Re: [OpenSIPS-Users] OpenSIPS libmysqlclient.so segfault after MySQL restart

2009-02-27 Thread Bogdan-Andrei Iancu
Hi Om, finally, after some struggle with the libmysqlclient, I hope the issue is solve. Please give it another try and let me know the result. Thanks and regards, Bogdan Om Bikram Thapa wrote: > Hi Bogdan, > > Just updated to latest svn, but I still see crashes after MySQL > restart. The syslog

Re: [OpenSIPS-Users] ERROR:db_mysql:db_mysql_do_prepared_query: mysql_stmt_execute() failed: (1) Unknown prepared statement handler (1) given to mysql_stmt_execute

2009-02-27 Thread Bogdan-Andrei Iancu
Hi John, Thanks for the info - it was a valuable one. I was fighting with some strange behaviour between different versions of libmysqlclient in reporting the error codes. Anyhow, a new (finally I hope) fix is available on SVN. Please update and let me know if it works for you or not. Regards

Re: [OpenSIPS-Users] Pickup of a ringing extension under OpenSIPS

2009-02-27 Thread Zahid Mehmood
Forgot to include the list before On Feb 26, 2009, at 2:42 PM, Yehavi Bourvine wrote: 2009/2/26 Zahid Mehmood i tested directed pickup and it worked fine in pure sip environment.. the only issues i had were with the cisco media gateway not working properly with REFER etc. Basically

Re: [OpenSIPS-Users] NOTIFY nat_keepalive - Bad Event

2009-02-27 Thread Carlo Dimaggio
Il giorno 27/feb/09, alle ore 15:01, Thomas Gelf ha scritto: > You can find this explanation also in the nat_traversal documentation: Thank you all, I have read "with attention" the nat traversal section :-) Now I have set the keepalive_method to "OPTIONS". Regards, Carlo Dimaggio _

Re: [OpenSIPS-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?

2009-02-27 Thread Antonio Reale
Sorry for the delay. I did not read your reply. Nice solution. I need to check that IOS based pbx can read a custom header. Regards. Antonio. > -Original Message- > From: users-boun...@lists.opensips.org [mailto:users- > boun...@lists.opensips.org] On Behalf Of IƱaki Baz Castillo > Sent:

Re: [OpenSIPS-Users] NOTIFY nat_keepalive - Bad Event

2009-02-27 Thread Thomas Gelf
There is nothing wrong, most phones just don't "understand" the "keep alive" event in your NOTIFY request - but that's fine, as you don't need a positive answer for the "keep-alive-effect", you just need a packet keeping the hole open. You can find this explanation also in the nat_traversal docume

Re: [OpenSIPS-Users] NOTIFY nat_keepalive - Bad Event

2009-02-27 Thread Bogdan-Andrei Iancu
Hi Carlo, I think the problem is with the phone as they do not understand the "keep-alive" event sent in NOTIFY. It is nothing you can do about this (in is not on the proxy side). Maybe in the future we can add a module parameter for removing the Event header from the REQUEST and make the dumm

Re: [OpenSIPS-Users] NOTIFY nat_keepalive - Bad Event

2009-02-27 Thread Dan Pascu
On Friday 27 February 2009, Carlo Dimaggio wrote: > Hi all, > > I'm using nat_traversal and the nat_keepalive function. For phones- > compatibility I have to use the "NOTIFY" method but all my registered > phones reply to opensips with "Bad Request / Bad Event" (with OPTIONS > method some phones re

[OpenSIPS-Users] NOTIFY nat_keepalive - Bad Event

2009-02-27 Thread Carlo Dimaggio
Hi all, I'm using nat_traversal and the nat_keepalive function. For phones- compatibility I have to use the "NOTIFY" method but all my registered phones reply to opensips with "Bad Request / Bad Event" (with OPTIONS method some phones reply with "200 OK"). What I am wrong? Can you help me?

Re: [OpenSIPS-Users] Force rtp proxy

2009-02-27 Thread michel freiha
Thanks Bogdan...I'll check this and get back to you Regards On Fri, Feb 27, 2009 at 1:34 PM, Bogdan-Andrei Iancu wrote: > Hi michel, > > Should do something like: > > if(!cr_route("default", "0", "$rU", "$rU", "call_id")){ > sl_send_reply("403", "Not allowed"); > } else { > # In cas

Re: [OpenSIPS-Users] Reg: help needed to get RLS module working

2009-02-27 Thread Anca Vamanu
Hi Visu, I tend to think that there is something wrong in your configuration. Can you please send me privately a message trace( with ngrep or wireshark) and the corresponding log for exactly the same scenario that your log captured - the Subscribe to the list and until the Subscribe that gets

Re: [OpenSIPS-Users] Using SST with Asterisk as a PSTN gateway

2009-02-27 Thread Bogdan-Andrei Iancu
Hi Robert, SST is available only in trunk Asterisk. What you can try is to use SST from OpenSIPS - see the SST module - http://www.opensips.org/html/docs/modules/1.4.x/sst.html Regards, Bogdan Robert Borz wrote: > I want to use sip session timers to ensure client sip-sessions are really >

Re: [OpenSIPS-Users] Force rtp proxy

2009-02-27 Thread Bogdan-Andrei Iancu
Hi michel, Should do something like: if(!cr_route("default", "0", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In cas of failure, re-route the request t_on_failure("1"); t_on_reply("1"); force_rtp_proxy(); t_relay

[OpenSIPS-Users] Using SST with Asterisk as a PSTN gateway

2009-02-27 Thread Robert Borz
I want to use sip session timers to ensure client sip-sessions are really active for billing purposes. We use Asterisk version 1.4 for pstn connectivity which does not support SSTs. So I want to use SSTs only on the client-side of a dialog. Would this be possible? Has anybody a setup like this?

Re: [OpenSIPS-Users] Force rtp proxy

2009-02-27 Thread michel freiha
Dear Bogdan, Do you mean doing something like that? if(!cr_route("default", "0", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In cas of failure, re-route the request t_on_failure("1"); force_rtp_proxy(); t_relay(); route(2) ;

Re: [OpenSIPS-Users] Presence benchmark

2009-02-27 Thread Bogdan-Andrei Iancu
Hi Romanov, not sure about IMS bench, but you could try with SIPP to generate Presence traffic. You simulate in a first step the PUBLISh traffic and to upload presence info, to see how the server reacts. In a second step, you can generate SUBSCRIBE . Regards, Bogdan Romanov Vladimir wrote: >

Re: [OpenSIPS-Users] Force rtp proxy

2009-02-27 Thread Bogdan-Andrei Iancu
Hi Michel, You have to call twice force_rtp_proxy() in order to complete the RTP session (and have RTP flowing). First for the INVITE (as you already do) and second for the 200 OK reply. So install a onreply_route and if the reply is 200 OK, call again force_rtp_proxy. Regards, Bogdan michel

[OpenSIPS-Users] Presence benchmark

2009-02-27 Thread Romanov Vladimir
Hi All! How I can test performance of presence server in OpenSips? We use IMS bench to test register, uac, message. -- Vladimir Romanov ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users