:core:db_free_columns: freeing RES_NAMES[0] at 0x78d010 Mar 27 12:36:45 [5492] DBG:core:db_free_columns: freeing result names at 0x78aab8 Mar 27 12:36:45 [5492] DBG:core:db_free_columns: freeing result types at 0x79ea58 Mar 27 12:36:45 [5492] DBG:core:db_free_rows: freeing 1 rows Mar 27 12:36:45 [5492] DBG:core:db_free_row: freeing row values at 0x79ea78 Mar 27 12:36:45 [5492] DBG:core:db_free_rows: freeing rows at 0x79ea38 Mar 27 12:36:45 [5492] DBG:core:db_free_result: freeing result set at 0x78d6c8 Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=14000 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: content_length=512 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: found end of header Mar 27 12:36:45 [5492] DBG:core:comp_scriptvar: str 29 : 1003 ---- Invite ---- Mar 27 12:36:45 [5492] DBG:core:comp_scriptvar: str 29 : 1003 Mar 27 12:36:45 [5492] ERROR:rr:w_record_route: Double attempt to record-route Mar 27 12:36:45 [5492] DBG:dispatcher:ds_select_dst: set [1] Mar 27 12:36:45 [5492] DBG:dispatcher:ds_select_dst: alg hash [789753065] Mar 27 12:36:45 [5492] DBG:dispatcher:ds_select_dst: selected [0-1/0] <sip:172.16.16.218:5070> Mar 27 12:36:45 [5492] DBG:tm:t_newtran: transaction on entrance=(nil) Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=ffffffffffffffff Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=78 Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: start searching: hash=44317, isACK=0 Mar 27 12:36:45 [5492] DBG:tm:matching_3261: RFC3261 transaction matching failed Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: no transaction found Mar 27 12:36:45 [5492] DBG:tm:run_reqin_callbacks: trans=0x7fd335eee6b0, callback type 1, id 0 entered Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=ffffffffffffffff Mar 27 12:36:45 [5492] DBG:core:check_via_address: params 172.16.17.85, 172.16.17.85, 0 Mar 27 12:36:45 [5492] DBG:core:_shm_resize: resize(0) called Mar 27 12:36:45 [5492] DBG:core:tcp_send: tcp connection found (0x7fd335eddc18), acquiring fd Mar 27 12:36:45 [5492] DBG:core:tcp_send: c= 0x7fd335eddc18, n=16 Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335eddc18, 1, fd -1 from 7 (5492) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after receive_fd: c= 0x7fd335eddc18 n=8 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: sending... Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 19 Mar 27 12:36:45 [5492] DBG:core:tls_write: write was successful (352 bytes) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after write: c= 0x7fd335eddc18 n=352 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: buf= SIP/2.0 100 Giving a try Via: SIP/2.0/TLS 172.16.17.85:4403;branch=z9hG4bK-d8754z-863b295f961d253f-1---d8754z-;rport=1301 To: <sip:1002@172.16.16.218:5091> From: "1003"<sip:1003@172.16.16.218:5091>;tag=9f574228 Call-ID: M2Y2OThmYmYwZjM4ODhhYzk4Mjc0MGFhMDQyNDUzZWU. CSeq: 2 INVITE Server: Kamailio (1.5.4-tls (x86_64/linux)) Content-Length: 0 Mar 27 12:36:45 [5492] DBG:tm:_reply_light: reply sent out. buf=0x79f5e8: SIP/2.0 1..., shmem=0x7fd335f007a8: SIP/2.0 1 Mar 27 12:36:45 [5492] DBG:tm:_reply_light: finished Mar 27 12:36:45 [5492] DBG:core:mk_proxy: doing DNS lookup... Mar 27 12:36:45 [5492] DBG:core:build_req_buf_from_sip_req: id added: <;i=1>, rcv proto=3 Mar 27 12:36:45 [5492] DBG:tm:set_timer: relative timeout is 500000 Mar 27 12:36:45 [5492] DBG:tm:insert_timer_unsafe: [4]: 0x7fd335eee8d0 (54200000) Mar 27 12:36:45 [5492] DBG:tm:set_timer: relative timeout is 30 Mar 27 12:36:45 [5492] DBG:tm:insert_timer_unsafe: [0]: 0x7fd335eee900 (83) Mar 27 12:36:45 [5492] DBG:tm:t_relay_to: new transaction fwd'ed Mar 27 12:36:45 [5492] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Mar 27 12:36:45 [5492] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5492] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5487] DBG:core:parse_msg: SIP Reply (status): Mar 27 12:36:45 [5487] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5487] DBG:core:parse_msg: status: <100> Mar 27 12:36:45 [5487] DBG:core:parse_msg: reason: <Trying> Mar 27 12:36:45 [5487] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKd1da.1a6f9b73.0>; state=6 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 236, = <1>; state=6 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 234, <received> = <172.16.16.218>; state=16 Mar 27 12:36:45 [5487] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5487] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5487] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5487] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5487] DBG:core:forward_reply: found module tm, passing reply to it Mar 27 12:36:45 [5487] DBG:tm:t_check: start=0xffffffffffffffff Mar 27 12:36:45 [5487] DBG:core:parse_headers: flags=22 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 234, <received> = <172.16.17.85>; state=6 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-863b295f961d253f-1---d8754z->; state=6 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 235, <rport> = <1301>; state=16 Mar 27 12:36:45 [5487] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5487] DBG:core:parse_headers: via found, flags=22 Mar 27 12:36:45 [5487] DBG:core:parse_headers: parse_headers: this is the second via Mar 27 12:36:45 [5487] DBG:core:parse_to: end of header reached, state=10 Mar 27 12:36:45 [5487] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5487] DBG:core:get_hdr_field: <To> [31]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5487] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091> ] Mar 27 12:36:45 [5487] DBG:core:get_hdr_field: cseq <CSeq>: <2> <INVITE> Mar 27 12:36:45 [5487] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5487] DBG:tm:t_reply_matching: hash 44317 label 934934177 branch 0 Mar 27 12:36:45 [5487] DBG:tm:t_reply_matching: REF_UNSAFE: after is 1 Mar 27 12:36:45 [5487] DBG:tm:t_reply_matching: reply matched (T=0x7fd335eee6b0)! Mar 27 12:36:45 [5487] DBG:tm:t_check: end=0x7fd335eee6b0 Mar 27 12:36:45 [5487] DBG:tm:reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) Mar 27 12:36:45 [5487] DBG:tm:t_should_relay_response: T_code=100, new_code=100 Mar 27 12:36:45 [5487] DBG:tm:relay_reply: branch=0, save=0, relay=-1 Mar 27 12:36:45 [5487] DBG:tm:set_timer: relative timeout is 120 Mar 27 12:36:45 [5487] DBG:tm:insert_timer_unsafe: [1]: 0x7fd335eee900 (173) Mar 27 12:36:45 [5487] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Mar 27 12:36:45 [5487] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5487] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5489] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5489] DBG:core:parse_msg: method: <INVITE> Mar 27 12:36:45 [5489] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5090> Mar 27 12:36:45 [5489] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5489] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5489] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK7f71d414>; state=6 Mar 27 12:36:45 [5489] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5489] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5489] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5489] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5489] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5489] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5489] DBG:core:parse_headers: flags=100 Mar 27 12:36:45 [5489] DBG:core:parse_to: end of header reached, state=10 Mar 27 12:36:45 [5489] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5090} Mar 27 12:36:45 [5489] DBG:core:get_hdr_field: <To> [31]; uri=[sip:1002@172.16.16.218:5090] Mar 27 12:36:45 [5489] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5090> ] Mar 27 12:36:45 [5489] DBG:core:get_hdr_field: cseq <CSeq>: <102> <INVITE> Mar 27 12:36:45 [5489] DBG:maxfwd:is_maxfwd_present: value = 70 ---- From SC/MS/IC ---- Mar 27 12:36:45 [5489] DBG:core:rewrite_uri: rewriting Request-URI with 'sip:1002@172.16.17.85:25923;rinstance=133b23266b1f2e83;transport=TLS' Mar 27 12:36:45 [5489] DBG:core:mk_proxy: doing DNS lookup... Mar 27 12:36:45 [5489] DBG:core:parse_headers: flags=2000 Mar 27 12:36:45 [5489] DBG:core:get_hdr_field: content_length=394 Mar 27 12:36:45 [5489] DBG:core:forward_request: sending: INVITE sip:1002@172.16.17.85:25923;rinstance=133b23266b1f2e83;transport=TLS SIP/2.0 Record-Route: <sip:172.16.16.218:5091;transport=tls;r2=on;lr=on> Record-Route: <sip:172.16.16.218:5090;r2=on;lr=on> Via: SIP/2.0/TLS 172.16.16.218:5091;branch=z9hG4bK7f71d414 Via: SIP/2.0/UDP 172.16.16.218:5070;received=172.16.16.218;branch=z9hG4bK7f71d414;rport=5070 From: "1003" <sip:1003@172.16.16.218:5070>;tag=as7351a581 To: <sip:1002@172.16.16.218:5090> Contact: <sip:1003@172.16.16.218:5070> Call-ID: 7c10eaf0305d3dff6dac07db404d5a60(a)172.16.16.218 CSeq: 102 INVITE User-Agent: PANTHER-SC Max-Forwards: 69 Date: Sat, 27 Mar 2010 07:06:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 394 v=0 o=root 2434 2434 IN IP4 172.16.17.85 s=session c=IN IP4 172.16.17.85 t=0 0 m=audio 54168 RTP/SAVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv AES_CM_128_HMAC_SHA1_80 inline:qGi2zArgAgHDiUoqc0wbBofpBISpZINZxrqDoa7W . Mar 27 12:36:45 [5489] DBG:core:forward_request: orig. len=935, new_len=1182, proto=3 Mar 27 12:36:45 [5489] DBG:core:tcp_send: no open tcp connection found, opening new one Mar 27 12:36:45 [5489] DBG:core:print_ip: tcpconn_new: new tcp connection to: 172.16.17.85 Mar 27 12:36:45 [5489] DBG:core:tcpconn_new: on port 25923, type 3 Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: entered: Creating a whole new ssl connection Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: name based TLS client domains are disabled Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: no TLS client domain AVP set, looking for socket based TLS client domain Mar 27 12:36:45 [5489] DBG:core:tls_find_client_domain: virtual TLS client domain not found, Using default TLS client domain settings Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: found socket based TLS client domain [0.0.0.0:0] Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: Setting in CONNECT mode (client) Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: TLS server extension is not activated Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: TLS server name is not set, sending ClientHello without server_name Mar 27 12:36:45 [5489] DBG:core:tcp_send: sending... Mar 27 12:36:45 [5489] DBG:core:tls_update_fd: New fd is 12 Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335f23098, 2, fd 25 from 4 (5489) Mar 27 12:36:45 [5499] DBG:core:tcpconn_add: hashes: 495, 4 Mar 27 12:36:45 [5499] DBG:core:io_watch_add: io_watch_add(0x74ee80, 25, 2, 0x7fd335f23098), fd_no=17 Mar 27 12:36:45 [5499] DBG:core:handle_tcpconn_ev: data available on 0x7fd335f23098 25 Mar 27 12:36:45 [5489] ERROR:core:tls_connect: SSL_ERROR_SYSCALL err=Connection reset by peer(104) Mar 27 12:36:45 [5499] DBG:core:io_watch_del: io_watch_del (0x74ee80, 25, -1, 0x0) fd_no=18 called Mar 27 12:36:45 [5489] ERROR:core:tls_connect: something wrong in SSL: 5 (ret=-1) err=Connection reset by peer(104) Mar 27 12:36:45 [5499] DBG:core:send2child: to tcp child 1 0(5494), 0x7fd335f23098 Mar 27 12:36:45 [5489] DBG:core:tcp_send: after write: c= 0x7fd335f23098 n=-1 fd=12 Mar 27 12:36:45 [5489] DBG:core:tcp_send: buf= INVITE sip:1002@172.16.17.85:25923;rinstance=133b23266b1f2e83;transport=TLS SIP/2.0 Record-Route: <sip:172.16.16.218:5091;transport=tls;r2=on;lr=on> Record-Route: <sip:172.16.16.218:5090;r2=on;lr=on> Via: SIP/2.0/TLS 172.16.16.218:5091;branch=z9hG4bK7f71d414 Via: SIP/2.0/UDP 172.16.16.218:5070;received=172.16.16.218;branch=z9hG4bK7f71d414;rport=5070 From: "1003" <sip:1003@172.16.16.218:5070>;tag=as7351a581 To: <sip:1002@172.16.16.218:5090> Contact: <sip:1003@172.16.16.218:5070> Call-ID: 7c10eaf0305d3dff6dac07db404d5a60(a)172.16.16.218 CSeq: 102 INVITE User-Agent: PANTHER-SC Max-Forwards: 69 Date: Sat, 27 Mar 2010 07:06:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 394 v=0 o=root 2434 2434 IN IP4 172.16.17.85 s=session c=IN IP4 172.16.17.85 t=0 0 m=audio 54168 RTP/SAVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv AES_CM_128_HMAC_SHA1_80 inline:qGi2zArgAgHDiUoqc0wbBofpBISpZINZxrqDoa7W Mar 27 12:36:45 [5494] DBG:core:handle_io: received n=8 con=0x7fd335f23098, fd=20 Mar 27 12:36:45 [5489] ERROR:core:tcp_send: failed to send Mar 27 12:36:45 [5494] DBG:core:io_watch_add: io_watch_add(0x74efe0, 20, 2, 0x7fd335f23098), fd_no=1 Mar 27 12:36:45 [5489] ERROR:core:msg_send: tcp_send failed Mar 27 12:36:45 [5494] DBG:core:io_watch_del: io_watch_del (0x74efe0, 20, -1, 0x10) fd_no=2 called Mar 27 12:36:45 [5494] DBG:core:release_tcpconn: releasing con 0x7fd335f23098, state -2, fd=20, id=4 Mar 27 12:36:45 [5494] DBG:core:release_tcpconn: extra_data 0x7fd335f00ef8 Mar 27 12:36:45 [5489] DBG:sl:sl_reply_error: error text is Send failed (477/SL) Mar 27 12:36:45 [5499] DBG:core:handle_tcp_child: reader response= 7fd335f23098, -2 from 1 Mar 27 12:36:45 [5499] DBG:core:tcpconn_destroy: delaying (0x7fd335f23098, flags 0002) ... Mar 27 12:36:45 [5489] DBG:core:parse_headers: flags=ffffffffffffffff Mar 27 12:36:45 [5489] DBG:core:get_hdr_field: found end of header Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335f23098, -2, fd -1 from 4 (5489) Mar 27 12:36:45 [5499] DBG:core:tcpconn_destroy: destroying connection 0x7fd335f23098, flags 0002 Mar 27 12:36:45 [5489] DBG:core:check_via_address: params 172.16.16.218, 172.16.16.218, 0 Mar 27 12:36:45 [5499] DBG:core:tls_close: closing SSL connection Mar 27 12:36:45 [5499] DBG:core:tls_update_fd: New fd is 25 Mar 27 12:36:45 [5499] DBG:core:tls_shutdown: shutdown successful Mar 27 12:36:45 [5499] DBG:core:tls_tcpconn_clean: Cleanup function entered Mar 27 12:36:45 [5486] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5486] DBG:core:parse_msg: method: <ACK> Mar 27 12:36:45 [5486] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5090> Mar 27 12:36:45 [5486] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5486] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5486] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK7f71d414>; state=6 Mar 27 12:36:45 [5486] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5486] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5486] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5486] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5486] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5486] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5486] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5486] DBG:core:parse_to_param: tag=2cf7f9a22b6374f465182a893bba7dec.adb6 Mar 27 12:36:45 [5486] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5486] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5090} Mar 27 12:36:45 [5486] DBG:core:get_hdr_field: <To> [73]; uri=[sip:1002@172.16.16.218:5090] Mar 27 12:36:45 [5486] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5090>] Mar 27 12:36:45 [5486] DBG:sl:sl_filter_ACK: local ACK found -> dropping it! Mar 27 12:36:45 [5486] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5486] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5489] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5489] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5485] DBG:core:parse_msg: SIP Reply (status): Mar 27 12:36:45 [5485] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5485] DBG:core:parse_msg: status: <404> Mar 27 12:36:45 [5485] DBG:core:parse_msg: reason: <Not Found> Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKd1da.1a6f9b73.0>; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 236, = <1>; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 234, <received> = <172.16.16.218>; state=16 Mar 27 12:36:45 [5485] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5485] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5485] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5485] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5485] DBG:core:forward_reply: found module tm, passing reply to it Mar 27 12:36:45 [5485] DBG:tm:t_check: start=0xffffffffffffffff Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=22 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 234, <received> = <172.16.17.85>; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-863b295f961d253f-1---d8754z->; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 235, <rport> = <1301>; state=16 Mar 27 12:36:45 [5485] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5485] DBG:core:parse_headers: via found, flags=22 Mar 27 12:36:45 [5485] DBG:core:parse_headers: parse_headers: this is the second via Mar 27 12:36:45 [5485] DBG:core:parse_to_param: tag=as06cb485b Mar 27 12:36:45 [5485] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5485] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: <To> [46]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091>] Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: cseq <CSeq>: <2> <INVITE> Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5485] DBG:tm:t_reply_matching: hash 44317 label 934934177 branch 0 Mar 27 12:36:45 [5485] DBG:tm:t_reply_matching: REF_UNSAFE: after is 1 Mar 27 12:36:45 [5485] DBG:tm:t_reply_matching: reply matched (T=0x7fd335eee6b0)! Mar 27 12:36:45 [5485] DBG:tm:t_check: end=0x7fd335eee6b0 Mar 27 12:36:45 [5485] DBG:tm:reply_received: org. status uas=100, uac[0]=100 local=0 is_invite=1) Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5485] DBG:tm:build_local: using FROM=<From: "1003"<sip:1003@172.16.16.218:5091>;tag=9f574228 >, TO=<To: <sip:1002@172.16.16.218:5091>;tag=as06cb485b >, CSEQ_N=<CSeq: 2> Mar 27 12:36:45 [5485] DBG:tm:extract_parsed_hdrs: ----parsing the buf req - first line Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=ffffffffffffffff Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKd1da.1a6f9b73.0>; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 236, = <1>; state=16 Mar 27 12:36:45 [5485] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5485] DBG:core:parse_headers: via found, flags=ffffffffffffffff Mar 27 12:36:45 [5485] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 234, <received> = <172.16.17.85>; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-863b295f961d253f-1---d8754z->; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 235, <rport> = <1301>; state=16 Mar 27 12:36:45 [5485] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5485] DBG:core:parse_headers: via found, flags=ffffffffffffffff Mar 27 12:36:45 [5485] DBG:core:parse_headers: parse_headers: this is the second via Mar 27 12:36:45 [5485] DBG:core:parse_to: end of header reached, state=10 Mar 27 12:36:45 [5485] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: <To> [31]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091> ] Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: cseq <CSeq>: <2> <INVITE> Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: content_length=512 Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: found end of header Mar 27 12:36:45 [5485] DBG:tm:t_should_relay_response: T_code=100, new_code=404 Mar 27 12:36:45 [5485] DBG:tm:t_pick_branch: picked branch 0, code 404 Mar 27 12:36:45 [5485] DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=404, flags=2 Mar 27 12:36:45 [5485] WARNING:core:run_actions: null action list (rec_level=1) Mar 27 12:36:45 [5485] DBG:tm:relay_reply: branch=0, save=0, relay=0 Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=2000 Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: content_length=0 Mar 27 12:36:45 [5485] DBG:core:build_res_buf_from_sip_res: old size: 548, new size: 454 Mar 27 12:36:45 [5485] DBG:core:build_res_buf_from_sip_res: copied size: orig:117, new: 23, rest: 431 msg= SIP/2.0 404 Not Found Via: SIP/2.0/TLS 172.16.17.85:4403;received=172.16.17.85;branch=z9hG4bK-d8754z-863b295f961d253f-1---d8754z-;rport=1301 From: "1003"<sip:1003@172.16.16.218:5091>;tag=9f574228 To: <sip:1002@172.16.16.218:5091>;tag=as06cb485b Call-ID: M2Y2OThmYmYwZjM4ODhhYzk4Mjc0MGFhMDQyNDUzZWU. CSeq: 2 INVITE User-Agent: PANTHER-SC Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Mar 27 12:36:45 [5485] DBG:tm:set_timer: relative timeout is 30 Mar 27 12:36:45 [5485] DBG:tm:insert_timer_unsafe: [0]: 0x7fd335eee828 (83) Mar 27 12:36:45 [5485] DBG:core:tcp_send: tcp connection found (0x7fd335eddc18), acquiring fd Mar 27 12:36:45 [5485] DBG:core:tcp_send: c= 0x7fd335eddc18, n=16 Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335eddc18, 1, fd -1 from 1 (5485) Mar 27 12:36:45 [5485] DBG:core:tcp_send: after receive_fd: c= 0x7fd335eddc18 n=8 fd=9 Mar 27 12:36:45 [5485] DBG:core:tcp_send: sending... Mar 27 12:36:45 [5485] DBG:core:tls_update_fd: New fd is 9 Mar 27 12:36:45 [5485] DBG:core:tls_write: write was successful (454 bytes) Mar 27 12:36:45 [5485] DBG:core:tcp_send: after write: c= 0x7fd335eddc18 n=454 fd=9 Mar 27 12:36:45 [5485] DBG:core:tcp_send: buf= SIP/2.0 404 Not Found Via: SIP/2.0/TLS 172.16.17.85:4403;received=172.16.17.85;branch=z9hG4bK-d8754z-863b295f961d253f-1---d8754z-;rport=1301 From: "1003"<sip:1003@172.16.16.218:5091>;tag=9f574228 To: <sip:1002@172.16.16.218:5091>;tag=as06cb485b Call-ID: M2Y2OThmYmYwZjM4ODhhYzk4Mjc0MGFhMDQyNDUzZWU. CSeq: 2 INVITE User-Agent: PANTHER-SC Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Mar 27 12:36:45 [5485] DBG:tm:relay_reply: sent buf=0x7916f0: SIP/2.0 4..., shmem=0x7fd335f007a8: SIP/2.0 4 Mar 27 12:36:45 [5485] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Mar 27 12:36:45 [5485] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Mar 27 12:36:45 [5485] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5485] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 18 Mar 27 12:36:45 [5492] DBG:core:_tls_read: 343 bytes read Mar 27 12:36:45 [5492] DBG:core:tcp_read_req: content-length= 0 Mar 27 12:36:45 [5492] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5492] DBG:core:parse_msg: method: <ACK> Mar 27 12:36:45 [5492] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5091;transport=tls> Mar 27 12:36:45 [5492] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-863b295f961d253f-1---d8754z->; state=6 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5492] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5492] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5492] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5492] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5492] DBG:core:parse_to_param: tag=as06cb485b Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: <To> [46]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091>] Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=100 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: cseq <CSeq>: <2> <ACK> Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: content_length=0 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: found end of header Mar 27 12:36:45 [5492] DBG:maxfwd:is_maxfwd_present: max_forwards header not found! Mar 27 12:36:45 [5492] DBG:siputils:has_totag: totag found Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=200 Mar 27 12:36:45 [5492] DBG:rr:find_first_route: No Route headers found Mar 27 12:36:45 [5492] DBG:rr:loose_route: There is no Route HF Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=78 Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: start searching: hash=44317, isACK=1 Mar 27 12:36:45 [5492] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d8754z-863b295f961d253f-1---d8754z- Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: transaction found (T=0x7fd335eee6b0) Mar 27 12:36:45 [5492] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Mar 27 12:36:45 [5492] DBG:tm:insert_timer_unsafe: [2]: 0x7fd335eee730 (58) Mar 27 12:36:45 [5492] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Mar 27 12:36:45 [5492] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5492] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 18 Mar 27 12:36:45 [5492] DBG:core:_tls_read: 4 bytes read Mar 27 12:36:45 [5492] DBG:core:tcp_read_req: content-length= 0 Mar 27 12:36:45 [5492] DBG:core:tcp_send: tcp connection found (0x7fd335eddc18), acquiring fd Mar 27 12:36:45 [5492] DBG:core:tcp_send: c= 0x7fd335eddc18, n=16 Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335eddc18, 1, fd -1 from 7 (5492) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after receive_fd: c= 0x7fd335eddc18 n=8 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: sending... Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 19 Mar 27 12:36:45 [5492] DBG:core:tls_write: write was successful (2 bytes) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after write: c= 0x7fd335eddc18 n=2 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: buf= one more thing, i have got two clients from one PC, still look at netstat result hemanshu@localhost user]$ netstat -antp | grep kamailio (Not all processes could be identified, non-owned process info will not be shown, you would have to be root to see it all.) tcp 0 0 172.16.16.218:5091 0.0.0.0:* LISTEN 4947/kamailio tcp 0 0 172.16.16.218:5092 0.0.0.0:* LISTEN 4947/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1295 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1296 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1289 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1293 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1291 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1286 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1285 ESTABLISHED 4964/kamailio Both clients keep creating new connection, and not reusing the already connected connection. i this this si the problem ---- Regards, Hemanshu Patel Sr. Software Engg Mo:09601295238 On Sat 27/03/10 11:37 AM , Hemanshu Patel <hemanshu.patel(a)saicare.com> wrote: > With TLS it still is not working > Mar 27 11:39:16 [4421] INFO:core:probe_max_receive_buffer: using a > UDP receive buffer of 255 kb > Mar 27 11:39:16 [4425] WARNING:usrloc:dbrow2info: non-local socket > ...ignoring > Mar 27 11:39:18 [4431] INFO:core:tls_accept: client did not present > a certificate > Mar 27 11:39:55 [4432] INFO:core:tls_accept: client did not present > a certificate > Mar 27 11:40:01 [4431] ERROR:rr:w_record_route: Double attempt to > record-route > Mar 27 11:40:01 [4426] ERROR:core:tls_connect: SSL_ERROR_SYSCALL > err=Connection reset by peer(104) > Mar 27 11:40:01 [4426] ERROR:core:tls_connect: something wrong in > SSL: 5 (ret=-1) err=Connection reset by peer(104) > Mar 27 11:40:01 [4426] ERROR:core:tcp_send: failed to send > Mar 27 11:40:01 [4426] ERROR:core:msg_send: tcp_send failed > Mar 27 11:40:01 [4425] WARNING:core:run_actions: null action list > (rec_level=1) > my doubt is that if somethings wrong with SSL certificates that i > created myself (i.e my own root CA, and self signed certificates), > then how come registration is working like charm? > if there is some problem with Certificates then registration should > also not work. > am i right? > ---- > Regards, > Hemanshu Patel > Sr. Software Engg > SIS,Ahmedabad > Mo:09601295238 > On Sat 27/03/10 9:50 AM , "Hemanshu Patel" wrote: > > I havent tested over TCP, let me check it > > but hardphone, i mean hardware based phones from grandstream > gvx3140 > > works > > fine with same implementation on TLS. > > -- > > Regards, > > Hemanshu Patel > > M: 09601295238 > > > Does eyebeam with SIP over TCP is working? > > > > > > Am 26.03.2010 13:43, schrieb Hemanshu Patel: > > >> > > >> i am still having this problem. > > >> when i use two grandstream phone everything works fine, > > >> i can make calls on TLS and users can talk to each other. > > >> > > >> But when i use two eyebream phone, it doesnt work, gives > > following error > > >> > > >> :33 [2875] WARNING:core:init_ssl_ctx_behavior: server > > verification NOT > > >> activated. Weaker security. > > >> [ panreg-tls]$ > > >> [ panreg-tls]$ > > >> [ panreg-tls]$ Mar 26 18:11:59 [2889] > > >> ERROR:rr:w_record_route: Double attempt to record-route > > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_blocking_connect: timeout > > 10 s > > >> elapsed from 10 s > > >> Mar 26 18:12:09 [2884] ERROR:core:tcpconn_connect: > > tcp_blocking_connect > > >> failed > > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_send: connect failed > > >> Mar 26 18:12:09 [2884] ERROR:core:msg_send: tcp_send failed > > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_blocking_connect: timeout > > 10 s > > >> elapsed from 10 s > > >> Mar 26 18:12:10 [2883] ERROR:core:tcpconn_connect: > > tcp_blocking_connect > > >> failed > > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_send: connect failed > > >> Mar 26 18:12:10 [2883] ERROR:core:msg_send: tcp_send failed > > >> Mar 26 18:12:10 [2883] WARNING:core:run_actions: null action > list > > >> (rec_level=1) > > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_blocking_connect: timeout > > 10 s > > >> elapsed from 10 s > > >> Mar 26 18:12:11 [2881] ERROR:core:tcpconn_connect: > > tcp_blocking_connect > > >> failed > > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_send: connect failed > > >> Mar 26 18:12:11 [2881] ERROR:core:msg_send: tcp_send failed > > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_blocking_connect: timeout > > 10 s > > >> elapsed from 10 s > > >> Mar 26 18:12:13 [2882] ERROR:core:tcpconn_connect: > > tcp_blocking_connect > > >> failed > > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_send: connect failed > > >> Mar 26 18:12:13 [2882] ERROR:core:msg_send: tcp_send failed > > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_blocking_connect: timeout > > 10 s > > >> elapsed from 10 s > > >> Mar 26 18:12:19 [2884] ERROR:core:tcpconn_connect: > > tcp_blocking_connect > > >> failed > > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_send: connect failed > > >> Mar 26 18:12:19 [2884] ERROR:core:msg_send: tcp_send failed > > >> > > >> > > >> > > >> Any idea? Same configuration works with grandstream, while they > > cant in > > >> softphone? and gives 477 errot code. > > >> > > >> my asterisk log is as below: > > >> > > >> 26 18:08:09] WARNING[2833]: chan_sip.c:1648 setup_crypto: > > Hemanshu: > > >> local_key641 jhbAJ7jFE1p/Ngn4kVxy7qTTmkJpeBiN6W98+gmM len 40 > > >> -- Called kamailio/1003 > > >> -- Got SIP response 477 "Send failed (477/SL)" back from > > >> 172.16.16.218 > > >> -- No one is available to answer at this time (1:0/0/0) > > >> -- Executing [_default:4] Hangup("SIP/5091-8c001430", "") > > >> in > > >> new stack > > >> == Spawn extension (hemu_default, 1003, 4) exited non-zero > on > > >> 'SIP/5091-8c001430' > > >> > ::Disconnected form Oracle, trying to connect > again.. > > >> > ::Tried a lot, not getting connected.. > > >> -- Got SIP response 477 "Send failed (477/SL)" back from > > >> 172.16.16.218 > > >> > > >> > > >> Any idea what could be the problem? > > >> > > > > > _______________________________________________ > > Kamailio (OpenSER) - Users mailing list > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > > ---- SIS Blade Server, Desktop & Thin Client is now available on DGS > & D rate contract. For more detail kindly visit our website > http://www.saicare.com > > ---- SIS Blade Server, Desktop & Thin Client is now available on DGS & D rate contract. For more detail kindly visit our website http://www.saicare.com

2 1

INVITE timers
by Alex Hermann 27 Mar '10

27 Mar '10

Hello, I'm trying to create a scenarion which was very simple with Kamailio 1.x, but seems impossible with sr3.0. I need to do failover for INVITE's and also limit the call setup time per-branch, so I defined the problem in 2 parts that need to work separately, but also combined into 1 proxy. 1) Fast failover. After t_relay, the destination should be tried for max 2 sec per host. No maximum for the whole set. Thus, if a destination has 4 SRV records, and each host fails after 2 secs, a 408 will be sent back after a total of 8 secs. If a (proivisional) response arrives within 2 sec, a 408 MUST NOT be sent. The exact value of the timer should be configurable per t_relay call. In kamailio this is accomplished by setting the fr_timer(_avp) to 2000ms. In sr3.0 I have yet to find a combination of timer settings that work. 2) Call setup duration. If a destination host returns a response, I want to limit the attempt to 80 secs (default, should be configurable per t_relay call) of ringing/progress/whatever. After that a failure route will take care to route to a different destination, at this time this timer should be restarted. At no other time after the _first_ provisional reponse after the t_relay call should the timer be restarted. In Kamailio this was accomplished by just setting fr_inv_timer(_avp) to the desired value before t_relay(). In sr3.0 this timer will be restarted after additional provisional responses and will not be resetted after a new t_relay call (from failure_route). Any suggestions on how to get the desired behaviours are welcomed (at the moment scenario 1 is more important than 2). -- Greetings, Alex Hermann

3 6

Re: [Kamailio-Users] Problem with secure TLS call
by Hemanshu Patel 27 Mar '10

27 Mar '10

Yups, it works fine with TCP. All packets are forwarded to other party perfectly and everything work smooth. only with TLS, on softphone it doesnt work. any more suggestions ---- Regards, Hemanshu Patel Sr. Software Engg SIS,Ahmedabad Mo:09601295238 On Sat 27/03/10 9:50 AM , "Hemanshu Patel" <hemanshu.patel(a)saicare.com> wrote: > I havent tested over TCP, let me check it > but hardphone, i mean hardware based phones from grandstream gvx3140 > works > fine with same implementation on TLS. > -- > Regards, > Hemanshu Patel > M: 09601295238 > > Does eyebeam with SIP over TCP is working? > > > > Am 26.03.2010 13:43, schrieb Hemanshu Patel: > >> > >> i am still having this problem. > >> when i use two grandstream phone everything works fine, > >> i can make calls on TLS and users can talk to each other. > >> > >> But when i use two eyebream phone, it doesnt work, gives > following error > >> > >> :33 [2875] WARNING:core:init_ssl_ctx_behavior: server > verification NOT > >> activated. Weaker security. > >> [ panreg-tls]$ > >> [ panreg-tls]$ > >> [ panreg-tls]$ Mar 26 18:11:59 [2889] > >> ERROR:rr:w_record_route: Double attempt to record-route > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:09 [2884] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:09 [2884] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:10 [2883] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:10 [2883] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:10 [2883] WARNING:core:run_actions: null action list > >> (rec_level=1) > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:11 [2881] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:11 [2881] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:13 [2882] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:13 [2882] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:19 [2884] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:19 [2884] ERROR:core:msg_send: tcp_send failed > >> > >> > >> > >> Any idea? Same configuration works with grandstream, while they > cant in > >> softphone? and gives 477 errot code. > >> > >> my asterisk log is as below: > >> > >> 26 18:08:09] WARNING[2833]: chan_sip.c:1648 setup_crypto: > Hemanshu: > >> local_key641 jhbAJ7jFE1p/Ngn4kVxy7qTTmkJpeBiN6W98+gmM len 40 > >> -- Called kamailio/1003 > >> -- Got SIP response 477 "Send failed (477/SL)" back from > >> 172.16.16.218 > >> -- No one is available to answer at this time (1:0/0/0) > >> -- Executing [_default:4] Hangup("SIP/5091-8c001430", "") > >> in > >> new stack > >> == Spawn extension (hemu_default, 1003, 4) exited non-zero on > >> 'SIP/5091-8c001430' > >> > ::Disconnected form Oracle, trying to connect again.. > >> > ::Tried a lot, not getting connected.. > >> -- Got SIP response 477 "Send failed (477/SL)" back from > >> 172.16.16.218 > >> > >> > >> Any idea what could be the problem? > >> > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > ---- SIS Blade Server, Desktop & Thin Client is now available on DGS & D rate contract. For more detail kindly visit our website http://www.saicare.com

1 0

[Kamailio-Users] Problem with secure TLS call
by Detlef Pilzecker 27 Mar '10

27 Mar '10

Hi, I am using kamailio 3.0.1 (x86_64/linux) 0822a9 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. and I am using the client "PhonerLite" (http://www.phonerlite.de). The phone is registered on my server with TLS. If I call someone or myself I get an error message. What is wrong with my server? (TLS calls with PhonerLite with the provider antisip.com are possible without problems!) With UDP calling myself I get "486:Busy Here" on the phone, this is OK and normal. With TLS I get "477:Unfortunately error on sending to next hop occurred(477/SL)" on the phone and some ERROR messages like the following on the kamailio.log (you can see the rest of the log in the attached zip): Mar 21 20:24:00 vs208140 /usr/local/sbin/kamailio[20013]: DEBUG: <core> [msg_translator.c:200]: check_via_address(95.90.205.74, 95.90.205.74, 0) Mar 21 20:24:00 vs208140 /usr/local/sbin/kamailio[20013]: DEBUG: <core> [tcp_main.c:1786]: tcp_send: no open tcp connection found, opening new one Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: <core> [tcp_main.c:618]: connect 95.90.205.74:5061 failed (timeout) Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: <core> [tcp_main.c:621]: ERROR: tcp_blocking_connect 95.90.205.74:5061: timeout 10 s elapsed from 10 s Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: <core> [tcp_main.c:1168]: ERROR: tcp_do_connect: tcp_blocking_connect 95.90.205.74:5061 failed Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: <core> [tcp_main.c:1237]: ERROR: tcp_do_connect 95.90.205.74:5061: failed (115) Operation now in progress Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: <core> [tcp_main.c:1955]: ERROR: tcp_send 95.90.205.74:5061: connect failed Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: tm [../../forward.h:191]: msg_send: ERROR: tcp_send failed The kamailio.cfg file is also in the zip attached. The PC is connected with the router (I opened the TCP ports 5060-5062) that is connected to the Internet. The problem also happens if I disable the firewall of the server and of the PC. Can someone give me help? Thanks in advance! Regards Detlef Pilzecker

3 8

topoh
by dotnetdub 26 Mar '10

26 Mar '10

Hi List, We are using Kamailo 3.01 and testing topoh for starters: Kamailio3 = IP of Kamailio Server and Asterisk=IP of Asterisk server Observations: U 2010/03/26 12:05:56.665583 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 100 trying -- your call is important to us^@ Via: SIP/2.0/UDP Kamailio3:5060;branch=z9hG4bK08fffea4;rport=5060^@ From: "xxxxxxxxx" <sip:xxxxx@asterisk>;tag=as7ac63338^@ To: <sip:Xxxxxxx@Kamailio3>^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Server: kamailio (3.0.1 (i386/linux))^@ Content-Length: 0^@ Warning: 392 Kamailio:5060 "Noisy feedback tells: pid=23782 req_src_ip=Asterisk req_src_port=5060 in_uri=sip:xxxxxx@Kamailioout_uri=sip:Translated@ *UPSTREAMPROVIDERSIP* Provider via_cnt==1"^@ The noisy feedback contains everything that is hidden by topoh in the contact: Contact: <sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**>^ Also: U 2010/03/26 12:06:00.125828 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 180 Ringing^@ From: "xxxxxxx"<sip:xxxxxxxx@Kamailio3>;tag=as7ac63338^@ To: <sip:xxxxxx@Kamailio3 >;tag=fccd978-b342ea59-13c4-50022-82e974-27d2fa70-82e974^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Via: SIP/2.0/UDP Asterisk:5060;branch=z9hG4bK08fffea4;rport=5060^@ Record-Route: <sip:10.1.1.10;line=sr-N6IAzBclOBF4WLZfWBF7M.VLz6fLHRQ7z6srpxusg9MXgSIqHReEgJZwgSKEgSjLMc**>^@ Record-Route: <sip:Kamailio3;lr=on;did=bff.60851b12>^@ Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY^@ Contact: <sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**>^@ Content-Type: application/sdp^@ Content-Length: 228^@ ^@ v=0^@ *o=Upstream Switch Name 160624700 160624700 IN IP4 Upstream Providers IP^@* *s=Upstream Switch Name^@* * * The upstream info appears here also. I am using engage_media_proxy and using topoh with defaul parameters. Am I missing something ? Thanks, Stephen.

3 6

[Kamailio-Users] Recording RTP session
by Hector.Ortiz＠swisscom.com 26 Mar '10

26 Mar '10

Hi everybody I've configured a SIP server (OpenSER 1.3.2) to work with rtpproxy and so far everything is fine, but now I want to be able to record the audio from a conversation. I've read that this is possible by using the function start_recording from the module NATHelper but I'm not able to get it working RTPproxy 1.2.1 is running with the following options: root@openser:/opt/rtpproxy-1.2.1/bin# ps uax | grep rtpproxy rtpproxy 30827 0.0 0.0 26968 964 ? Ssl 14:49 0:00 /opt/rtpproxy-1.2.1/bin/rtpproxy -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.34.1 -s udp:127.0.0.1 7890 -r /etc/openser/stored_conversations -S /tmp -a -d DBUG -P root 31547 0.0 0.0 7524 892 pts/1 R+ 15:59 0:00 grep rtpproxy Output from netstat root@openser:/opt/rtpproxy-1.2.1/bin# netstat -tupna | grep rtpproxy udp 0 0 127.0.0.1:7890 0.0.0.0:* 30827/rtpproxy Relevant parts of the configuration file ... # ------ nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@openser.org") modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7890") modparam("nathelper", "force_socket", "udp:localhost:7890") ... route[6] { xlog("L_INFO", "ROUTE 6"); # NAT handling # Set reply routing block, to which control is passed after a transaction # completed with a negative result but before sending a final reply t_on_failure("1"); # Check if we're NATed if (isflagset(5) || isbflagset(6)) { # Use rtpproxy force_rtp_proxy(); if ( start_recording() ) { xlog("L_INFO", "RECORDING..."); }; } # Set reply routing block, to which control is passed each time a reply # (provisional or final) for the transaction is received t_on_reply("1"); } ... onreply_route[1] { xlog("L_INFO", "ONREPLY_ROUTE"); if ((isflagset(5) || isbflagset(6)) && status=~"(180)|(183)|(2[0-9][0-9])") { xlog("L_INFO", "NAT'd TRANSACTION"); fix_nated_contact(); force_rtp_proxy(); if ( start_recording() ) { xlog("L_INFO", "RECORDING..."); }; } else if (nat_uac_test("1")) { fix_nated_contact(); } exit; } When I start OpenSER I see the following in syslog Mar 19 16:08:45 openser /usr/sbin/openser[31630]: NOTICE:core:main: version: openser 1.3.2-notls (x86_64/linux) Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:core:main: using 256 Mb shared memory Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:core:main: using 1 Mb private memory per process Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:sl:mod_init: Initializing StateLess engine Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:tm:mod_init: TM - initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:maxfwd:mod_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:usrloc:ul_init_locks: locks array size 512 Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:registrar:mod_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:textops:mod_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:xlog:mod_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:avpops:avpops_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:acc:mod_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:dialog:mod_init: Dialog module - initializing Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31632_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31632_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31632_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31632_1 1 " Mar 19 16:08:45 openser /usr/sbin/openser[31632]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31631_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31631_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31633_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31633_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31634_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31634_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31635_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31635_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31636_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31636_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31637_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31637_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31638_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31638_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31637_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31637_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31631_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31631_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31634_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31634_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31636_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31636_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31633_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31633_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31635_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31635_1 1 " Mar 19 16:08:45 openser /usr/sbin/openser[31631]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31634]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31637]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31633]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31636]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31635]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31638]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31639_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31639_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31638_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31638_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31640_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31640_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31642_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31642_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31643_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31643_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31644_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31644_0 20040107 " Mar 19 16:08:45 openser /usr/sbin/openser[31642]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31643]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31645_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31645_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31639_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31639_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31642_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31642_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31644_1 VF 20050322" Mar 19 16:08:45 openser /usr/sbin/openser[31644]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31644_1 1 " Mar 19 16:08:45 openser /usr/sbin/openser[31640]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31639]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31640_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31640_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31643_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31643_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31645_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31645_1 1 " Mar 19 16:08:45 openser /usr/sbin/openser[31645]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled But when I place the call no debug info from rtpproxy is being generated nor the RTP session is being saved to file. Any idea what the problem can be? Thanks in advance Héctor

3 18

[Kamailio-Users] Kamailio as proxy and Asterisk as registrar
by Vic Jolin 26 Mar '10

26 Mar '10

Hi all! Kamailio newbie here. Well my setup is like this Kamailio as load balancer and proxy -> Multiple clustered asterisk as registrars What I wanted to do was have all clients registered to asterisk so that most of the pbx stuff will be done thru it and I'm using Asterisk Realtime as well and doing A2billing at the same time so I was thinking if my setup will actually work. I have already configured kamailio to do load balancing and send to my asterisk servers and my xlite is able to register to my asterisk server, the problem I have right now is when I try to call, the call gets dropped. I believe it's something about asterisk not being able to send the correct INVITE or not being able to receive the ACK or something to that effect. Please help me, as I've said I'm just a newbie with kamailio. TIA!

1 0

[Kamailio-Users] Fwd: Kamailio as a proxy between Trunk and PBX.
by Espen Berg 26 Mar '10

26 Mar '10

We are planning a new voip system. The trunk is provided to us on a private 10.x.x.x/29 range, and therefore not available direct to our end customers. We plan to have some kind of SIP/RTP proxy in front of the trunk, and a PBX in front of this. The trunk doesn't have any authentication, the end customers should connect and authenticate to the PBX in front, and the RTP streams should terminate to the proxy. Is this possible to achieve by using Kamailio and RTPProxy? Trunk | | 10.x.x.x/29 | SIP Proxy | PBX Asterisk etc. | Customer I have tried many different configurations and browsed every bundled example and web, but still I am not able to do what I want. This have already given me some grey hears, so I guess it's time to check if this is even possible. If not, is there any other approach that is more recommendable? If someone has done this before I would appreciate some example configuration that I could work with. Regards Espen, Norway

2 2

[Serusers] change in mailing list management
by Pavel Kasparek 26 Mar '10

26 Mar '10

Hello all, since now it is possible to use only web interface to manage mailing lists hosted on lists.iptel.org server. We had to disable the email commands interface as part of fight against spam (it was causing too many messages being sent from us, recognized by some blacklists as spam). This means that for all actions like subscribing, un-subscribing, changing your preferences etc. you have to use the web interface. You can find a link to it at end of every message sent to the mailing list. Sorry if it causes any inconvenience to you. Pavel Kasparek

1 0

[Kamailio-Users] Kamailio 3.0.1
by dotnetdub 26 Mar '10

26 Mar '10

Hi List, I have Kamailio 3.0.1 running from GIT with my old 1.5.3 config file. Everything seems fine except the RPID. I'm not getting any RPID sent in the SIP message and if I put debug=3 I see: DEBUG: auth [rpid.c:177]: no rpid AVP My config: # auth modparam("auth", "rpid_avp", "$avp(s:rpid)") modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix","@asterisk.ie ;party=calling;id-type=subscriber;privacy=off;screen=yes>") # auth_db modparam("auth_db", "db_url", "mysql://openser:XSXXXX@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "$avp(s:rpid)=rpid") in my route: append_rpid_hf(); Probably missing something very simple. Any help would be appreciated. Thanks, Stephen.

2 13

route() call with pv argument
by jh＠tutpro.com 25 Mar '10

25 Mar '10

i just made a test and noticed that route() accepts a pv argument. at least i didn't get anhy syntax error from route("$var(test)"); this is good if it works. Core functions section of http://sip-router.org/wiki/cookbooks/core-cookbook/devel#route does not list route() function call at all. -- juha

2 6

[Kamailio-Users] fr_inv_timer not honored (fr_inv_timer/2 is used in fact)
by Iñaki Baz Castillo 25 Mar '10

25 Mar '10

Hi, I set fr_inv_timer=150 and use lcr module to balance (and failover) between two gateways. In the failure route I set (simplified): failure_route[FAILURE_ROUTE_OUT] { if (t_local_replied("last")) { next_gw(); t_relay(); } } This means that if a INVITE routed to gateway-1 replies a provisional response but no final response within 150 seconds, then Kamailio should generate a local 408 and failure route would relaty to gateway-2. But what I see is that such "timeout" time is 75 seconds (the half of 150 !). I remember a similar "issue" when handling with SRV DNS domains returning more than one destinations, but in my case I just use single IP's in lcr module. But the fact is that I use exactly 2 gateways in lcr (150/2 = 75). But anyhow it doesn't make sense. Do I miss something? Thanks. PD: Important, could occur the same with fr_timer? -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 11

route is not called and no error is produced
by jh＠tutpro.com 25 Mar '10

25 Mar '10

i tested if route() with pv arg would work. i created a route block: route [test] { xlog("here we are\n"); return; } and then as the very first thing in the main route block i have: $var(test) = "test"; route("$var(test)"); then i send the proxy a sip request and don't get anything related to this to syslog, i.e., no "here we are" line nor any error message. this looks like i bug to me, since if route() accepts a pv, i should see "here we are" syslog entry or if it doesn't, i should see some kind of error message. comments? -- juha

1 0

[Kamailio-Users] Establish Call but no audio transferred on different network
by RAJNIKANT VANZA 25 Mar '10

25 Mar '10

Hello all, I am using kamailio-1.5.0 + mediaproxy. it works properly when connection is in between two clients between same lan. But when I connecting between two client who are on differet proxy and behind different NAT, call has been established but no audio get transferred. What are the changed i suppose to do, or any parameters should get enabled while compilation, Thanks in advance. -- Best Regards, Rajnikant Vanza

2 1

[Kamailio-Users] Radius
by dotnetdub 25 Mar '10

25 Mar '10

Hi, I'm using Kamailio - posted this to the opensips list as I thought I may get support from the ag guys but nothing.. Has anyone else experienced this or have an idea as to what it may be? Having a very small but annoying issue with Radius Accounting. Everything is getting written fine except for 'Source-Port' This appears in radius.log everytime a record is written/updated. - Error: rlm_radutmp: NAS OpenSER port 5060 unknown packet type 15 It's writing this into the text files and value to the radacct table: Source-Port = "\304\023" All other information is being written correctly. CDRTool version: 7.0 Freeradius version: 1.1.7-4 mediaproxy version: 2.3.10 Kamailio Version: 1.5.3 Using the dictionary files that ship with CDRTool >From accounting: modparam("acc", "db_url", "mysql://openser:XXXXXXXXXXXX@localhost/openser") modparam("acc", "db_flag", 2) modparam("acc", "log_flag", 2) modparam("acc", "db_missed_flag", 3) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$avp(s:translated_uri); \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") Obviously it's not a show stopper but it's just bugging the hell out of me :) If any more info is required please let me know and I will provide ASAP. TIA, Stephen

2 2

[Kamailio-Users] Kamdbctl not installing and Install error
by Nathaniel L Keeling 24 Mar '10

24 Mar '10

I have installed Kamailio 3.0 but without db support. When I tried to add support for postgres, the install errors and the kamdbctl does not install in order to create the database tables. I am installing on Solaris 10 and have included the error from the install: Makefile.defs defs skipped gmake[1]: Entering directory `/usr/local/src/kamailio-3.0.0/modules_k/xlog' touch /usr/local/kamailio-3.0.0/lib/kamailio/modules_k/xlog.so ginstall -m 755 xlog.so /usr/local/kamailio-3.0.0/lib/kamailio/modules_k gmake[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/modules_k/xlog' # other configs /bin/sh: syntax error at line 1: `;' unexpected gmake: *** [install-cfg] Error 2 Thanks Nathaniel

4 7

why is while loop not terminated?
by jh＠tutpro.com 24 Mar '10

24 Mar '10

i'm in the process of enhancing enum mudule enum_query, so that suffic param can be given in a pseudo variable. before committing to git, i tested if the new version is working correctly, and for that purpose wrote this kind of route: route [PERFORM_ENUM_QUERIES] { xlog("L_INFO", "suffix_list = <$var(suffix_list)>\n"); $var(i) = 0; $var(domain) = $(var(domain_list){s.select,0,,}); while ($var(domain) != "") { xlog("L_INFO", "Query $var(i) for $var(domain)\n"); if (enum_query("$var(domain)")) { xlog("L_INFO", "Return success!\n"); return(1); }; $var(i) = $var(i) + 1; $var(domain) = $(var(domain_list){s.select,$var(i),,}); }; return(-1); } looks like the enhanced enum_query works as expected, but the while loop is not terminating by the return(1) statement (see below). i'm puzzled. have i done some stupid mistake that i cannot see? -- juha Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: suffix_list = <e164.test.fi.> Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: enum [enum.c:724]: Got suffix <e164.test.fi.> Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Return success! Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: ERROR: <core> [action.c:1144]: ERROR: runaway while (1342, 25): more then 100 loops

1 1

what is $name?
by jh＠tutpro.com 24 Mar '10

24 Mar '10

i made a typo in script and wrote $domain instead of $var(domain). then i started sip router and didn't get any error messages. what does $name stand for when $name is not any of the ones listed in SIP Router Pseudo-Variables document? -- juha

2 2

Re: [Kamailio-Users] Establish Call on different network.
by john david 24 Mar '10

24 Mar '10

Dear all, started rtpproxy, enabled nat in cfg file. but still there is only connection between two client from different network, unable to get audio if used kamailio server. something more that i should add, except rtpproxy and enabling nathelper from cfg file. or else is there any way to debug where it get stuck. Thanks in advance.

2 1

[Kamailio-Users] need help on xilte
by false 24 Mar '10

24 Mar '10

Hi all i use kamailio sip proxy, its version is: [root@localhost kamailio]# /opt/sbin/kamctl fifo version Server:: kamailio (3.0.1 (i386/linux)) i use 2 xlite softphones to register to sip proxy and register successfull and use xlite make call to another xlite softphone and get 488 Not Accept Here all sip proxy, xlite softphones in the same LAN please help to fix in the sip proxy that doesnt check the SDP Thank you Ha` and the debug log: DEBUG: tm [t_reply.c:1059]: ->>>>>>>>> T_code=100, new_code=488 DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=2 global id=2 T start=0xb60ae9f0 DEBUG: tm [t_lookup.c:1152]: DEBUG: t_check_msg: T already found! DEBUG: tm [t_reply.c:1549]: DEBUG: relay_reply: branch=0, save=0, relay=0 DEBUG: <core> [msg_translator.c:1876]: old size: 500, new size: 440 DEBUG: <core> [msg_translator.c:1894]: build_res_from_sip_res: copied siz ontent-Length: 0ull "SDP: Incompatible media format: no common codec."19-1--d87543-;rport=51020 "Tốt hơn, thoáng gọn hơn, nhanh hơn -Trải nghiệm Yahoo! Mail mới hôm nay! http://vn.mail.yahoo.com"

2 1

[Kamailio-Users] need help on ACC modules
by false 24 Mar '10

24 Mar '10

Hi all i use the kamailio version 3.0.1 and try to understand ACC module and found the tutorial http://www.voice-system.ro/docs/uac/ does it still work with kamailio 3.0.1, could you please present a working config with kamailio 3.0.1 How to work with the uac-create.sql, the sip account from VOIP provider username: x.x.x.x password: x.x.x.x outbound proxy: x.x.x.x domain name: sbc.net expired time : 3600s what info should i insert into database l_uuid: l_username: l_domain: r_username: r_domain: realm: auth_username: auth_password: auth_proxy: expires: when i start the sip server with acc module and got error and the config parameter: modparam("uac", "reg_db_url","mysql://openser:openserrw@localhost/openser") DEBUG: <core> [sr_module.c:617]: find_param_export: parameter <reg_db_url> not found in module <uac> ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <reg_db_url> not found in module <uac> ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <reg_db_url> not found in module <uac> Thank you Phần mềm chống virus và adware miễn phí đã có cùng Yahoo! Thanh Công Cụ. Tải về ngay! Tải ngay. http://vn.toolbar.yahoo.com/

2 1

[Kamailio-Users] Kamailio+asterisk
by Marcus Kvarsell 24 Mar '10

24 Mar '10

Hello everyone! If i want to have Kamailio as front-end and 3 asterisk as back-end. Sip-registration should be done by Kamailio, and usage of meet-me rooms by asterisk-servers which should be load-balanced by Kamailio. is it possible of creating outgoing call with Kamailio and when call is answered, realtime route this call dynamically based on some information coming from unixODBC source, to a meet-me room? any help appreciated!! Regards / Marcus

3 2

[Serusers] Check out my photos on Facebook
by Sarfaraz Chougule 24 Mar '10

24 Mar '10

Hi, I set up a Facebook profile where I can post my pictures, videos and events and I want to add you as a friend so you can see it. First, you need to join Facebook! Once you join, you can also create your own profile. Thanks, Sarfaraz To sign up for Facebook, follow the link below: http://www.facebook.com/p.php?i=520931674&k=54EUZ6Q3QT2BXJMFYDYVPVWZWUKJ64&r Already have an account? Add this email address to your account http://www.facebook.com/n/?merge_accounts.php&e=serusers%40iptel.org&c=71a7… was invited to join Facebook by Sarfaraz Chougule. If you do not wish to receive this type of email from Facebook in the future, please click on the link below to unsubscribe. http://www.facebook.com/o.php?k=679489&u=557896866&mid=213a1b7G2140d4a2G0G8 Facebook's offices are located at 1601 S. California Ave., Palo Alto, CA 94304.

1 0

[Kamailio-Users] Establish Call on different network.
by john david 24 Mar '10

24 Mar '10

Hello all, I am using kamailio server, it works properly when connection is in between two clients between same lan. But when I connecting between two client who are on different proxy and behind different NAT, call has been established but no audio or video get transferred. What are the changed i suppose to do, or any parameters should get enabled while compilation, Thanks in advance.

2 1

[Kamailio-Users] dinner in berlin, thursday, march 25, 19:00
by Daniel-Constantin Mierla 23 Mar '10

23 Mar '10

Hello, couple of developers from SIP and VoIP area are in Berlin this week and plan to meet for dinner and beer Thursday, March 25, 19:00 at the traditional place by now: Lemke Brauhaus Luisenplatz 1, 10585 Berlin Across the corner with Charlottenburg Castle http://www.brauhaus-lemke.com/index.php?area=4 If you are around, just pop up. You can write me if you need more details about how to get there. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] Shared memory
by Bruno Machado 23 Mar '10

23 Mar '10

I have some questions regarding the shared memory. I'm working with the version 1.5.2 and its environment is quite simple. It does'nt storage any informations in the memory, like dialplan, contacts of subscribers, carrierroute, LCR and etc. The only purpose of this server is being a redirect server. Some calls have several possible destinations, reaching easily 12 contacts. The config.h's constant MAX_BRANCHES is set in 12, the default number. Ok. Like I said, almost the requests have more than 12 possible contacts. When it happens, Kamailio logs these messages: /usr/local/kamailio-1.5.2/sbin/kamailio[26188]: ERROR:core:append_branch: max nr of branches exceeded /usr/local/kamailio-1.5.2/sbin/kamailio[26188]: ERROR:pv:pv_set_branch: append_branch action failed Ok. It is not a problem. It is an error but it does not stop the system. The SIP messages still flowing normally. The server works for several days, 24/7 without functional errors, receiving five hundred thousand requests per day . But, usually at the weekends and the clock tells 2:30 am, the company's cell phone rings bringing a message: 'the system is down'. Kamailio shows: /usr/local/kamailio-1.5.2/sbin/kamailio[25492]: ERROR:core:add_avp: no more shm mem /usr/local/kamailio-1.5.2/sbin/kamailio[25492]: ERROR:exec:exec_avp: unable to add avp Kamailio stills running but it is not possible to route any request. All the new incoming calls generates the same error: 'no more shm mem'. I changed the size of shared memory, running Kamailio with '-m 64', but the server stopped again after some days. I see the options: - increase MAX_BRANCHES - increase again 'm' parameter (but it will just add more some days in the running time) - reduce the maximum number of contacts to 11 of the script that returns them to Kamailio So, my doubts are: - is it possible to have a memory leak when the message "max nr of branches exceeded" is showed? - changing '-m parameter' to a higher value, like 256, could affect the performance? tks friends bruno machado ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

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[Kamailio-Users] Media Bridge tips and hints
by Uriel Rozenbaum 23 Mar '10

23 Mar '10

Hi gentlemen, I'm having a bit of a hard time developing a bridge for media sessions. I have a box with 2 interfaces; eth0 has the private IP p.r.i.v and eth1 has public IP p.u.b.l. When I receive a message in the private IP I try sending the call to a public proxy we have (p.u.b.2) using Carrierroute module but the source IP used is p.r.i.v. It is obvious that p.u.b.2 will not be able to route back messages. # U 192.168.200.11:5060 -> p.r.i.v:5060 INVITE sip:1568767139@p.r.i.v SIP/2.0. Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK5d94bc09. From: "60911000" <sip:60911000@p.r.i.v>;tag=as6d477d46. To: <sip:1568767139@p.r.i.v>. Contact: <sip:60911000@192.168.200.11 <sip%3A60911000(a)192.168.200.11>>. ... # U p.r.i.v:5060 -> p.u.b.2:5060 INVITE sip:1568767139@p.u.b.2 SIP/2.0. Record-Route: <sip:p.r.i.v;lr=on;ftag=as6d477d46>. Via: SIP/2.0/UDP p.r.i.v;branch=z9hG4bKf531.440788e1.0. Via: SIP/2.0/UDP 192.168.200.11:5060 ;rport=5060;received=192.168.200.11;branch=z9hG4bK5d94bc09. From: "60911000" <sip:60911000@p.r.i.v>;tag=as6d477d46. To: <sip:1568767139@p.r.i.v>. Contact: <sip:60911000@192.168.200.11 <sip%3A60911000(a)192.168.200.11>>. ... I try then to set listen=0.0.0.0 so kamailio will be binded to all IPs but then the Route headers and Via use this 0.0.0.0 # U 192.168.200.11:5060 -> p.r.i.v:5060 INVITE sip:1568767139@p.r.i.v SIP/2.0. Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK14a00df7. From: "60911000" <sip:60911000@p.r.i.v>;tag=as502405cf. To: <sip:1568767139@p.r.i.v>. Contact: <sip:60911000@192.168.200.11 <sip%3A60911000(a)192.168.200.11>>. ... # U p.u.b.l:5060 -> p.u.b.2:5060 INVITE sip:1568767139@p.u.b.2 SIP/2.0. Record-Route: <sip:0.0.0.0;lr=on;ftag=as502405cf>. Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK711b.8d28549.0. Via: SIP/2.0/UDP 192.168.200.11:5060 ;rport=5060;received=192.168.200.11;branch=z9hG4bK14a00df7. From: "60911000" <sip:60911000@p.r.i.v>;tag=as502405cf. To: <sip:1568767139@p.r.i.v>. Contact: <sip:60911000@192.168.200.11 <sip%3A60911000(a)192.168.200.11>>. ... In this case the destination UA cannot return the call because it does not have a valid IP to do so. Is there any way to make Kamailio act in a different way? Like letting me modify the IP it will use for Via and Record-Route headers. Be aware that I'm supposed to do some kind of mirror for calls coming from outside the network (origin is public and destination is private). I'm sure I'll be asking some more on this same thread as I test. Thanks in advance! Uriel

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[Kamailio-Users] VUC Podcast: The SIP Router Project
by Daniel-Constantin Mierla 22 Mar '10

22 Mar '10

Hello, the recording of the VUC session last Friday about SIP Router Project is now available online at: http://www.voipusersconference.org/2010/kamailio3/ Here you find some details about who was in the call: http://www.kamailio.org/w/2010/03/vuc-listen-the-sip-router-project-podcast/ Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] GSoC 2010 - Call for students
by Daniel-Constantin Mierla 22 Mar '10

22 Mar '10

Hello, if you are a student (or you know a student) interested in working with SIP Router over the summer, earning some money as well, please apply for GSoC Conferencing Presence support (or forward this email). The description of the project is done at: http://www.sip-communicator.org/index.php/GSOC2010/Kamailio4575 If you have technical details about the project please ask them on sr-dev(a)lists.sip-router.org. For GSoC related questions, please address to: gsoc(a)sip-communicator.dev.java.net. A good FAQ for applicants is available at: http://www.sip-communicator.org/index.php/GSOC2010/HowToApply Application must be done directly to the google site, link provided in the FAQ. I will be in charge of mentoring this particular project, helping you as much as possible to understand the current presence server architecture and the core API of sip router. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] Centos 5 kamailio.init-file
by Detlef Pilzecker 22 Mar '10

22 Mar '10

Hi, does someone have a kamailio.init file for Centos5? The .../pkg/kamailio/[debian, fedora, ...]/kamailio.init files don't work! Thank you very much! Detlef Pilzecker

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[Kamailio-Users] Quick RTPproxy question
by Uriel Rozenbaum 22 Mar '10

22 Mar '10

Hi guys, I have some easy doubt about nathelper functions using RTPProxy. I'm trying to bridge from an external IP to an internal IP. The start-line for rtpproxy is: rtpproxy -l PUBLIC_IP/PRIVATE_IP -s udp: 127.0.0.1:7999 -F It starts OK and I see it when kamailio starts. I'm going to use something like Daniel showed on some other mail: if(dst_ip==private) force_rtp_proxy("ocfaei"); else force_rtp_proxy("ocfaei"); if i have the invite from a public IP to someone in private on the request I'll run force_rtp_proxy("ocfaei"); then the reply will be from private to public... should I run "force_rtp_proxy("ocfaei");"? or should it be the same? As allways, thanks for your help. Kind regards, Uriel

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[Kamailio-Users] Monitoring Upstream Carrier Health
by Geoffrey Mina 22 Mar '10

22 Mar '10

Hello, I am wondering if anyone has a clever way to remotely monitor a Kamailio 1.5 server. I am not looking for the standard monitoring, what I am looking to achieve is catching situations where my upstream carrier is having problems. We have a certain level of 404, 500, 503 errors throughout the day which are not indicative of a major carrier problem. I want to be able to monitor the ratio of properly setup calls to failed setups - so I can know when a carrier is having issues and is responding with many 503 errors. Any push in the right direction would be greatly appreciated. thanks.

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[Serusers] ser auth_db registration problem
by Anand Kumar 19 Mar '10

19 Mar '10

Hi All, i m trying to register softphone on ser using modparam("auth_db", "db_url", "mysql://baaaa:dddd@xx.x.x.x/ser") modparam("auth_db", "use_domain", 0) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") if(!www_authorize("", "subscriber")) { www_challenge("", "1"); break; } and found the following log but phone is not register . please help me... -- 2(28254) find_mod_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so] 2(28254) get_connection(): Connection found in the pool 2(28254) DEBUG: register_fifo_cmd: new command (DB) registered 0(28252) DEBUG: init_mod_child (1): dispatcher 0(28252) DISPATCHER:init_child #1 / pid <28252> 0(28252) SIP Request: 0(28252) method: <REGISTER> 0(28252) uri: <sip:123.49.54.226> 0(28252) version: <SIP/2.0> 0(28252) parse_headers: flags=1 0(28252) Found param type 232, <branch> = <z9hG4bK-d8754z-3d07d32ccc286b5c-1---d8754z->; state=6 0(28252) Found param type 235, <rport> = <n/a>; state=17 0(28252) end of header reached, state=5 0(28252) parse_headers: Via found, flags=1 0(28252) parse_headers: this is the first via 0(28252) After parse_msg... 0(28252) preparing to run routing scripts... 0(28252) parse_headers: flags=128 0(28252) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(28252) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(28252) parse_headers: flags=256 0(28252) end of header reached, state=9 0(28252) DEBUG: get_hdr_field: <To> [39]; uri=[sip:123456@123.49.54.226<sip%3A123456(a)123.49.54.226> ] 0(28252) DEBUG: to body ["123456"<sip:123456@123.49.54.226<sip%3A123456(a)123.49.54.226> > ] 0(28252) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 0(28252) DEBUG: get_hdr_body : content_length=0 0(28252) found end of header 0(28252) find_first_route: No Route headers found 0(28252) loose_route: There is no Route HF 0(28252) grep_sock_info - checking if host==us: 13==13 && [123.49.54.226] == [123.49.54.226] 0(28252) grep_sock_info - checking if port 5060 matches port 80 0(28252) grep_sock_info - checking if host==us: 13==13 && [123.49.54.226] == [123.49.54.226] 0(28252) grep_sock_info - checking if port 5060 matches port 80 0(28252) check_self: host != me 0(28252) parse_headers: flags=-1 0(28252) DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff 0(28252) parse_headers: flags=-1 0(28252) parse_headers: flags=60 0(28252) t_lookup_request: start searching: hash=57920, isACK=0 0(28252) DEBUG: RFC3261 transaction matching failed 0(28252) DEBUG: t_lookup_request: no transaction found 0(28252) DEBUG: mk_proxy: doing DNS lookup... 0(28252) check_via_address(122.162.142.65, 192.168.1.2, 0) 0(28252) DEBUG: add_to_tail_of_timer[4]: 0x2acf403e64e0 0(28252) DEBUG: add_to_tail_of_timer[0]: 0x2acf403e6500 0(28252) SER: new transaction fwd'ed 0(28252) DEBUG:destroy_avp_list: destroying list (nil) 0(28252) receive_msg: cleaning up 1(28253) DEBUG: timer routine:4,tl=0x2acf403e64e0 next=(nil) 1(28253) DEBUG: retransmission_handler : request resending (t=0x2acf403e6308, REGISTER ... ) 1(28253) DEBUG: add_to_tail_of_timer[5]: 0x2acf403e64e0 1(28253) DEBUG: retransmission_handler : done 1(28253) DEBUG: timer routine:5,tl=0x2acf403e64e0 next=(nil) 1(28253) DEBUG: retransmission_handler : request resending (t=0x2acf403e6308, REGISTER ... ) 1(28253) DEBUG: add_to_tail_of_timer[6]: 0x2acf403e64e0 1(28253) DEBUG: retransmission_handler : done 1(28253) DEBUG: timer routine:6,tl=0x2acf403e64e0 next=(nil) 1(28253) DEBUG: retransmission_handler : request resending (t=0x2acf403e6308, REGISTER ... ) 1(28253) DEBUG: add_to_tail_of_timer[7]: 0x2acf403e64e0 1(28253) DEBUG: retransmission_handler : done 1(28253) DEBUG: timer routine:7,tl=0x2acf403e64e0 next=(nil) 1(28253) DEBUG: retransmission_handler : request resending (t=0x2acf403e6308, REGISTER ... ) 1(28253) DEBUG: add_to_tail_of_timer[7]: 0x2acf403e64e0 1(28253) DEBUG: retransmission_handler : done 1(28253) DEBUG: timer routine:7,tl=0x2acf403e64e0 next=(nil) 1(28253) DEBUG: retransmission_handler : request resending (t=0x2acf403e6308, REGISTER ... ) 1(28253) DEBUG: add_to_tail_of_timer[7]: 0x2acf403e64e0 1(28253) DEBUG: retransmission_handler : done Anand Dayal -- Anand Dayal

1 0

[Serusers] proxy health
by Sabatella, Michael 19 Mar '10

19 Mar '10

So I sent an email out a few weeks ago regarding the stateful stats in ser 0.9.6. Stateful Statistics Current: 6 (200 waiting) Total: 6387581 (0 local) Replied localy: 6869160 The number waiting is slowly going up. Can somebody point me to a developer of ser I can talk to regarding this? So far there is no affect to any of my users but I want to make sure this does not get out of hand. Thanks Mike ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

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[Kamailio-Users] Reminder: VUC Today, SIP Router Project
by Daniel-Constantin Mierla 19 Mar '10

19 Mar '10

Hello, a kind reminder for today's audio conference scheduled for 16:00GMT. Check the start time for your zone at: http://vuc.me/next After presenting the achievements so far within SIP Router, with what is new in 3.0.0 release and development version, you have an unique opportunity to ask questions to: - Andrei Pelinescu-Onciul, the creator of SIP Express Router (SER), the architect behind the core (transport layers, memory management, asynchronous processing, timers, etc), who will be also able to answer anything from project's history started in 2001 - Alex Balashov, Kamailio management team member, experienced consultant in building large SIP platforms - myself, as co-founder of Kamailio (OpenSER) You can join via irc on #vuc (note that some of us started to hang out on #sip-router) channel at irc.freenode.net. Dialing to VUC is possible via sip, skype, pstn, web page and more, see: http://vuc.me Five hours to go right now, hear you then! Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] Kamailio - SIP Router on GSoC 2010
by Daniel-Constantin Mierla 19 Mar '10

19 Mar '10

Hello, I have the pleasure to announce that our project has an accepted application for Google Summer of Code 2010. The proposed application was done within SIP Communicator Organization and is related to extending the presence server for conference calls notifications. See more: http://www.sip-communicator.org/index.php/GSOC2010/Kamailio4575 http://www.sip-communicator.org/index.php/Development/Gsoc2010 The friends from SEMS project will be part as well, with separate application, implementing the audio mixer part. More details will be published soon. Start thinking about good candidates (they must be students (college or university)), we need them to get the project accepted for implementation. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] uac_replace_from unexpected behavior
by Brandon Armstead 18 Mar '10

18 Mar '10

Hello, As always thank you ahead of time for your help and input! I am currently calling uac_replace_from("", "") in effort to "leave uri" and "toss away display name" Which does seem to work....... for the initial INVITE However upon receiving an ACK with an empty display, however "" <- quotations, it does not clear the display "" which is causing issues with one of my upstream vendors. Example / Scenario: From: "" <sip:uri@host> Expected Result upon uac_replace_from("",""): From: <sip:uri@host> Current Result: From: "" <sip:uri@host> As you can see it is not stripping the "" empty display quotes. Any thoughts / ideas / suggestions to get my desired affect? Sincerely, Brandon Armstead

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[Kamailio-Users] Rtpproxy/Kamailio modification to support high capacity encryption, transcoding
by Vikram Ragukumar 18 Mar '10

18 Mar '10

Hello, We're making initial modifications to rtpproxy to support high channel capacity transcoding and encryption. At this point we want to get some general idea of the scope of changes needed for rtpproxy and Kamailio... so we're starting with small steps. We've been studying rtpproxy source and our current thinking is to add a sub-structure to the existing rtpp_session structure (defined in rtpp_session.h). The new sub-structure would include: -encryption options (type, key length, salt size, type of key mgt protocol, etc) -encode / decode options (type, VAD/CNG, VIF size, etc) Any comments or advice on this approach appreciated. Not sure whether to start a separate thread, but also there is the issue of what changes are necessary to Kamailio to support sending updated commands to rtpproxy. Would modifying Nathelper alone be sufficient? Thanks and Regards, Vikram. PS : I'm posting on Kamailio's mailing list because it seems that both Kamailio and rtpproxy developers closely follow this list.

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[Kamailio-Users] SIP Router Project of VUC, March 19
by Daniel-Constantin Mierla 18 Mar '10

18 Mar '10

Hello, this Friday, March 19, late afternoon, the weekly VoIP User Conference is hosting a session about SIP Router project. My goals are to present the achievements so far within SIP Router projects, what is new in Kamailio 3.0 release and plans for the future. More details can be found at: http://www.kamailio.org/w/vuc-the-sip-router-project/ You can join the audio conference via sip, skype, pstn line or other several options presented on http://vuc.me site. There is a irc channel available for it: #vuc on irc.freenode.net. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] kamailio logo artwork
by Daniel-Constantin Mierla 18 Mar '10

18 Mar '10

Hello, courtesy of Elio Rojano (http://www.sinologic.net), very nice logo artworks are available now, using openser or sip-router thematics. You can browse at: http://www.asipto.com/gallery/v/ksr-artwork/ Available for download as well at: http://www.kamailio.org/pub/ksr-artwork/ If you are good at graphic design and have new ideas, please submit. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] Kamailio 3.0.0 Install Error
by Nathaniel L Keeling 17 Mar '10

17 Mar '10

I am trying to install the new release of Kamailio 3.0.0 on Solaris 10. When I perform the install, I am getting this error. Here is the command that I am entering for the install: make prefix=/usr/local/kamailio-3.0.0 INSTALL=install group_include="standard standard-dep postgres" CPU=ultrasparc install install mode make[2]: Entering directory `/usr/local/src/kamailio-3.0.0/lib/kcore' make[2]: Nothing to be done for `install-if-newer'. make[2]: Leaving directory `/usr/local/src/kamailio-3.0.0/lib/kcore' touch /usr/local/kamailio-3.0.0/lib/kamailio/modules_k/speeddial.so install -m 755 speeddial.so /usr/local/kamailio-3.0.0/lib/kamailio/modules_k make[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/modules_k/speeddial' mkdir -p /usr/local/kamailio-3.0.0/etc/kamailio/ # other configs /bin/sh: syntax error at line 1: `;' unexpected make: *** [install-cfg] Error 2 I am new to Kamailio and would appreciate any help. Thanks Nathaniel

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Re: [Kamailio-Users] Is Timestamp supported in kamailio?
by Iñaki Baz Castillo 17 Mar '10

17 Mar '10

2010/3/16 mustafa rifaee <mustafa.rifaee(a)gmail.com>: > Hi all, > I would like to asking you Hi, it's just better to keep the thread in the maillist. > about the Time-stamp header in SIP messages, and > why this header does not supported in most open SIP servers, Is it > recommended or not? > Is this fact true ?? I think The problem is that both UAS and UAC must have > the same system time to generate a useful result. So therefore both or all > SIP devices (think about an intermediate proxy) must have NTP activated. That's not true. The request can contain a header like: TimeStamp: 25 and the proxy/server replies in the 100 Trying: TimeStamp: 25 0.001 This means that it has token 0.001 seconds to the server in order to process the request and generate the 100 response. -- Iñaki Baz Castillo <ibc(a)aliax.net> -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] [OT] MSN presence flow
by Iñaki Baz Castillo 17 Mar '10

17 Mar '10

Hi, does somebody have a MSN protocol flow related to presence rules or buddies management? This is, I would like to know how MSN protocol imlements some tasks as: - Adding a buddy. - Blocking a buddy for presence. - Blocking a contact (not a buddy) for presence. Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] kamailio crashes
by Panagiotis Skoulikaritis 17 Mar '10

17 Mar '10

Hello Daniel I do have quite a few core files, please send me the gdb commands. Regards Panagiotis. Daniel-Constantin Mierla wrote: > Hello Panagiotis, > > On 03/16/2010 02:21 PM, Panagiotis Skoulikaritis wrote: >> Hello Daniel >> >> I will need time to recreate the problem, >> attached are the only traces I kept. >> on the amaze-4.cap the calls are originated by a softphone registered >> on the kamailio >> on the crash.cap the calls are originated from the "PSTN". >> >> we do account the $pd and the $pn >> >> I reply to you directly since i don't want to give the traces on the >> mailing list, I hope you will understand. > > I asked to be sent private, it is ok. > >> >> Also for workaround Alex took out the PDT module from the route and >> the kamailio do not crash anymore. > > Hmm, so you say pdt module is related? That is pretty small and old > module... > > Do you still have the core file? I can send you some gdb commands to > get more details out of it. > > Thanks, > Daniel > >> >> Regards >> >> Panagiotis >> >> >> Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> can you send me ngrep/pcap file with ip addresses so I can match >>> which 200ok is causing the problem (coming from B or coming from >>> Asterisk)? The backtrace shows ip while the sip trace is masked. >>> >>> Also, I would need a bit more info from the core file. Please keep >>> one around. The issue seems to be related to P-Asserted-Identity >>> header, but I couldn't find any such header in the sip trace you sent. >>> >>> Are you accounting the PAI header? >>> >>> Thanks, >>> Daniel >>> >>> On 03/10/2010 03:30 PM, Panagiotis Skoulikaritis wrote: >>>> Dear Marius >>>> >>>> The scenario is as follows: >>>> >>>> 1. A Call is placed by a sip subscriber "A" >>>> 2. kamailio forwards the call to the asterisk server >>>> 3. Asterisk plays an IVR message on the subscriber "A", creates a >>>> new call to a "virtual" number which is forwarded to the kamailio >>>> server, and plays an ivr to this leg as well when the call is >>>> answered, then it connects the two calls. >>>> 4. Kamailio translates the "virtual" number to the pstn number of >>>> subscriber B >>>> >>>> >>>> I have attached a picture of the above scenario. >>>> >>>> The modules that are loaded are: >>>> >>>> loadmodule "db_mysql.so" >>>> loadmodule "mi_fifo.so" >>>> loadmodule "mi_datagram.so" >>>> loadmodule "sl.so" >>>> loadmodule "tm.so" >>>> loadmodule "rr.so" >>>> loadmodule "pv.so" >>>> loadmodule "maxfwd.so" >>>> loadmodule "usrloc.so" >>>> loadmodule "registrar.so" >>>> loadmodule "textops.so" >>>> loadmodule "uri_db.so" >>>> loadmodule "siputils.so" >>>> loadmodule "xlog.so" >>>> loadmodule "acc.so" >>>> loadmodule "dispatcher.so" >>>> loadmodule "pdt.so" >>>> loadmodule "dialplan.so" >>>> loadmodule "siptrace.so" >>>> loadmodule "dialog.so" >>>> loadmodule "sqlops.so" >>>> loadmodule "userblacklist.so" >>>> loadmodule "htable.so" >>>> loadmodule "uac.so" >>>> >>>> >>>> The config that does all the routing is : >>>> >>>> route[10] { >>>> >>>> xlog("alx ------- This is Route 10 -------"); >>>> >>>> if($rU =~ "^.*%+") >>>> { >>>> xlog("alx ------- The number contains %23 "); >>>> $rU = $(rU{re.subst,/^(.*)%23(.*)/\1\2/}); >>>> #$rU = $(rU{s.unescape.user}); #It changes the %23 >>>> to # !! >>>> xlog("alx ------- The perl $rU ------- "); >>>> } >>>> >>>> if($rU =~ "^.*#+") >>>> { >>>> xlog("alx ------- The number contains #"); >>>> $rU = $(rU{re.subst,/^(.*)#(.*)/\1\2/}); >>>> #$rU = $(rU{s.unescape.user}); #It changes the %23 >>>> to # !! >>>> xlog("alx ------- The perl $rU ------- "); >>>> } >>>> >>>> if(prefix2domain("2", "0")) { >>>> >>>> $var(dial_grp) = $(rd{s.select,0,.}{s.int}); # >>>> Dialplan group prefix for routing >>>> $var(num_pr) = $(rd{s.select,1,.}{s.int}); # The >>>> number of digits that prefix has >>>> $var(num_translation) = >>>> $(rd{s.select,2,.}{s.int}); # Called number translation >>>> $avp(s:port_translation) = >>>> $(rd{s.select,3,.}{s.int}); # Port number translation >>>> #$var(test_var) = $(rd{s.select,4,.}{s.int}); # >>>> Future property >>>> >>>> $avp(s:cust_prefix) = $(rU{s.substr,0,$var(num_pr)}); >>>> $rU = $(rU{s.substr,$var(num_pr),0}); >>>> >>>> xlog("alx ------- The new rU is $rU and properties $rd >>>> -------"); >>>> >>>> if($var(num_translation) == 1) >>>> { >>>> if($sht(a=>$rU)!=null){ >>>> >>>> $rU = $sht(a=>$rU); >>>> xlog("alx ------- Translation Done. DST >>>> num=$rU ----------"); >>>> >>>> } else { >>>> xlog("alx ------- Translation NOT Done >>>> ----------"); >>>> } >>>> >>>> >>>> #xlog("alx ------- We have DST number >>>> translation for user fU $avp(s:frm_user_name) ----------"); >>>> #if(dp_translate("31", "$rU/$rU")) >>>> #{ >>>> # xlog("alx ------- Translation Done. >>>> DST num=$rU ----------"); >>>> #} else { >>>> # xlog("alx ------- Translation NOT >>>> Done ----------"); >>>> #} >>>> } >>>> >>>> if(dp_translate("$var(dial_grp)", "$rU/$rU")) >>>> { >>>> xlog("alx ------- The $rU and with >>>> attributes :$avp(s:dest) -------\n"); >>>> >>>> $var(i) = 0; >>>> while($(avp(s:dest){s.select,$var(i),.})!="#") >>>> { >>>> $avp(s:dstgrp) = >>>> $(avp(s:dest){s.select,$var(i),.}{s.int}); >>>> $var(i) = $var(i) + 1; >>>> xlog("alx ------- The >>>> avp(s:dstgrp)=$avp(s:dstgrp) var(i)=$var(i) -------"); >>>> } >>>> >>>> # backup the username so we can use >>>> different prefixes >>>> $avp(s:user) = $rU; >>>> >>>> # select destination from first group >>>> >>>> if(ds_select_domain("$avp(s:dstgrp)", "4")) >>>> { >>>> >>>> if($(ru{uri.param,prefix})!=null) >>>> { >>>> $ru >>>> = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; >>>> >>>> } else { >>>> $ru >>>> = "sip:" + $avp(s:user) + "@" + $rd; >>>> } >>>> } >>>> >>>> $avp(s:dstgrp) = null; >>>> xlog("alx ------- The final >>>> RURI is $ru ------- "); >>>> if($avp(s:port_translation) >>>> == 1) >>>> { >>>> rewriteport("5061"); >>>> } >>>> t_on_failure("3"); >>>> t_relay(); >>>> exit; >>>> >>>> } >>>> >>>> >>>> >>>> >>>> } >>>> >>>> >>>> } >>>> >>>> Attached is the trace >>>> >>>> Regards. >>>> >>>> P. >>>> >>>> marius zbihlei wrote: >>>>> Panagiotis Skoulikaritis wrote: >>>>>> Hello Daniel >>>>>> >>>>>> the kamailio version is 1.5.3 >>>>>> >>>>>> Regards >>>>>> >>>>>> P. >>>>> Hello, >>>>> >>>>> Can you give us more details like the sip message that generates >>>>> the coredump (or if every sip message received generates the >>>>> core), if your config does something more out of the >>>>> ordinary(let's say exotic). Can we reproduce it ? >>>>> >>>>> It would also be helpful if you specify the list of modules you >>>>> have loaded. >>>>> >>>>> Cheers, >>>>> Marius >>>>>> >>>>>> Daniel-Constantin Mierla wrote: >>>>>>> Hello, >>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (OpenSER) - Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> -- >>> Daniel-Constantin Mierla >>> Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 >>> * http://www.asipto.com/index.php/sip-router-masterclass/ >>> > > -- > Daniel-Constantin Mierla > Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 > * http://www.asipto.com/index.php/sip-router-masterclass/ >

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Re: [Kamailio-Users] Is Timestamp supported in kamailio?
by Iñaki Baz Castillo 16 Mar '10

16 Mar '10

2010/3/16 mustafa rifaee <mustafa.rifaee(a)gmail.com>: > Hi Castillo, > I would like to asking you about the Time-stamp header in SIP messages, and > why this header does not supported in most open SIP servers, Is it > recommended or not? > Is there any open source support the time stamp header. > > Is this fact true ?? I think The problem is that both UAS and UAC must have > the same system time to generate a useful result. So therefore both or all > SIP devices (think about an intermediate proxy) must have NTP activated. > > Please Help me; I think I *already* helped you as you *already* asked me directly. Please, keep the threads in the maillist, nobody likes to give private free support. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Is Timestamp supported in kamailio?
by mustafa rifaee 16 Mar '10

16 Mar '10

Hi all; I would like to ask you about the timestamp in kamailio, Is it supported or not ? if yes how i can enable it ? Please Help me; Thanks. Best Regards Mustafa Rifaee

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[Kamailio-Users] [dialog module] dialog is not terminated when no ACK is received
by Iñaki Baz Castillo 16 Mar '10

16 Mar '10

Hi, using Kamailio 1.5.4. I use dlg_manage() for an INVITE. 200 Ok is replied by the callee but the UAC doesn't send the ACK (due to a crash). The dialog remains in Kamailio dialog memory/table in state 3 and would expire after default_timeout (which usually is 3600 seconds or more, unsuitable for this case). Yes, it could occur that the proxy is not doing loose_routing, but in that case it doesn't make sense to use dialog module, so shouldn't dialog module expire dialogs in state 3 after ~32 seconds? I'm using profiles_with_value to limit the number of calls per user, so this issue is a bit important, as a UAC not sending the ACK for a 200 means one less available channel for this user during dialog module default_timeout. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] maintenance this afternoon
by Daniel-Constantin Mierla 15 Mar '10

15 Mar '10

Hello, some small maintenance tasks will be performed afternoon today to the server hosting the mailing lists and kamailio.org site, if you notice some downtime, try again a bit later. Thanks, Daniel

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[Kamailio-Users] kamailio crashes
by Panagiotis Skoulikaritis 14 Mar '10

14 Mar '10

Dear fellow kaimailio users We have a kamailio server which crashes. below is the backtrace from the core files any idea why the kamailio is crashing Regards Panagiotis core.29568 Mar 10 09:27 #0 fm_status (qm=0x73a040) at mem/f_malloc.c:609 #1 0x0000000000423d5c in sig_usr (signo=15) at main.c:563 #2 <signal handler called> #3 0x00000037e3cd4711 in __recvfrom_nocancel () from /lib64/libc.so.6 #4 0x00000000004790cc in udp_rcv_loop () at udp_server.c:408 #5 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 core.19350 Mar 10 09:05 (gdb) backtrace #0 free_to (tb=0x775c00) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad2432de100) at parser/hf.c:187 #2 0x00002ad23fe3e525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad2432dcf58, rpl=0x772d28, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad23fe47b46 in t_reply_matching (p_msg=0x772d28, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad23fe47fa2 in t_check (p_msg=0x772d28, param_branch=0x7ffff9c016bc) at t_lookup.c:964 #5 0x00002ad23fe58ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x772d28) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bK45f7.70f91294.0;received=77.247.97.11\r\nV", len=920, rcv_info=0x7ffff9c017a0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7ffff9c019b8) at main.c:774 (gdb) core.29567 Mar 10 09:27 gdb) backtrace #0 free_to (tb=0x776460) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2ad1805fa100) at parser/hf.c:187 #2 0x00002ad17d15a525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2ad1805f8f58, rpl=0x771920, code=<value optimized out>) at sip_msg.h:54 #3 0x00002ad17d163b46 in t_reply_matching (p_msg=0x771920, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002ad17d163fa2 in t_check (p_msg=0x771920, param_branch=0x7fff8bf6a8cc) at t_lookup.c:964 #5 0x00002ad17d174ac2 in reply_received (p_msg=0x73a040) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771920) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKddc9.fa58f7e.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bKb973f6a69c9bea270e9db867dd7cc90f\r\nRecord-Route: <si"..., len=919, rcv_info=0x7fff8bf6a9b0) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=3, argv=0x7fff8bf6abc8) at main.c:774 (gdb) core.5326 Mar 10 09:02 (gdb) backtrace #0 free_to (tb=0x772c48) at parser/parse_to.c:79 #1 0x000000000047fd42 in clean_hdr_field (hf=0x2b2578e22560) at parser/hf.c:187 #2 0x00002b257597b525 in run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2b2578e213b8, rpl=0x771b50, code=<value optimized out>) at sip_msg.h:54 #3 0x00002b2575984b46 in t_reply_matching (p_msg=0x771b50, p_branch=<value optimized out>) at t_lookup.c:888 #4 0x00002b2575984fa2 in t_check (p_msg=0x771b50, param_branch=0x7fff7cdc63ac) at t_lookup.c:964 #5 0x00002b2575995ac2 in reply_received (p_msg=0x772c48) at t_reply.c:1395 #6 0x000000000041eebc in forward_reply (msg=0x771b50) at forward.c:521 #7 0x0000000000445313 in receive_msg ( buf=0x718dc0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 77.247.97.11;branch=z9hG4bKec2f.dc3d97a7.0;received=77.247.97.11\r\nVia: SIP/2.0/UDP 213.170.194.47:5060;branch=z9hG4bK47703457194f5be415efc231f6b3e923\r\nRecord-Route: <s"..., len=918, rcv_info=0x7fff7cdc6490) at receive.c:212 #8 0x00000000004794ae in udp_rcv_loop () at udp_server.c:449 #9 0x000000000042760e in main (argc=5, argv=0x7fff7cdc66a8) at main.c:774 (gdb) quit

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[Kamailio-Users] Loadbalancer/outbound_proxy and ACK problem
by Pavel Miskov 13 Mar '10

13 Mar '10

Hello list, let me first show my scenario: UAC_A ---> LB ---> | | PROXY+REG #1 | or | PROXY+REG #2 UAC_B <--- LB <--- | LB cfg: modparam("path", "use_received", 1) loose_route(); #for REG+PROXY if(is_method("REGISTER")){ add_path_received(); ds_select_dst("1", "0"); fix_nated_register(); } else { record_route(); add_path(); ds_select_dst("1", "0"); fix_nated_register(); } PROXY+REG cfg: ------------- modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) INVITE if working fine until ACK. UAC_A sends ACK to LB "ACK sip:UAC_B@LB_IP:5678;" insted of "ACK sip:UAC_B@UAC_B_PUB_IP:". In "200 OK" which PROXY sends to LB Contact is rewritten with ip addres of LB "Contact: <sip:UAC_B@LB_IP:5678;" and this is why UAC_A sends wrong ACK. Obviously I miss something so if someone could point me where to look. Thanks Pavel U UAC_A_PUB_IP:31488 -> LB_IP:5678 INVITE sip:UAC_B@test.com SIP/2.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport..Max-Forwards: 70..Contact: <sip:UAC_A@UAC_A_PRIV_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID:ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U LB_IP:5678 -> UAC_A_PUB_IP:31488 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488;received=UACs_PUB_IP..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Server: Kamailio (1.5.1-tls (x86_64/linux))..Content-Length: 0.... # U LB_IP:5678 -> REG_PROXY_IP:5166 INVITE sip:UAC_B@test.com SIP/2.0..Record-Route: <sip:LB_IP:5678;lr=on>..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Max-Forwards: 69..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Path: <sip:LB_IP:5678;lr>.... # U REG_PROXY_IP:5166 -> LB_IP:5678 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Server: Kamailio (1.5.1-tls (x86_64/linux))..Content-Length: 0.... # U REG_PROXY_IP:5166 -> LB_IP:5678 INVITE sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d SIP/2.0..Record-Route: <sip:REG_PROXY_IP:5166;lr=on;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Via: SIP/2.0/UDP REG_PROXY_IP:5166;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Route: <sip:LB_IP:5678;lr;received="sip:UAC_B_PUB_IP:31468">..Max-Forwards: 68..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Path: <sip:LB_IP:5678;lr>.... # U LB_IP:5678 -> REG_PROXY_IP:5166 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP REG_PROXY_IP:5166;branch=z9hG4bKa1b1.c7ca19a7.0;rport=5166;received=REG_PROXY_IP..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Server: Kamailio (1.5.1-tls (x86_64/linux))..Content-Length: 0.... # U LB_IP:5678 -> UAC_B_PUB_IP:31468 INVITE sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d SIP/2.0..Record-Route: <sip:REG_PROXY_IP:5166;lr=on;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.56778065.0..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Max-Forwards: 67..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..Path: <sip:LB_IP:5678;lr>.... # U UAC_B_PUB_IP:31468 -> LB_IP:5678 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.56778065.0..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport =5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@UAC_B_PRIV_IP:31 468;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41.. Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U LB_IP:5678 -> REG_PROXY_IP:5166 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip: LB_IP:5678;lr=on>..Contact:<sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U REG_PROXY_IP:5166 -> LB_IP:5678 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To:"UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U LB_IP:5678 -> UAC_A_PUB_IP:31488 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE..User-Agent: X-Lite release 1100l stamp 47546..Content-Length: 0.... # U UAC_B_PUB_IP:31468 -> LB_IP:5678 SIP/2.0 200 OK..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.56778065.0..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@UAC_B_PRIV_IP:31468;r instance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U LB_IP:5678 -> REG_PROXY_IP:5166 SIP/2.0 200 OK..Via: SIP/2.0/UDP REG_PROXY_IP:5166;rport=5166;received=REG_PROXY_IP;branch=z9hG4bKa1b1.c7ca19a7.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact:<sip:UAC_B@UAC_B_PRIV_IP:31468;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U REG_PROXY_IP:5166 -> LB_IP:5678 SIP/2.0 200 OK..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U LB_IP:5678 -> UAC_A_PUB_IP:31488 SIP/2.0 200 OK..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-d33a97176d49f84b-1---d8754z-;rport=31488..Record-Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Record-Route: <sip:LB_IP:5678;lr=on>..Contact: <sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com >;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 INVITE.. # U UAC_A_PUB_IP:31488 -> LB_IP:5678 ACK sip:UAC_B@LB_IP:5678;rinstance=ed47096f240b6b6d SIP/2.0..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;branch=z9hG4bK-d8754z-0472a03b5d1aa106-1---d8754z-;rport..Max-Forwards: 70..Route: <sip:LB_IP:5678;lr>..Route: <sip:REG_PROXY_IP:5166;lr;nat=yes>..Contact: <sip:UAC_A@UAC_A_PRIV_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From: "UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 ACK.. # U LB_IP:5678 -> REG_PROXY_IP:5166 ACK sip:REG_PROXY_IP:5166;lr;nat=yes SIP/2.0..Via: SIP/2.0/UDP LB_IP:5678;branch=z9hG4bKa1b1.46778065.2..Via: SIP/2.0/UDP LB_IP:5678;rport=5678;received=LB_IP;branch=z9hG4bKa1b1.46778065.2..Via: SIP/2.0/UDP UAC_A_PRIV_IP:31488;received=UACs_PUB_IP;branch=z9hG4bK-d8754z-0472a03b5d1aa106-1---d8754z-;rport=31488..Max-Forwards: 68..Contact: <sip:UAC_A@UAC_A_PUB_IP:31488>..To: "UAC_B"<sip:UAC_B@test.com>;tag=2e79423e..From:"UAC_A"<sip:UAC_A@test.com>;tag=8c3b5c41..Call-ID: ZTliZWQ4 YTM5MjVkYjUxYTk3MmUxZmVhOGY5MzlmM2Q...CSeq: 2 ACK..

3 5

[Kamailio-Users] What is the best module for routing INVITE based on DID in R-URI?
by Geoffrey Mina 13 Mar '10

13 Mar '10

Hello, I am putting up a Kamailio server which will do nothing but route INVITE requests from my upstream carrier to individual offices on my side. The office locations will NOT be registered SIP UAs, but other Kamailio proxy servers. What I want to have is a database of DIDs associated with a forwarding IP:Port and/or SRV records. 5555551212 ==> 1.2.3.4:5060 5555551213 ==> 1.2.3.5:5060 etc. Any guidance on what the best approach to achieving my goal would be much appreciated. Thanks, Geoff

4 12

[Kamailio-Users] need help to use the Kamailio register to remote sip proxy
by Cucku Cucku 12 Mar '10

12 Mar '10

Hi all i found the Kamailio supports register to remote sip proxy: http://www.mail-archive.com/users@lists.kamailio.org/msg07451.html it mentions to use : SQL to create the mysql table is in utils/kamctl/mysql/uac-create.sql but i didnt find the sql script My sip proxy version : sercmd> core.version Server: kamailio (3.0.1 (i386/linux)) 679736 my util folder : [root@localhost kamailio-3.0.1]# ls utils/kamctl/mysql/ acc-create.sql domainpolicy-create.sql purple-create.sql alias_db-create.sql drouting-create.sql registrar-create.sql auth_db-create.sql group-create.sql rls-create.sql avpops-create.sql htable-create.sql siptrace-create.sql carrierroute-create.sql imc-create.sql speeddial-create.sql cpl-create.sql lcr-create.sql standard-create.sql dialog-create.sql msilo-create.sql uri_db-create.sql dialplan-create.sql pdt-create.sql userblacklist-create.sql dispatcher-create.sql permissions-create.sql usrloc-create.sql domain-create.sql presence-create.sql please help Thank you Muốn đổi mới cách xưng hô với ông xã/ bà xã của bạn? Xem có bao nhiêu cách xưng hô giữa vợ và chồng nhé! http://vn.answers.yahoo.com

3 3

[Kamailio-Users] Stateless failover
by Santiago Soares 12 Mar '10

12 Mar '10

We're using Kamailio as a stateless load balancer, and would like to add failover support. Is it possible to have failover support in stateless mode? What i want is that when a server doesn't answer to a request, Kamailio set its state to probing, and start to ping it until it answer, then setting its state to active again. And that only active servers are selected by ds_select_dst(). Is it possible? How can I do this? Thank you. Santiago

4 4

[Kamailio-Users] Advanced Call Scenario / One Way Audio
by Brandon Armstead 12 Mar '10

12 Mar '10

Hello, As always, thank you for all / any help and input you may provide in advance. Call Scenario: UA1 -> REGISTRAR-01 -> Kamailio-01 -> Asterisk (New Call-ID + Asterisk in Media Path) -> Kamailio-01 -> REGISTRAR-02 -> UA2 UA1 is behind NAT UA2 is behind NAT The purpose of this is when using a shared "USRLOC" database to simulate calls from "PSTN" to generate both legs of the call, i.e. incoming and outgoing, and also allow for easier / cleaner "traversal" This aids from scenario's happening where UA1 calls UA2 (while UA1 exists on P1 and UA2 exists on P2) this prevents P1 -> UA2, and forces P2 -> UA2 We determine that this is a call from P1 to P2 (internal call) and thus create this "bridge / interconnection" We are running into a problem it seems with one way audio, i.e. the CALLEE can hear the CALLER, however the CALLER CAN NOT hear the CALLEE. REGISTRAR-01 AND REGISTRAR-02 are both "proxying" RTP As well as the initial Asterisk in "the middle" SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Perhaps there are better ways to go about this, let me know if I am way off course, thank you! Sincerely, Brandon Armstead

3 2

[Kamailio-Users] IP Authentication
by Joao Gomes Pereira 12 Mar '10

12 Mar '10

Hello to all Is is possible to implement IP authentication in Kamailio? Does Kamailio has a manual to do that? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

5 8

basic sip-router installation tutorial
by Meftah Tayeb 12 Mar '10

12 Mar '10

hi all, please could anyone give me a simple step by step tutorial to setup sip-router in a debian OS with mysql and other things? from jit, no zip file... thanks

4 3

[Kamailio-Users] Handling 3XX messages
by Bruno Machado 11 Mar '10

11 Mar '10

Hi! I have a Kamailio 1.5.2 handling 3XX messages. The commands below are executed in a failure route. If the first contact is not reachable or even a negative answer is received, the algorithm will call a second failure route, executing the 't_next_contacts'. Following this logic, the server will try several endpoints. The first 200 OK received from one this endpoints will establish the call. If all the contacts are tried, the server will finish the session with sending an error to the source. get_redirects("15:15","Redirect"); if(!t_load_contacts()) { t_reply("500", "Server Internal Error - Cannot load contacts"); exit; }; if(!t_next_contacts()) t_reply("404", "Not found"); else t_relay(); Ok... I found just one problem. If the 3XX message brings 5 different contacts, (all them with different q value), Kamailio just try 4 of them. Contact: sip:20182#556250982222@aaa.aaa.aaa.aaa;q=0.015, sip:3441#556250982222@bbb.bbb.bbb.bbb;q=0.014, sip:3441#556250982222@ccc.ccc.ccc.ccc;q=0.013, sip:50014#556250982222@ddd.ddd.ddd.ddd;q=0.012, sip:777#556250982222@eee.eee.eee.eee;q=0.011 After the 300 message, the server sends a invite to the first contact. Seconds later, analysing the debug screen, I see: DBG:tm:t_next_contacts: next contact is <sip:3441#556250982222@bbb.bbb.bbb.bbb The INVITEs are sent to this IP. After some seconds the server calls the third contact: DBG:tm:t_next_contacts: next contact is <sip:3441#556250982222@ccc.ccc.ccc.ccc And so on until this, the last contact: DBG:tm:t_next_contacts: next contact is <sip:777#556250982222@eee.eee.eee.eee ERROR:core:add_avp: 0 ID or NULL NAME AVP ERROR:tm:t_next_contacts: setting of fr_inv_timer_avp failed The proxy sent INVITEs for all them, except the last one because this failure. The function 't_next_contact' returns an error value. Somebody has a hint about this? Regards, bruno machado ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

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[Kamailio-Users] Advanced Call Scenario / One Way Audio
by Brandon Armstead 11 Mar '10

11 Mar '10

**resending as I am not sure if this made it out the first time, I do not believe it did, if this is a duplicate -- my apologies.** Hello, As always, thank you for all / any help and input you may provide in advance. Call Scenario: UA1 -> REGISTRAR-01 -> Kamailio-01 -> Asterisk (New Call-ID + Asterisk in Media Path) -> Kamailio-01 -> REGISTRAR-02 -> UA2 UA1 is behind NAT UA2 is behind NAT The purpose of this is when using a shared "USRLOC" database to simulate calls from "PSTN" to generate both legs of the call, i.e. incoming and outgoing, and also allow for easier / cleaner "traversal" This aids from scenario's happening where UA1 calls UA2 (while UA1 exists on P1 and UA2 exists on P2) this prevents P1 -> UA2, and forces P2 -> UA2 We determine that this is a call from P1 to P2 (internal call) and thus create this "bridge / interconnection" We are running into a problem it seems with one way audio, i.e. the CALLEE can hear the CALLER, however the CALLER CAN NOT hear the CALLEE. REGISTRAR-01 AND REGISTRAR-02 are both "proxying" RTP As well as the initial Asterisk in "the middle" SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Perhaps there are better ways to go about this, let me know if I am way off course, thank you! Sincerely, Brandon Armstead

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[Kamailio-Users] 400 Bad Request
by Ernest Mavrel 10 Mar '10

10 Mar '10

3 2

[Kamailio-Users] cdr_id not registered in acc
by Anders 10 Mar '10

10 Mar '10

Hi, (Kamailio 1.5 + freeradius + cdrtool) After a call is finished, it's INVITE and BYE are registered just fine in the acc table, but the records in acc never receive a cdr_id, and (therefore) nothing is inserted the cdrs table. Ideas on where to start looking? Thanks, Anders

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[Kamailio-Users] DEFINE question
by Uriel Rozenbaum 10 Mar '10

10 Mar '10

Hi guys, The question's simple, is there any pre-processor command to DEFINE constants? Thanks! Uriel

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[Kamailio-Users] OpenSER+Asterisk+BLF
by Mark Sayer 10 Mar '10

10 Mar '10

We are needing to modify the configure of a currently operating OpenSER to properly relay the SUBSCRIBE and SIP-NOTIFY messages that are sent between Asterisk and a phone that supports BLF (like the Snom 300 or Yealink T26). Our setup includes an OpenSER 1.2 & Asterisk 1.4.17 in the same box. OpenSER performs all registration, authentication and NAT. Asterisk handles the media and the accounting. In a pure Asterisk environment a "hint" would be setup in the Asterisk extensions.conf file and the phone (UA) would SUBSCRIBE to that HINT. Once Asterisk has registered that UA to the HINT then it sends SIP-NOTIFY messages to the UA as the status of the channel changes (available, ringing, busy). Our current openser.cfg file makes no mention of either SUBSCRIBE or NOTIFY which is an obvious reason that my Asterisk installation never registers the UA to the HINT. Is anyone interested in getting paid to fix this for us (we're too stupid to do it ourselves) or to offer another solution for controlling BLF in this setup. Thanks, Mark

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[Kamailio-Users] small typo in rewriteport example on Kamailio Wiki
by Andreas Heise 09 Mar '10

09 Mar '10

Hello, there is a small typo in rewriteport example on Kamailio Wiki should be rewriteport("5070"); instead of rewriteport(5070); http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#rewriteport_s… regards, Andreas

2 1

[Kamailio-Users] Kamailio (OpenSER) v3.0.1 Released
by Daniel-Constantin Mierla 08 Mar '10

08 Mar '10

Hello, the first patch release for 3.0 series is out as version 3.0.1. It includes the fixes to issues discovered since 3.0.0. Database structure and configuration file compatibility are preserved so the upgrade from 3.0.0 is straightforward. Links and more details are available at: http://www.kamailio.org/w/2010/03/kamailio-v3-0-1-released/ Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] [OT] BYE sip:SIMPLE&XCAP@ietf.org SIP/1.0
by Iñaki Baz Castillo 08 Mar '10

08 Mar '10

Hi, if somebody is bored and has 5-10 minutes of spare time I suggest to read the mail I've sent to sip-implementors maillist, full of hate and fury: https://lists.cs.columbia.edu/pipermail/sip-implementors/2010- March/024529.html Enjoy. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Serusers] ser numbers going up
by Sabatella, Michael 08 Mar '10

08 Mar '10

Hi all, Running ser 0.9.6. I notice these numbers going up and never going back down: I am a bit concerened with the red text. Is there any reason why that does not come down to 0. Do I need to restart the ser process once a month or something around that timing? So far I have not heard of any issues with user call setup but I just want to remain proactive. Serctl moni outputs the following: Server: Sip EXpress router (0.9.6 (i386/linux)) Now: Mon Mar 8 10:05:18 2010 Up Since: Fri Jan 8 11:24:17 2010 Up time: 5092861 [sec] Transaction Statistics Current: 1 (176 waiting) Total: 5624219 (0 local) Replied localy: 6049354 Completion status 6xx: 0, 5xx: 121, 4xx: 308504, 3xx: 0,2xx: 5316846 Stateless Server Statistics 200: 13145729 202: 0 2xx: 0 300: 0 301: 0 302: 0 3xx: 0 400: 42 401: 0 403: 0 404: 350766 407: 16536998 408: 0 483: 0 4xx: 2 500: 1 5xx: 0 6xx: 0 xxx: 0 failures: 72 Michael Sabatella IPC Systems Network Services Engineer Office: 1-201-253-2280 Cell: 1-908-380-5321 @ IPCGlobalNetworks.com<http://www.ipcglobalnetworks.com/> ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

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[Kamailio-Users] Get all the values for dialogs with a specific profile
by Iñaki Baz Castillo 08 Mar '10

08 Mar '10

Hi, by running "kamctl fifo profile_get_size togw" I can see the number of dialog tagged as "togw". Also if I run "kamctl fifo profile_get_size togw 999888777" I can see the number of dialogs tagged as "togw" with value "999888777". Is there any way to display all current values for dialogs with a specific profile/tag (i.e. "togw")? The idea is building a simple script to display the current number of calls per user (having that the value applied to dialog profiles means the calling user). Thanks for any suggestion. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] new website look
by Daniel-Constantin Mierla 08 Mar '10

08 Mar '10

Hello, http://www.kamailio.org has a new look! The former site was built with a cms system that discontinued the development, providing no longer updates for newer technologies and nicer design. We hope the new design helps to locate project resources easier and makes site navigation more straightforward via comprehensive list of links in the right sidebar. Credits go to Elio Rojano for the logo artwork and to developers of GPL Wordpress theme Cordobo Green Park which was used as start for the new design. If you are skilled in web design and can build a more personalized look on top of wordpress, please contact us. Cheers, Daniel PS. Do not forget, if you are tomorrow in London, UK, consider attending Present and Future of SIP Routing to get latest update about the project: http://www.kamailio.org/w/present-and-future-of-sip-routing-2010-london/ -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

1 0

[Kamailio-Users] Dispatcher ds_list results in flag=P
by Geoffrey Mina 08 Mar '10

08 Mar '10

Hello, I have a weird issue with Dispatcher module. When I do a ds_list I get some of my destinations returned with flag=P. It appears that when they are in this state, they are not included in the list of destinations to send traffic to. I have tried restarting Kamailio, and these certain servers always return to flag=P. I have confirmed the servers are up and running fine. They are communicating without issue at the SIP and network level. Any idea what could cause this? Thanks,

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Re: [Kamailio-Users] Users Digest, Vol 57, Issue 85
by Zhe 07 Mar '10

07 Mar '10

Help. Please remove me from this mail list. Thanks 在2010-02-27 19:00:01，users-request@lists.kamailio.org 写道： >Send Users mailing list submissions to > users(a)lists.kamailio.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > >You can reach the person managing the list at > users-owner(a)lists.kamailio.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > > 1. remove parameter (Elgar Onel) > 2. Exotic case (but real) in which RtpProxy doesn't work > (I?aki Baz Castillo) > 3. Re: Exotic case (but real) in which RtpProxy doesn't work > (Daniel-Constantin Mierla) > 4. Re: Exotic case (but real) in which RtpProxy doesn't work > (I?aki Baz Castillo) > 5. Re: Exotic case (but real) in which RtpProxy doesn't work > (Daniel-Constantin Mierla) > 6. Re: Exotic case (but real) in which RtpProxy doesn't work > (I?aki Baz Castillo) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Fri, 26 Feb 2010 03:00:25 -0800 (PST) >From: Elgar Onel <elgaronel(a)yahoo.com> >Subject: [Kamailio-Users] remove parameter >To: users(a)lists.kamailio.org >Message-ID: <444271.31134.qm(a)web114207.mail.gq1.yahoo.com> >Content-Type: text/plain; charset="us-ascii" > >Dear friends, > >how to remove parameters from INVITE sip address? > >tia, e. > > > > >

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[Kamailio-Users] 3.0 Modules
by dotnetdub 07 Mar '10

07 Mar '10

Hi List, I see two modules in particular that I could really do with running on 1.5.3 - I have started testing 3.0 but will not put into production just yet. Call control and topoh Would it be possible to backport these to 1.53 ? Would there be much to it.?. Call Control in particular is something that is very useful for us. We will eventually move to 3.0 of course as it is amazing product with lots of really nice new features just been burned badly before by moving to something so new. TIA. Stephen.

2 1

[Kamailio-Users] 10 years sip express router (ser)
by Daniel-Constantin Mierla 05 Mar '10

05 Mar '10

Hello, yesterday at CeBIT I've run into Fraunhofer booth and a banner attracted my attention, photo at: http://www.asipto.com/gallery/v/cebit2010/P3030721.JPG.html?g2_imageViewsIn… According to the middle right bubble, 2000 was the first year FhG presented SER at CeBIT. Given that, means the project should celebrate (at least) 10 years these days?!? I was among FhG staffers at CeBIT 2002, presenting latest SER at that time. According to CVS repository on BerliOS, files such as Makefile and main.c are dated 2001. Officially, SER was GPL-ed and open sourced in 2002 (year when ser project was registered to berlios.de software forge). Andrei, more insights from non-open-source era? I got into the project in January 2002, would be interesting to trace the birthday... Cheers, Daniel

2 1

[Kamailio-Users] local 408 from fr_timer and fr_inv_timer
by Robert McGilvray 04 Mar '10

04 Mar '10

I'm looking for a way to distinquish between locally generated 408s triggered by the timers. My problem is that I'm using gateway failover in a failure route and on 408s it would retry phones multiple times if they didn't answer (no voicemail etc) - so fr_inv_timer was triggered after receiving a provisional reply. I need to failover in case a gateway doesn't respond with provisional (it's set at 5 seconds currently). Will this work, is there a cleaner solution? I haven't had the oppurtunity to test it so I may have some syntax errors. onreply_route { if (t_check_status("1[0-9][0-9]")) { xlog("L_NOTICE", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu TO $ru -> Provisional Response --> $T_reply_code\n"); setflag(FLAG_PR_RECEIVED); } } failure_route[ROUTE_DP_FAILURE] { if (t_local_replied("last") && t_check_status("408") && isflagset(FLAG_PR_RECEIVED)) { xlog("L_INFO", "KAM_HOST: ($ci) $pr $mi route[$rm][0] $fu TO $ru -> Local 408 after provisional response, giving up\n"); exit; } } Thanks! Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] Present and Future of SIP Routing 2010 London
by Daniel-Constantin Mierla 04 Mar '10

04 Mar '10

Hello, I am pleased to announce a dedicated event in London, Tuesday, March 9 afternoon, where to present the version 3.0 and latest developments in Kamailio and SIP Router projects. The event includes as well presentations from various players in the VoIP and SIP market, more details at: http://www.kamailio.org/mos/view/Present-and-Future-of-SIP-Routing-2010-Lon… It is a free event, the number of seats is limited, room being available courtesy of Sun Microsystems (address 55 King William St, London EC4), therefore the first come first serve policy is applied. You can also enroll for a presentation, just send a short description about the topic in the registration email. Hope to see many of you in London next week, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Serusers] Is this a SER Problem?
by Samson Luk 03 Mar '10

03 Mar '10

I use SPA3102 with a iptel.org VoIP a/c, whenever my Firewall external IP changed (SPA3102's WAN port only connected to the internal LAN with a static IP, router function not used) the iptel.org VoIP a/c will report registration failed. I have to remove the proxy domain field and re-insert it then reboot to fix the problem, just wonder if this is a problem on the SER side or on the SPA side. I am sorry if this is not a correct appropriate place to ask such a question.

1 0

Re: [Kamailio-Users] Kamailio and SIP <-> XMPP gateway
by Daniel-Constantin Mierla 03 Mar '10

03 Mar '10

On 03/02/2010 11:25 AM, Even André Fiskvik wrote: > Thanks for the answers Daniel. > > On 1. mars 2010, at 12.23, Daniel-Constantin Mierla wrote: > > >> Hello, >> >> On 02/28/2010 06:43 PM, Even André Fiskvik wrote: >> >>> Reading through various earlier discussion though, I've found an issue that I wonder wether has been solved or not. >>> In some mailing-lists I saw that only text/plain MESSAGEs was supported through the xmpp gateway, is this still true? >>> >>> >> yes and 'no' :-). Yes because nothing was done in this respect to the module itself. 'No' because now (kamailio 3.0) you can use transformations to get content of xml documents and if the body is xhtml. >> > That's good to know. > > >>> If that's the case, is there any way to get CounterPath eyeBeam/Bria to work in the decribed scenario? >>> As to my knowledge it can only send text/html ? >>> >>> >> You can build a new message with uac module or update the existing SIP message (content type, body), do msg_apply_changes(): >> http://kamailio.org/docs/modules/3.0.x/modules_k/textops.html#id2513318 >> >> and they use xmpp functions. >> > I did a "quick" hack to support relying the MESSAGE by just setting the content-type to text/plain for now to do some testing. > The error about unsupported content-type thus went away. > > >> You can use ngrep to sniff the network to see connection between kamailio (openser) and xmpp server as well as run kamailio in debug mode in order to get more verbose log messages for each operation. >> > I've tried sniffing traffic on the server, and it doesn't seem like kamailio tries to connect to the XMPP server at all :/ > Any clues what might be causing this? Did you run in debug mode? Can you post the log messages? > I've attached my kamailio config-file as well if anyone would take You sent the email only to me. Did you intend it for mailing list? I recommend to keep sending to mailing list apart of messages with private data. In this way many can reply and other can learn from discussion. > a look at it (the main config is in xmpp.cfg) and also a diagram showing how the components are realated to each other. > I will try to look over configs when I get some time -- because of a trip ahead I am pretty constrained today. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] preparing for kamailio 3.0.1
by Daniel-Constantin Mierla 03 Mar '10

03 Mar '10

Hello, I think it is time to prepare for first minor release (3.0.1) in 3.0 series. We are approaching two months since kamailio 3.0.0, period within several issues were reported and fixed. Please report anything you are aware is not currently working in 3.0 to sr-dev(a)lists.sip-router.org or bug tracker: http://sip-router.org/tracker/ As for date, end of this week or beginning of next week would be a good time? Other opinions? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Serusers] Need a way to substitute SIP Response Code 504 for something else
by Frank Durda IV 03 Mar '10

03 Mar '10

There is a commercial switch vendor who has recently decided to make two changes that combined are quite evil: 1, They decided to devivate from RFC 3398 and when a given call INVITE timer expire (any of the SS7 ones, plus the SIP ones, including any limit specified in the INVITE itself), instead of translating the ISUP Release Cause 102 into a SIP Response Code of 504 (Gateway Timeout), they send out a 503 (Service Unavailable) instead. (503 is commonly used in low-cost carriers and the commercial equipment they use to trigger a route-advance, where the call doesn't end but is relaunched via another slightly more expensive route/carrier who perhaps can get the call to the destination. Some of these low-cost outfits may retry the same call with five or six carriers before reluctantly sending to the local access tandem which is usually the most expensive choice, but almost certain to work. This probably isn't what the inventors of the 503 code had in mind, but that is how it is used. A minority use 404 the same way.) 2. The same vendor implemented an automatic "Address Reach" testing mechanism for outbound calls. This thing means that when sending a call to a distant switch, if a single or small number of calls are returned from that distant switch with a SIP Response Code of 504 from the destination, the sending switch will take that destination switch completely out of route ("blacklist") and not send any calls at all to that destination until the blacklist is manually reset. So a result that applies to one call is fatal to all subsequent calls. This also means that if you sent the INVITE Expires timer to a value less than the number of seconds the calling party is willing to let the called number ring (called party doesn't have voice mail or it doesn't pick up for several rings), you trigger a 504 and bingo, you have killed the route between those two switches. So your own choices for how long to wait on ring-back for a call dictate how quickly it could kill the entire route for that call. Now, the two changes taken together are also somewhat convienent because this equipment maker just happened to make it impossible for their own equipment to ever return a 504 and do can never trip off their own booby-trap. So in a way what these two changes effectively create a not-our-hardware-detector, triggering route failures only when in contact with RFC-compliant equipment made by somebody else. I think that behavior is anti-competitive, and probably illegal in quite a few countries (notably the EU), but that is what exists at the moment. This creation started getting deployed in the past few months. So, to get around this nonsense, I need a way in SER to detect that a 504 has come in and change it to something else before sending it on. I won't quibble about the "proxies can't forward a 503", so I won't try that. I would like to turn the 504 into a 404, which is better than having route get pulled all over the place. Now, the last time (a year or so ago) I asked about trying to change a SIP Response Code in SER, the suggested ser.cfg "fix" created all sorts of strange warnings in the logs about timers expiring and I was told to not worry about that. Well, it seemed to cause a lot more problems than just warnings like memory leaks or cores caused by something, so I ended up having to hard-code the translation I wanted into the SER source code and not use rules in ser.cfg to do this. I'm writing now to (A) alert people about what this major brand switch maker is doing with regard to 504s so you will recognize it if you encounter it, but also (B) to see if maybe has a more elegant way to change SIP Responses via the ser.cfg file has emerged. The obvious things like subst subst_uri don't appear to work, obviously busily editing the non-existent INVITE method message and ignoring the reply message. Using a reply/drop combination to generate a new reply and kill the real one doesn't seem to work either. As I recall it ends up sending the response specified in the reply, and then passing the original response as well. "drop" apparently doesn't kill it completely and I suspect reply is really meant to be used only during handling of the method messages, and not the response messages. Suggestions appreciated.

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[Kamailio-Users] kam 1.5/asterisk 1.6 hints/blf?
by Robert McGilvray 02 Mar '10

02 Mar '10

Please excuse the noob questions, I'm not very familiar with presence. Users are registered to kamailio, authenticated against radius with LDAP backend. I'd like to setup some queues on asterisk for helpdesk call center but in order to do it correctly asterisk needs the device state (ringing, in use etc). I assume I'll need to use realtime ldap and setup device entries for all my phones, then use hints on the extensions as if they were local. If I understand correctly this will make asterisk subscribe to kamailio (I will use kams IP in host field), kam then has the responsibility of handling the subscribe/publish/notify stuff. What needs to be done on the kam side? Thanks! Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] Kamailio (1.5 rev5834) and/or rtpproxy gets frozen
by Iñaki Baz Castillo 01 Mar '10

01 Mar '10

Hi, I'm experimenting an annoying issue in a Debian Lenny 32 bits under a DELL 850 (2 cores). Same occured before with Debian Etch 32 and 64 bits in other host (also DELL 850). Kamailio (1.5 rev 5834) behaves as a load balancer in front of 2 PSTN gateways. There are ~200 calls and for all of them rtpproxy is applied. Sometimes, with no reason (not just in the moments of highest traffic) kamailio gets totally frozen, this is, it doesn't reply to SIP messages, neither relays them. Then Kamailio cannot be killed (just with -9). Sometimes it occurs when performing a fifo command (i.e. "kamctl fifo address_reload"). After killing it, kamailio cannot be started again. It starts and logs the usual logs (to syslog) but after "rtp proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled" there is no more logs. Then if I do a "ps aux | grep kamailio" I just see a single process, it's like the children fail when connecting to rtpproxy and die but the master process doesn't realize of it and remains alive showing no error at all. So I suspect from rtpproxy as even when kamailio was killed rtpproxy process consumes lot of CPU. I'll try with rtpproxy-1.2.1 (the latest stable version) as before I was using a old "truk" version (unfortunatelly I've no idea about how to know the CVS revision of a CVS working directory...). Anyhow I also suspect that the problem could be in the server itself, as sometimes the above problem occurs when reloading iptables and so on, very very strange, no a definitive cause or reason, very annoying. So nothing is clear for me. But I know that the problem occurs: - In two DELL 850, one with 32 and the other with 64 bits. - In both Debian Leeny and Etch. - With kamailio 1.5 rev 5834. - With rtpproxy rev XXX (how to get the last modification date in a CVS working dirrectory?) So, the fact is that I'm completely lost, no idea of what is happening. Any suggestion? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Project Update, London Meeting, March 9
by Daniel-Constantin Mierla 01 Mar '10

01 Mar '10

Hello, Tuesday, March 9, I am going to be in London and plan to organize a social networking event for folks around SIP/VoIP, Kamailio (OpenSER) & SIP Router. One is going to be the usual evening dinner/pub meeting, see pics from some past events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ I addition, if there are people interested, I would like to have a session in the afternoon (2-3 hours) to give an update about Kamailio (OpenSER) 3.0, present and future development of SIP Router. Time is rather short, but I encourage attendees to prepare short presentations as well, about their products and services, to have more information exchanged there. Let me know if you wish to attend such event in order to find a proper place to host it (drop me an email). Participation is free for everybody, at the dinner, the classic rule, everyone is going to pay for himself/herself. Looking forward to meeting many of you in London. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] Kamailio and SIP <-> XMPP gateway
by Even André Fiskvik 01 Mar '10

01 Mar '10

Hi. I've just started looking at kamailio, and I'm experimenting a bit with configuring a SIP <-> XMPP gateway with it using the pua_xmpp module. The configuration part hasn't beed straight-forward, but I think I'm starting to get somewhere. Both kamailio and XMPP/Jabber are new technologies for me serverside, so I don't have a great deal of experience with either. In short, what I'm trying to accomplish at this stage is: - Have CounterPath eyeBeam/Bria to register to my kamailio server - Have Aastra 6755i register to my kamailio server - Connect an XMPP client to my XMPP server - The CounterPath eyeBeam/Bria, Aastra and the XMPP client should all be able to see each others presence - The CounterPath eyeBeam/Bria should be able to IM other eyeBeams/Brias - The CounterPath eyeBeam/bria should be able to IM the XMPP clients and the other way around Reading through various earlier discussion though, I've found an issue that I wonder wether has been solved or not. In some mailing-lists I saw that only text/plain MESSAGEs was supported through the xmpp gateway, is this still true? If that's the case, is there any way to get CounterPath eyeBeam/Bria to work in the decribed scenario? As to my knowledge it can only send text/html ? Could anyone with a good knowledge of Kamailio and the various modules pua_xmpp, xmpp, pua, pua_bla comment on this kind of setup is at all possible with kamailio and it's current state? Is there any way I can verify that opensips connects to the XMPP server correctly? I tried connecting FreeSWITCH to my XMPP server (Openfire) in component mode, and it would show up connected in Openfire's administration GUI, however I can not see the same when I try to configure kamailio. If anyone has experience with this setup and have another XMPP server they have tested it with, I would also want to try to swap Openfire with that (given they also have some guidance on XMPP server configuration). Best regards, Even André Fiskvik

2 1

[Kamailio-Users] remove parameter
by Elgar Onel 26 Feb '10

26 Feb '10

Dear friends, how to remove parameters from INVITE sip address? tia, e.

1 0

[Kamailio-Users] ratelimit enhancements
by Daniel-Constantin Mierla 26 Feb '10

26 Feb '10

Hi Marius, since you did some updates to this module, I am opening for debate some needed enhancements I did during 3.0 testing phase and want to get opinions how to get in the code repo. Practically is a new module I named for now ratelimit2 and my last idea is to get it named pipelimit in the trunk. The reason for a new module are some major changes. The module uses the same algorithm but its core is overhaul. - definitions of pipes are loaded from database - there can be unlimited number of pipes - pipes are identified by string names - should be possible to reload pipes at runtime (iirc, not yet in) - new pipes can be added at runtime - functions accept variables to identify the pipe Since I never used queues from this module and haven't spent time to understand the concept behind, this functionality is completely missing. The old module might be good to keep in place, probably many people are using it in this form. So, proposals? What is the way to go on? Common code (algorithms) can be made lib at some point. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] tracking new features
by Daniel-Constantin Mierla 25 Feb '10

25 Feb '10

Hello, a reminder for developers to write a short note for each new feature either to NEWS file in source tree or to wiki page: http://sip-router.org/wiki/features/new-in-devel Makes life much easier when the next major release is going to happen and help people to spot quickly what new features are already available in devel branch, hopefully attracting some of them to go for it and test. Now the wiki page is updated with most of NEWS content for 3.1. Thanks, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

1 0

[Kamailio-Users] CeBIT 2010
by Daniel-Constantin Mierla 25 Feb '10

25 Feb '10

Hello, I am going to visit CeBIT 2010 in Hannover next week. Drop me an email if you are around and want to meet and chat about Kamailio, SIP Router projects, etc. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

1 0

[Kamailio-Users] Storing nonscalar types (lists)
by Alex Balashov 24 Feb '10

24 Feb '10

Are AVP arrays still the only way to create a list of scalars as of K 3.0.0? Or are there other, more graceful ways to accommodate this need now? AVPs are a very useful feature, though the syntax is not documented in 'avpops' or the cookbook except rather incidentally: $var(i) = 0; while(is_avp_set("$(avp(s:somename)[$var(i)])")) { xlog("L_INFO", "Value at index $var(i): $(avp(s:somename)][$var(i)])\n"); $var(i) = $var(i) + 1; } Is there a better way to handle this now, perhaps via some features imported from SER? Script variables ($var(...)) do not support subscripts judging by the cookbook; do they? Are there other options I am unaware of? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

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[Kamailio-Users] Logging of db_extra parameters + serial forking
by Alex Balashov 24 Feb '10

24 Feb '10

Here is a problem I am sure is not new (1.5.x): modparam("acc", "db_extra", "gateway_id=$avp(s:gateway_id)") route { ... $avp(s:gateway_id) = 13; $ru = "some.gw1"; t_on_failure("1"); if(!t_relay()) sl_reply_error(); } # Serial forking rollover route failure_route[1] { if(t_was_cancelled()) exit; $avp(s:gateway_id) = 14; $rd = "some.gw2"; append_branch(); t_relay(); } The problem is, when serial forking happens, the CDR rows in 'acc' still show a value of 13 for 'gateway_id'. This is obviously because the value is sampled at the time of the initial stateful relay processing and is not updated afterward. Any easy/elegant fix? I can think of several fixes that don't involve using 'acc', obviously, but I'd rather avoid that... -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

1 1

Re: [SR-Users] function calling in the config script with $var() as attribute(s)
by Klaus Feichtinger 24 Feb '10

24 Feb '10

Klaus, Alex, you're right. I've seen Klaus's suggestion only after writing my email message. This is the best solution and solves my problem! regards, Klaus > > > Am 24.02.2010 15:41, schrieb Klaus Feichtinger: > > Hello SR-users and -developers, > > > > I have the question if parameters for a function in the config script > > (e.g. "t_relay(host, port)") in general can be set with a variable > > ($var(xy)). Because of having a set of SIP servers / gateways > distributed > > over a wide area and therefore keeping the configuration as simple as > > possible, I would like to define a custom variable, assign it to a $var > > and use this variable as attribute for a relay function. In this case > the > > variable would represent a string of an IP-address (e.g. "10.1.1.55"). > > When this would work I only have to adapt the custom variables on top of > > each config script and could let the rest "as it is". > > > > > > First practical tests have shown, that the functions are NOT happy with > these "attributes". It can be used without any problems for if-else- > > statements, but I fear not for functions.... > > > > The practical scenario looks like: > > > > tc.ip11 = "10.16.65.101" desc ""; > > [...] > > route { > > $var(test) = $sel(cfg_get.tc.ip11); > > [...] > > if (!t_relay($var(test), "5061")) > > A workaround for t_relay: > > $du = "sip:"+$var(test)+":5061"; > t_relay(); > > > klaus > > > > [...] > > } > > > > This configuration resulted in following error messages: > > > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: syntax error > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 > > <core> [cfg.y:3326]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 413, column 38-41: Function parameter with integer > value not allowed#012 > > > > > > Please, could anybody clarify if this should work or not? If not - is > any > > alternative possibility available for "automating" function attributes > > instead of re-writing each function using the same string attribute? > > > > Thanks in advance, > > > > Klaus > > -- GMX DSL: Internet, Telefon und Entertainment für nur 19,99 EUR/mtl.! http://portal.gmx.net/de/go/dsl02

1 0

function calling in the config script with $var() as attribute(s)
by Klaus Feichtinger 24 Feb '10

24 Feb '10

Hello SR-users and -developers, I have the question if parameters for a function in the config script (e.g. "t_relay(host, port)") in general can be set with a variable ($var(xy)). Because of having a set of SIP servers / gateways distributed over a wide area and therefore keeping the configuration as simple as possible, I would like to define a custom variable, assign it to a $var and use this variable as attribute for a relay function. In this case the variable would represent a string of an IP-address (e.g. "10.1.1.55"). When this would work I only have to adapt the custom variables on top of each config script and could let the rest "as it is". First practical tests have shown, that the functions are NOT happy with these "attributes". It can be used without any problems for if-else- statements, but I fear not for functions.... The practical scenario looks like: tc.ip11 = "10.16.65.101" desc ""; [...] route { $var(test) = $sel(cfg_get.tc.ip11); [...] if (!t_relay($var(test), "5061")) [...] } This configuration resulted in following error messages: <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: syntax error <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 18-35: call params error#012 <core> [cfg.y:3326]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 413, column 38-41: Function parameter with integer value not allowed#012 Please, could anybody clarify if this should work or not? If not - is any alternative possibility available for "automating" function attributes instead of re-writing each function using the same string attribute? Thanks in advance, Klaus -- Sicherer, schneller und einfacher. Die aktuellen Internet-Browser - jetzt kostenlos herunterladen! http://portal.gmx.net/de/go/chbrowser

3 4

[Kamailio-Users] Kamailio 3.0.0: configuration request regarding VARIABLES usage (custom variables + $var(x))
by Klaus Feichtinger 24 Feb '10

24 Feb '10

Hello Kamailio-3 users, I have two questions regarding usage of variables within the config file (as published in the NEW FEATURES of Kamailio 3.0.0). The reason why I want to use variables instead of standard configuration lines is the maintainability of my SIP servers. I have a group of about 20 sip servers distributed over a wide area. Especially routing rules might change during operation of the servers and must be adapted on all servers. Therefore I want to use variables, defined at the beginning of each config file. Within the config file / the different routing blocks I want to reference to these variables. That would be much easier for automatic config script adaptation with a linux script (e.g.) than parsing the whole config file for special text phrases. QUESTION 1: ========== Is it impossible using a variable for requesting a SRC_IP address? With my current configuration I always see an entry in the SYSLOG that looks as follows: "<core> [route.c:1531]: BUG: comp_ip: invalid type for src_ip or dst_ip (20)" The configuration lines look like: $var(lst_gwIpRange) = "10.16.48.6[0-2]"; [...] if (src_ip=~$var(lst_gwIpRange)) { [...] } Is maybe a transformation missing/necessary to avoid the error message (which causes a malfunction)? Can anybody give me a hint? QUESTION 2: ========== The second problem I found is, that the config variables can be configured within a route block only and not - as expected - e.g. at the top of the config file (in the same way as the custom variables). I can't use the custom variables as they are, because I will use more complex variables, which are a mixture of up to three custom variables. These "complex" variables will be used in more than one routing block. Therefore I have to define these variables in every routing block...... Is an alternative (easier) solution with a single definition available, too? An example of my problem is as follows: # custom variables defined on top of the kamailio.cfg file tst.loc_ipaddr = "10.10.1.1" desc "IP addr of the local SIP Server" tst.loc_prefix = "11" desc "prefix of this SRV" tst.distRu1 = "51[0-9].*" desc "variable representing ID1" # the final variable should look like "sip:1151[0-9].*@10.10.1.1" # and is used in 3 routing blocks route[] $var(test) = "sip:" + $sel(cfg_get.tst.loc_prefix) + $sel(cfg_get.tst.distRu1) + "@" + $sel(cfg_get.tst.loc_ipaddr); [...] route[TEST1] $var(test) = "sip:" + $sel(cfg_get.tst.loc_prefix) + $sel(cfg_get.tst.distRu1) + "@" + $sel(cfg_get.tst.loc_ipaddr); [...] route[TEST2] $var(test) = "sip:" + $sel(cfg_get.tst.loc_prefix) + $sel(cfg_get.tst.distRu1) + "@" + $sel(cfg_get.tst.loc_ipaddr); [...] Thanks in advance for any hints! Regards, Klaus Feichtinger -- Sicherer, schneller und einfacher. Die aktuellen Internet-Browser - jetzt kostenlos herunterladen! http://portal.gmx.net/de/go/atbrowser

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[Kamailio-Users] 3.0.0 and snmpstats
by Kelvin Chua 24 Feb '10

24 Feb '10

hi guys, anybody here tried using snmpstats with 3.0.0? i keep on getting "No such Object" already verified that kamailio is talking to agentx master. net-snmp version 5.3.3 compiled from source also disabled openssl and md5 in net-snmp to prevent any undefined symbols been reading the mailing lists and so far nothing helped. seems like only a couple of guys are using this module? Kelvin Chua

2 4

[Kamailio-Users] Freeing sqlops results
by Alex Balashov 23 Feb '10

23 Feb '10

Should sql_result_free() always be called, no matter what the circumstance, in order to discharge memory allocated to query result sets and/or metadata? Or is it not necessary in certain situations, for example when script is broken via 'exit', i.e. sql_query("db", "SELECT ...", "res"); if(! $dbr(res=>rows)) { # Is it necessary to call sql_result_free() here? sl_send_reply("404", "Not Found"); exit; } Are results implicitly freed at the end of route script execution anyway? Are they tied to transaction persistence in any way? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

3 3

[Kamailio-Users] Forward calls from Asterisk to SIP provider via Kamailio for termination
by Slot Zero 23 Feb '10

23 Feb '10

Hello, I am a Kamailio noob :). I am trying to get Asterisk to forward calls to my SIP provider via Kamailio. The same machine is running Kamailio and Asterisk. I do not want to consume credentials as they have to be passed on all the way to my SIP provider. There is no NAT of any sorts. SIP Phone/Users connect to Asterisk.I do not need to authenticate when forwarding call from Asterisk to Kamailio as they are both running on the same server but I do need to make sure that Kamailio dials and forwards 011+number to be sent from local host port 5062(Asterisk listener) to SIP provider only. I have 6 Public IP addresses mapped on the server. I want to use the force_send_socket to allow me to change source IP of SIP requests when being sent to the SIP provider on the basis of credentials username in the request. I have pasted my config below. Please tell me what am I doing wrong here. In the kamctlrc file i have SIP_DOMAIN=localhost Thank you #------CONFIG BEGINS------------------ mpath="/lib/kamailio/modules_k/" debug=3 fork=yes children=4 auto_aliases=no alias=localhost alias=192.168.10.1 alias=192.168.10.2 alias=192.168.10.3 alias=192.168.10.4 alias=192.168.10.5 alias=192.168.10.6 disable_tcp=yes loadmodule "sl.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "/lib/kamailio/modules/tm.so" loadmodule "textops.so" modparam("rr", "enable_full_lr", 1) route { # Sanity Check # ------------ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if(msg:len>2048) { sl_send_reply("413", "message too large to be forwarded over UDP without fragmentation"); exit; }; # Record Route and NAT Preset # -------------------- if (method != "REGISTER") { record_route(); }; # Loose Route # ----------- if (loose_route()) { route(1); return; }; # Call Type Processing # -------------------- if (uri != myself) { route(1); return; }; if (uri == myself) { if (method == "BYE") { route(4); return; } else if (method == "CANCEL") { route(4); return; } else if (method == "INVITE") { route(3); return; } else if (method == "NOTIFY") { sl_send_reply("200", "Understood"); return; } else if (method == "OPTIONS") { sl_send_reply("200", "Got it"); return; } }; route(1); } # Default Message Handling # ----------------------- route[1] { t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } # INVITE Message Handling # ---------------------------------- # ---------------------------------- route[3] { if (uri =~ "^sip:011[0-9]@*") { rewritehostport("sip.voipprovider.com:5060"); if (search("^(Contact|m): .*user01*(a)(127\.0\.0\.1|localhost)")) { force_send_socket(192.168.0.2:5060); }; route(1); return; }; } # CANCEL and BYE Message Handling # ---------------------------------- route[4] { route(1); }

3 8

[Kamailio-Users] Rtpproxy hook
by Vikram Ragukumar 22 Feb '10

22 Feb '10

Hello, I am trying to encrypt/decrypt all packets belonging to an RTP stream. I am using Kamailio v1.5 with rtpproxy, where Kamailio forces the rtpstream through rtpproxy. Does rtpproxy provide a hook so that i can gain access to each RTP packet to perform encryption/decryption ? I do not want to end up having a patch for rtpproxy that i need to update whenever there is a new release of rtpproxy. -- Thanks and Regards, Vikram Ragukumar.

2 1

[Kamailio-Users] Error in Registration with TLS on
by Hemanshu Patel 22 Feb '10

22 Feb '10

Dear friends, Since last few days i am working on Kamailio with TLS support. I had followed each and every steps in installation docs...created certificates as well. Then i started testing the server with TLS on using SIPP. First i didnt added any certificate to SIPP, and Registration wasnt happening... When i added a certificate and key to SIPP....it started working fine....i was been able to test Registrations Successfully. Then i started working with one open source soft phone supporting TLS named mumble. IT Supports. Now i hadnt added any certificate to Mumblem. In my settings of kamailio i have set clietn_verify = 0 and require_client_certificate = 0. So without certificate as well i should be able to Authenticate my self successfully. Instead it gives following error in kamailio log: Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:print_ip: tcpconn_new: new tcp connection to: 172.16.16.218 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_new: on port 58125, type 3 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: entered: Creating a whole new ssl connection Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: looking up socket based TLS server domain [172.16.16.218:5091] Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_find_server_domain: socket based TLS server domain found Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: found socket based TLS server domain [172.16.16.218:5091] Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: Setting in ACCEPT mode (server) Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_add: hashes: 929, 1 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:handle_new_connect: new connection: 0x7fd6f4a58208 23 flags: 0002 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:send2child: to tcp child 0 0(3296), 0x7fd6f4a58208 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:handle_io: received n=8 con=0x7fd6f4a58208, fd=18 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_add: io_watch_add(0x73a0a0, 18, 2, 0x7fd6f4a58208), fd_no=1 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:tls_update_fd: New fd is 18 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: ERROR:core:tls_accept: SSL_accept failed: SSL_ERROR_SSL Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_del: io_watch_del (0x73a0a0, 18, -1, 0x10) fd_no=2 called Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: releasing con 0x7fd6f4a58208, state -2, fd=18, id=1 Feb 22 09:50:51 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: extra_data 0x7fd6f4a683a0 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:handle_tcp_child: reader response= 7fd6f4a58208, -2 from 0 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_destroy: destroying connection 0x7fd6f4a58208, flags 0002 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_close: closing SSL connection Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_update_fd: New fd is 23 Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_shutdown: shutdown successful Feb 22 09:50:51 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_clean: Cleanup function entered Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:print_ip: tcpconn_new: new tcp connection to: 172.16.16.218 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_new: on port 58126, type 3 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: entered: Creating a whole new ssl connection Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: looking up socket based TLS server domain [172.16.16.218:5091] Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_find_server_domain: socket based TLS server domain found Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: found socket based TLS server domain [172.16.16.218:5091] Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_init: Setting in ACCEPT mode (server) Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_add: hashes: 930, 2 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:handle_new_connect: new connection: 0x7fd6f4a58208 23 flags: 0002 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:send2child: to tcp child 0 0(3296), 0x7fd6f4a58208 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:handle_io: received n=8 con=0x7fd6f4a58208, fd=18 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_add: io_watch_add(0x73a0a0, 18, 2, 0x7fd6f4a58208), fd_no=1 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:tls_update_fd: New fd is 18 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: ERROR:core:tls_accept: SSL_accept failed: SSL_ERROR_SSL Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:io_watch_del: io_watch_del (0x73a0a0, 18, -1, 0x10) fd_no=2 called Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: releasing con 0x7fd6f4a58208, state -2, fd=18, id=2 Feb 22 09:51:01 localhost ./sbin/kamailio[3296]: DBG:core:release_tcpconn: extra_data 0x7fd6f4a683a0 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:handle_tcp_child: reader response= 7fd6f4a58208, -2 from 0 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tcpconn_destroy: destroying connection 0x7fd6f4a58208, flags 0002 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_close: closing SSL connection Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_update_fd: New fd is 23 Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_shutdown: shutdown successful Feb 22 09:51:01 localhost ./sbin/kamailio[3300]: DBG:core:tls_tcpconn_clean: Cleanup function entered And in Mumble soft phone log it gives me following Error: [9:50 AM] Welcome to Mumble. [9:50 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. [9:51 AM] Reconnecting. [9:51 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. [9:51 AM] Reconnecting. [9:51 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. [9:51 AM] Reconnecting. [9:51 AM] Server connection failed: Error during SSL handshake: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure. Can any one suggest what could be the problem? My Server works great with SIPP with TLS....so i dont think theres any config related error and i have set client_require_certificate = 0 thats for sure.... In real life scenario, hard or soft phones wont have certificates...so they should be able to connect to server and authenticate/Authorize themselves if server has valid certificate.But its not happening. So i need help from experienced guys.... -- Regards, Hemanshu Patel M: 09601295238

2 2

[Kamailio-Users] acc
by Anders 22 Feb '10

22 Feb '10

Hi, Probably a simpel fix some...not for me ;-) I get this here below in the log when I try to start Kamailio on a new installation. I suspect something is wrong with the acc module. What I read elsewhere is that it needs to recompile... - could that be, and if so, how would I recompile only a module? Thanks!! //Anders Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_config> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 140, column 25-26: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_flag> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 141, column 27-28: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_missed_flag> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 142, column 20-21: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:set_mod_param_regex: parameter <radius_extra> not found in module <acc> Feb 19 07:52:12 vn1031 kamailio: CRITICAL:core:yyerror: parse error in config file, line 157, column 2-3: Can't set module parameter Feb 19 07:52:12 vn1031 kamailio: ERROR:core:main: bad config file (4 errors)

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[Kamailio-Users] Demerits of using normal Authentication insted of radius or diameter auth
by Hemanshu Patel 21 Feb '10

21 Feb '10

Hello guys..... Recent now i had started taking keen interest in kamailio. I was testing kamailio, and i had test it under two scenario. first of all let me tell u about my configuration. I am testing on normal Desktop pc, installed kamailio,mysql, radius everything on same pc. I am using MYSQL anyway for both sort of testing. When i test with radius authentication, means using radius_www_authorize at max i am getting around 500 cps for Registration, but when i use normal www_authorize i am getting around 2000 CPS.... whopping difference guys..... So, i was wondering what is merits and demerits of not using Radius based authentication. Is it insecure if i use normal Authentication? Its not in my knowledge that radius do encryption...so i dont think by using radius it provides more security. What is your opinions about it? -- Regards, Hemanshu Patel M: 09601295238

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[Kamailio-Users] Username of P-Asserted-Identity header
by Marco Bungalski 19 Feb '10

19 Feb '10

Hi, all! Is there an easy possibility to get the Username of the P-Asserted-Identity header? When i use $ai i get for example "sip:+4942317769999@testdomain.de". I need only "+4942317769999" in a variable. I know for example that i can use $fU and get only the Username (and $fu gives me the whole From URI).... Cheers, Marco Marco Bungalski GmbH Traversale 5 27283 Verden Telefon: +49 4231 - 776 9999 Fax: +49 4231 - 776 9998 Mobil: +49 172 4204774 e-mail: Marco(a)Bungalski.de www.bungalski.de www.t-m-net.de Geschäftsführender Gesellschafter: Marco Bungalski Sitz der Gesellschaft: D-27283 Verden, AG Walsrode HRB 120586

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[Kamailio-Users] CANCEL goes bananas!
by Anders 19 Feb '10

19 Feb '10

Ok, "bananas" might not be the right word, but it resembles what is happening ;-) When I get a CANCEL from my customer to Kamailio, it is not passed on to my vendor - this is with a specific customer using VoipSwitch. The result is that the phone called keeps ringing even though it's been (attempted) cancelled, and a sideeffect is that it goes nuts at the carrier side because it can send anymore and returns 480 or 486. Below is the dialog from the Session Progress to when the call is cancelled - and then it goes bananas. Any ideas? Thanks a lot! U 2010/02/12 14:40:25.435793 MyKamailioIP:5060 -> MyCustomerIP:5060 SIP/2.0 183 Session Progress. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Contact: <sip:84370551184517324@MyVendorIP:5060>. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. CSeq: 1 INVITE. Content-Type: application/sdp. Via: SIP/2.0/UDP MyCustomerIP:5060;rport=5060;received=MyCustomerIP;branch=z9hG4bk120246101250121320412156. Content-Length: 210. . v=0. o=prxams02 0 0 IN IP4 MyVendorIP. s=sip call. c=IN IP4 62.93.140.13. t=0 0. m=audio 42972 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 2010/02/12 14:40:30.625842 MyCustomerIP:5060 -> MyKamailioIP:5060 CANCEL sip:890912551184517324@MyKamailioIP SIP/2.0. CSeq: 1 CANCEL. Via: SIP/2.0/UDP MyCustomerIP:5060;branch=z9hG4bk120246101250121320412156. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. Contact: <sip:MyCustomerIP:5060;transport=udp>. Content-Length: 0. . U 2010/02/12 14:41:25.496605 MyVendorIP:5060 -> MyKamailioIP:5060 SIP/2.0 480 Temporarily Unavailable. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Contact: <sip:84370551184517324@MyVendorIP:5060>. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. CSeq: 1 INVITE. Via: SIP/2.0/UDP MyKamailioIP;branch=z9hG4bKf18c.fa7b81d2.0. Via: SIP/2.0/UDP MyCustomerIP:5060;rport=5060;received=MyCustomerIP;branch=z9hG4bk120246101250121320412156. Content-Length: 0. . U 2010/02/12 14:41:25.496888 MyKamailioIP:5060 -> MyVendorIP:5060 ACK sip:84370551184517324@MyVendorIP:5060 SIP/2.0. Via: SIP/2.0/UDP MyKamailioIP;branch=z9hG4bKf18c.fa7b81d2.0. From: "69912390" <sip:001128650136@MyCustomerIP:5060>;tag=1202461012501320412156. Call-ID: GWV0iI5c5vUSEUnPBz@MyCustomerIP. To: <sip:890912551184517324@MyKamailioIP>;tag=3474974816-620042. CSeq: 1 ACK. Max-Forwards: 70. User-Agent: Kamailio. Content-Length: 0.

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[Kamailio-Users] Testing of Kamailio TLS with Sipp TLS
by Hemanshu Patel 19 Feb '10

19 Feb '10

Hello friendsm, I am testing TLS feature of both kamailio and Sipp. I first downloaded kamailio 1.5.x TLS supported version, uncommented TLS=1 from Makefiel and then build the kamailio. first i test kamailio without TLS with sipp for registration and everything works file. Then i follow "http://www.kamailio.org/docs/tls-devel.html" and creates rootCA, user certificates and all configuration parameters to kamailio.cfg file Config paras are as below: /* uncomment the following lines to enable TLS support (default off) */ disable_tls = no listen = tls:172.16.16.218:5091 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/data/hemanshu/install/kam-tls/etc/kamailio/tls/user/user-cert.pem" tls_private_key = "/data/hemanshu/install/kam-tls/etc/kamailio/tls/user/user-privkey.pem" tls_ca_list = "/data/hemanshu/install/kam-tls/etc/kamailio/tls/user/user-calist.pem" and restart kamailio server It works and i can see via netstat on port 5091. but when i starts sipp it gives me following error. [hemanshu@localhost sipp.3.1]$ ./sipp -sf ./data/rauth.xml -inf ./data/user.csv -r 1 -m 1 -trace_err -trace_stat -nd -fd 1 -i 172.16.16.218 172.16.16.218:5091 -t l1 2010-02-18 13:51:40:244 1266481300.244432: FI_init_ssl_context: SSL_CTX_use_certificate_file failed. I know i have built sipp with TLS support, then i can not figure out where is the problem. Sipp says user certification file failed ..but client doesnt need any certification file...or does it? i even tried with tls_verify_client = 1 , tls_require_client_certificate = 1, playing with different combinations...but still same ans. Can anyone suggest me what could be wrong? Have i made any mistake in configuring kamailio or theres some problem in SIPP. -- Regards, Hemanshu Patel M: 09601295238

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