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37455 discussions

[Kamailio-Users] 302 redirect inside redirect
by Kent Pirlo 09 Sep '08

09 Sep '08

Ok, I am trying to get a redirect server working correctly, only cache is I need to hit another redirect server first, and grab info from the contact in the that 302 reply before processing and sending back a 300 to the original switch. call flow would be something like this: call comes into kamailio, 1) kamailio sends the invite to another proxy(1), 2) that proxy returns a 302,(if it times out it moves on to step 5) 3) kamailio sends an ACK back to that proxy(1), 4) kamailio looks at the contact header of the 302 reply, 5) depending on that info, I decide where to tell the originating switch where to go, 5) kamailio sends originating switch a 300 redirect with new contact header. 6) originating switch sends back ACK. I have tried doing this with a first step going to route(1) , which does a "t_on_reply" and "t_relay" to the second proxy. The onreply route, graps the contact saves it into variable and then sends to route(2), Route(2) looks at the variable, does some other lookups and sends t_reply("300","redirect"). The switch then sends the ACK, but kamailio says its not a local ACK and ignores it.. Then it keeps sending 300's back to the switch... Please help.. Thanks in Advance. kent

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[Kamailio-Users] Documentation Site Down
by Jon Farmer 09 Sep '08

09 Sep '08

Hi When I go to http://kamailio.org/docs/ i get a 404 error. Any ideas anyone? Regards Jon

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[Kamailio-Users] Forward call
by Thiago Rondon 09 Sep '08

09 Sep '08

Folks, My problem for forward call, that I need to change username/password and realm to do my forward. So, I try this: modparam("uac","from_restore_mode", "auto") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") modparam("uac","auth_username_avp","$avp(s:uac_username)") modparam("uac","auth_password_avp","$avp(s:uac_password)" route { ... if(uri=~"^sip:0[0-9]{10}@") { xlog("L_INFO","uac_atual: $avp(s:uac_username) !!"); $avp(s:uac_username)="username"; $avp(s:uac_realm)="1.1.1.1"; $avp(s:uac_password)="password"; xlog("L_INFO","uac_trocado $avp(s:uac_username) !!"); route(3); exit; } .. } ... route[3] { t_on_failure("3"); # reset flag to mark no authentication yet performed resetflag(7); # forward to PSTN uac_replace_from("username","sip:username@1.1.1.1"); rewritehostport("1.1.1.1:5060"); xlog("L_INFO","rewritehost !!!!!!!!"); t_relay(); } .. So, when I do xlog, in the first entry, uac_username don't display, why ? And its strange, because in one supplier its ok, and in another its not run. But I don't undestand why uac_username is empty in the first xlog! Thanks, -Thiago Rondon

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[Kamailio-Users] populate_leg_info
by David Villasmil 08 Sep '08

08 Sep '08

Anyone lnow what this means? dialog:populate_leg_info: bad sip message or missing Contact hdr ERROR:dialog:dlg_onreq: could not add further info to the dialog openser 1.3.2 thnks

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[Kamailio-Users] New OpenXCAP release 1.0.0
by Adrian Georgescu 08 Sep '08

08 Sep '08

Hello, There is a new release of OpenXCAP server available. It contains many bug fixes and additions for important features like XML partial manipulation, xcap-caps, xcap-diff and RLS services that were not available at the initial release last year. With this features OpenXCAP fulfills now most if not all relevant requirements for an XCAP server implementation. A generic xcap client library and command line client is available in the clients directory, you can use it to manipulate documents on OpenXCAP or other XCAP server implementation. The server software is also operational at http://sip2sip.info so you can test against it by registering a SIP account. Changes in version 1.0.0 * Added RLS services (RFC 4662 and RFC 4826) * Added support for xcap-diff based on draft-ietf-simple-xcap-diff-09 * Added partial get/put/delete of elements in the XCAP documents * Added test suite for rls-services, resource-lists and partial updates * Many bug fixes from field experiences * Development status from beta to production * Switched to Python 2.5 * Improved documentation and testing suite * Changed license to GPL * Fixed MySQL operational error (2006) The software can be downloaded from: http://openxcap.org Many thanks to Denis Bilenko who did most of the works for this new release. Kind regards, Adrian Georgescu

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[Kamailio-Users] CDRTool and Prepaid
by Adrian Georgescu 08 Sep '08

08 Sep '08

Hello Thanks for the feedback from everyone. You can try cdrtool version 6.6.0, I have added parallel prepaid call support. The algorithm is described in PREPAID.txt cdrtool (6.6.0) unstable; urgency=low * Allow simultaneous calls for prepaid accounts, see the Simultaneous prepaid calls section in doc/PREPAID.txt * Added pseudo code examples for call control engine * Apply the changes from setup/mysql/alter_tables.mysql * Added interface to kill ongoing sessions in Prepaid page Testing and feedback is useful and welcome. Regards, Adrian

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[Kamailio-Users] call forward problem
by muhammad akl 06 Sep '08

06 Sep '08

hey all after making call forward feature using avpops module and its exported functions avp_db_load and avp_pushto i have a problem : assuming all calls to extension 12001 will be forwarded to extension 12008, in opensers's database i have added the following : username=12001 domain =mydomain attribute = callfwd value = sip:12008@mydomain type = 0 untill that every thing was fine and all calls comes to 12001 forward to 12008 without any problem , the problem comes from when i am using 12008 to call 12001 it calls itself ( mean 12008) so anyone can tell me what i have missed ? i have configured my openser.cfg like this in loaded modules section added loadmodules "avpops.so" > in module parameter section added : modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") in Invite message handler added : if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(1); exit; } any suggestions ?

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[Kamailio-Users] two proxies sharing the same database
by toly 05 Sep '08

05 Sep '08

Greetings, I have 2 proxies having the same database on backend. One is the primary, second is standby - OpenSER 1.2.1. That's how I implement DNS based failover. What I see, when all UA's registered on primary and I start standby, I's starting to cleanup location table according to it's own timer. So record is gone, when primary trying to updated the record on re-register it of course fails, as a result location table is wrong. What will be a solution? Thanks, Toly -- View this message in context: http://www.nabble.com/two-proxies-sharing-the-same-database-tp19335185p1933… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] How to solve early media during serial forking
by Iñaki Baz Castillo 05 Sep '08

05 Sep '08

Hi, AFAIK there is an issue in typical SIP/PSTN enviroments in the case of failover using different gateways serially. The main issue is that SIP doesn't define how a UAC should handle the case in which it receives various "183 Session Progress" from different early-dialogs, each of them containing early-media. It occurs to me that a SIP/PSTN gateway sometimes replies 183 with early media for 1 or 2 seconds, and inmediatelly it replies a 502. Well, this 502 could be caused by other node in the PSTN network, it's not necessarialy a problem fo the gateway [1]. In this case I do serial forking forwarding the request to other gateway. But the UAC already received early media from the first gateway, and most of the SIP phones just render the first early media they receive in case of multiple 183, and drop others. So basically the problem is that the UAC user doesn't listen the ringing coming from PSTN after the request has been forwarded to the second gateway. Is there something I could do at proxy level to avoid this issue? Sincerely I don't think so, since it makes no sense to disallow the first 183 (I cannot know that the gateway will later reply a 502). Thanks. [1] Anyway, *who* is creating the early media? Why the PSTN generates it if the request doesn't arrive to the called? "imagine there is no PSTN..." -- Iñaki Baz Castillo

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[Kamailio-Users] custom routing
by toly 05 Sep '08

05 Sep '08

Greetings, Suppose I need to do custom routing for outbound calls from UA - softphone. Each user account may have the following info attached to it: 1:provider_a:2:provider_b:3:provider_c which means: forward call to provider_a, if failed forward call to provider_b if failed forward call to provider_c if failed send response back to UA I do database lookup based on from, and get above string. Suppose I wrote my own dispatcher module which initially gets set to above string: 1:provider_a:2:provider_b:3:provider_c and forwards to first provider, then on failure_route I call next_provider and so on. Will it work across multiple messages? Will it work correctly for each child - I mean joe@domain and paul@domain should be handled according their own providers? Any other suggestions? Thanks, Toly -- View this message in context: http://www.nabble.com/custom-routing-tp19322478p19322478.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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