[OpenSIPS-Users] global variable
hi, there is any way to have a global variable over dialogs and scrips, not depending by dialogs ? i need to check if a subscriber had calls in last amount of time(1 second, let's say) then don't engage it in next call. Working with databases, looks is not good enough because of very hight call rate. (updating, inserting into databse is show then call rate) Thanks, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips git revision: 76e9809 crash
Hi, I got a crash. Thanks, version: opensips 2.1.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT 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. git revision: 76e9809 main.c compiled on 21:32:22 Jul 5 2016 with gcc 4.9.2 #0 0x7fcf270f5779 in db_mysql_val2bind (v=v@entry=0x7fcf25d30ce0 <db_vals+480>, binds=binds@entry=0x7fcf28668d08, i=i@entry=15) at val.c:295 #1 0x7fcf270fbe1a in db_mysql_do_prepared_query (conn=conn@entry=0x7fcf28664e90, v=v@entry=0x7fcf25d30b00 , n=n@entry=17, uv=uv@entry=0x0, un=un@entry=0, query=0x7fcf27315980 ) at dbase.c:676 #2 0x7fcf27101508 in db_mysql_insert (_h=0x7fcf28664e90, _k=0x7fcf25d32180 , _v=0x7fcf25d30b00 , _n=17) at dbase.c:1265 #3 0x7fcf25af85fa in acc_db_request (rq=rq@entry=0x7fcf28219f40 , rpl=rpl@entry=0x7fcf286677e0, ins_list=ins_list@entry=0x7fcf25d334d8 , cdr_flag=2) at acc.c:638 #4 0x7fcf25b0713d in on_missed (code=, reply=0x7fcf286677e0, req=0x7fcf28219f40 , t=) at acc_logic.c:456 #5 tmcb_func (t=, type=, ps=) at acc_logic.c:685 #6 0x7fcf27ff4326 in run_trans_callbacks (type=type@entry=32, trans=trans@entry=0x7fcf22c7c1b0, req=req@entry=0x7fcf28219f40 , rpl=, code=) at t_hooks.c:209 #7 0x7fcf27faea86 in run_failure_handlers (t=0x7fcf22c7c1b0) at t_reply.c:569 #8 t_should_relay_response (reply=, cancel_bitmap=, should_relay=, should_store=, branch=, new_code=500, Trans=0x7fcf22c7c1b0) at t_reply.c:911 #9 relay_reply (t=0x7fcf22c7c1b0, p_msg=, branch=, msg_status=500, cancel_bitmap=) at t_reply.c:1125 #10 0x7fcf27fb2325 in reply_received (p_msg=0x7fcf286677e0) at t_reply.c:1505 #11 0x0047b585 in forward_reply (msg=msg@entry=0x7fcf286677e0) at forward.c:517 #12 0x0045d9bd in receive_msg ( buf=0x85f540 "SIP/2.0 500 Internal server error\r\nVia: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK7615.63e34a06.0\r\nFrom: \"XX\" <sip:xxx...@xxx.xxx.xxx.xxx>;tag=g8t85U635m04N\r\nTo: <sip:AA@AA"..., len=, rcv_info=rcv_info@entry=0x7ffc8eadc4e0) at receive.c:243 #13 0x005a04c5 in udp_read_req (si=, bytes_read=) at net/proto_udp/proto_udp.c:190 #14 0x0058bfbe in handle_io (fm=, fm=, fm=, idx=, event_type=2) at net/net_udp.c:260 #15 io_wait_loop_epoll (h=, t=, repeat=) at net/../io_wait_loop.h:190 #16 udp_rcv_loop (si=si@entry=0x7fcf28641f68) at net/net_udp.c:308 #17 0x0058db9c in udp_start_processes (chd_rank=chd_rank@entry=0x84c26c , startup_done=startup_done@entry=0x0) at net/net_udp.c:448 #18 0x0041a9d3 in main_loop () at main.c:722 #19 main (argc=, argv=) at main.c:1259 -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] little help with this bug
Hi, I need a little help with this bug: gdb /usr/local/unified_opensips/sbin/opensips /core GNU gdb (GDB) 7.4.1-debian Copyright (C) 2012 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "i486-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/local/unified_opensips/sbin/opensips...done. [New LWP 18111] [New LWP 18112] [New LWP 17585] warning: Can't read pathname for load map: Input/output error. [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/i386-linux-gnu/i686/cmov/libthread_db.so.1". Core was generated by `/usr/local/unified_opensips/sbin/opensips -f /usr/local/unified_opensips/etc/op'. Program terminated with signal 6, Aborted. #0 0xb77e2424 in __kernel_vsyscall () (gdb) bt #0 0xb77e2424 in __kernel_vsyscall () #1 0xb7468661 in raise () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 #2 0xb746ba92 in abort () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 #3 0x081404a7 in qm_free (qm=0xb7222008, p=p@entry=0xb729571c, file=file@entry=0x8203506 "parser/hf.c", func=func@entry=0x820366a "free_hdr_field_lst", line=line@entry=243) at mem/q_malloc.c:448 #4 0x08143a9e in free_hdr_field_lst (hf=0xb7295680, hf@entry=0xb729571c) at parser/hf.c:243 #5 0xb71c8d86 in mi_tm_uac_dlg (cmd_tree=0xb72942ec, param=0x0) at mi.c:539 #6 0xb6985fc5 in run_mi_cmd (param=0xb485c150, f=, t=0xb72942ec, cmd=) at ../../mi/mi.h:109 #7 default_method (env=0xb485c150, host=0xb485c270 "\001", methodName=, paramArray=0xa2b8880, serverInfo=0x0) at xr_server.c:223 #8 0xb68c2ac7 in xmlrpc_dispatchCall () from /usr/lib/libxmlrpc_server.so.3 #9 0xb68c2c16 in xmlrpc_registry_process_call2 () from /usr/lib/libxmlrpc_server.so.3 #10 0xb68c96e0 in ?? () from /usr/lib/libxmlrpc_server_abyss.so.3 #11 0xb68b7f57 in ?? () from /usr/lib/libxmlrpc_abyss.so.3 #12 0xb68b1e30 in ?? () from /usr/lib/libxmlrpc_abyss.so.3 #13 0xb68bac93 in ?? () from /usr/lib/libxmlrpc_abyss.so.3 #14 0xb6e0bc39 in start_thread () from /lib/i386-linux-gnu/i686/cmov/libpthread.so.0 #15 0xb7514c6e in clone () from /lib/i386-linux-gnu/i686/cmov/libc.so.6 (gdb) quit root@sp01:/# cd /usr/local/^C root@sp01:/# cd /home/openips/opensips_1_11/ root@sp01:/home/openips/opensips_1_11# git pull remote: Counting objects: 43, done. remote: Compressing objects: 100% (43/43), done. remote: Total 43 (delta 14), reused 0 (delta 0), pack-reused 0 Unpacking objects: 100% (43/43), done. >From https://github.com/OpenSIPS/opensips f880642..66ae29f 1.11 -> origin/1.11 1dfeb25..522a9e3 2.1-> origin/2.1 9ed154b..757389d 2.2-> origin/2.2 7e2d6e4..c63e14d master -> origin/master Updating f880642..66ae29f Fast-forward modules/sst/sst_handlers.c | 42 +- modules/sst/sst_handlers.h |1 + 2 files changed, 34 insertions(+), 9 deletions(-) root@sp01:/home/openips/opensips_1_11# Thanks, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips crash when imc_mi_list_rooms
Hi, My opensips crash when try to list imc rooms: opensipsctl fifo imc_list_rooms see the trace: (gdb) bt full #0 imc_mi_list_rooms (cmd_tree=0x0, param=0x0) at imc.c:714 i = optimized out len = 1 rpl_tree = 0xb71b49b0 rpl = 0xb71b49c0 node = 0xb71b49f4 attr = optimized out irp = 0xb49b3b94 p = optimized out #1 0xb70d60fc in run_mi_cmd (param=0x8b3c8a8, f=optimized out, t=0x0, cmd=optimized out) at ../../mi/mi.h:109 ret = optimized out #2 mi_fifo_server (fifo_stream=fifo_stream@entry=0x8b38378) at fifo_fnc.c:490 mi_cmd = optimized out mi_rpl = 0xb71a6a10 hdl = 0x0 line_len = 1 file_sep = optimized out command = optimized out file = optimized out f = 0xb71a6a10 reply_stream = 0x8b3c8a8 __FUNCTION__ = mi_fifo_server #3 0xb70d7601 in fifo_process (rank=0) at mi_fifo.c:213 fifo_stream = 0x8b38378 __FUNCTION__ = fifo_process #4 0x080ed8bf in start_module_procs () at sr_module.c:586 m = optimized out n = optimized out l = optimized out x = optimized out __FUNCTION__ = start_module_procs #5 0x0805df6d in main_loop () at main.c:865 i = optimized out pid = optimized out si = optimized out startup_done = 0x0 chd_rank = 0 rc = optimized out load_p = 0x0 #6 main (argc=5, argv=0xbfd244e4) at main.c:1634 cfg_log_stderr = optimized out cfg_stream = 0x8b24008 c = optimized out r = optimized out tmp = 0x5 Address 0x5 out of bounds tmp_len = optimized out port = optimized out proto = optimized out options = 0x81e78f8 f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o: ret = -1 seed = 1704724837 rfd = optimized out __FUNCTION__ = main Regards, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] USSD like over SIP signalling with opensips
Hi, If i need to send commands from client to opensips in order to setup some features, what should be the correct approach ? Let's suppose i want as SIP client, to setup call forward status (enable/disable) from client sending SIP message to opensips(having all the logic already implemented on proxy side). This service should be similar with dialing *123# from mobile or some ussd to do something. My question is, what should i use to send those commands, MESSAGE sip method with particular body, NOTIFY sip method with differnt event type, i mean i'd like to use sip protocol as transport for this kind of services. thanks, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] t_uac_dlg and tcp socket
Hi, It works like a charm! Thanks, Dani On Sat, Feb 28, 2015 at 2:25 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Dani, The socket you set may be ignore if not compatible from proto perspective with the protocol required by the SIP destination (RURI or DU). If the RURI requires UDP (has no ;transport=tcp in it) and the socket is TCP, the socket info will be discarded and a new sock will be searched. So, try to put the transport=tcp in your next hop URI. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 27.02.2015 23:18, Dani Popa wrote: Hi, t_uac_dlg with socket 'tcp:x.x.x.x' should work ? When i try to use t_uac_dlg with socket 'tcp:x.x.x.x' i see that the SIP message is sent over udp. Thanks -- Dani Popa ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] t_uac_dlg and tcp socket
Hi, t_uac_dlg with socket 'tcp:x.x.x.x' should work ? When i try to use t_uac_dlg with socket 'tcp:x.x.x.x' i see that the SIP message is sent over udp. Thanks -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] watchers and active_watchers
Hi, What's the reasons having watchers table and active_watchers table? As i understand, active_watchers is use for currently subscriptions presence and watchers is kind of history table ? Do i understand ok? , in watchers_table, if any presentity_uri change it's presence (presence status), each watcher_username should receive a NOTIFY, this is how i should understand this table ? Thanks, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated
Well, RTPproxy doesn't announced even till today that it can handle at least one call! Still waiting! On Thu, Mar 6, 2014 at 12:50 PM, H Yavari hyav...@rocketmail.com wrote: Hi, This means that we can't use mediaproxy for high load environments that CPS is high? What about RTPproxy? For using the maximum capacity of OpenSIPS(250k), what solution is best for media handling? Best Regards, H.Yavari -- Wait wait wait Can you please clarify.. does one simultaneous call mean one call, or two? Certainly this means two? On Tue, Mar 4, 2014 at 10:15 AM, a...@ag-projects.com wrote: http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MediaProxy scalability FAQ updated
truly, i saw it, it can handle it least one call! On Thu, Mar 6, 2014 at 12:40 AM, Adrian Georgescu a...@ag-projects.comwrote: Some people reportedly saw at least two calls, but they had a different sense of humor. -- Adrian On 04 Mar 2014, at 14:27, david da...@styleflare.com wrote: Mediaproxy can handle at least one simultaneous call, regardless of the hardware resources available providing no other program competes with the same resources on that machine. Bigger scalability can be achieved by adding more hardware. ??? Mediaproxy can handle at least one simultaneous call? On 3/4/14 11:15 AM, a...@ag-projects.com wrote: http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/Scalability Adrian ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Invite with Replaces header
Hi all, There is any way to check if Opensips instance have dialog in any state defined by Replaces Header of new incoming call ? -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] handle on reply for keepalive OPTIONS sip packet
There is any way to handle replay for sip keepalive OPTIONS packet when using nathelper module ? Thanks, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] handle on reply for keepalive OPTIONS sip packet
thanks, I played wrong with on_replay route. On Fri, Sep 13, 2013 at 2:19 PM, Aamir aamir_...@yahoo.com wrote: Hi Dani, You need to make a logic in opensips config to handle 200OK, cause in 200OK of OPTIONS the method is only OPTIONS and then you can handle it. Thanks Regards, Aamir Chougule Cell: 08097989101 Skype-ID: aamir_ryu --- Sent from my BlackBerry --- -Original Message- From: Dani Popa dani.p...@gmail.com Sender: users-boun...@lists.opensips.org Date: Fri, 13 Sep 2013 13:12:51 To: OpenSIPS users mailling listusers@lists.opensips.org Reply-To: OpenSIPS users mailling list users@lists.opensips.org Subject: [OpenSIPS-Users] handle on reply for keepalive OPTIONS sip packet ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips + asterisk 1.4
set opensips peer to insecure=port,invite On Wed, Jul 17, 2013 at 1:12 PM, Willian Mazzardo - SYSSVOIP will...@syssvoip.com.br wrote: Hi Stephens... how do I do this? Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Stephen Vigus svi...@gmail.com Hi Willian You most likely need to configure Asterisk to not authenticate SIP requests coming from Opensips. Regards Stephen On Wed, Jul 17, 2013 at 3:32 AM, Willian Mazzardo - SYSSVOIP will...@syssvoip.com.br wrote: Hi all.. I know this is a very simple scenario, all PSTN calls be routed to asterisk to do the billing job, but im having some problems, this is my scenario: Sip Client (10.0.0.3) Opensips (10.1.1.2) Asterisk (10.1.1.247) . PSTN Calls between sip clients on Opensips are working, but when I try to call over Asterisk, I have Proxy authentication problem. Here is my logs: Opensips: http://pastebin.com/SWpuRHku Asterisk: http://pastebin.com/6jp50LSS [opensips] host=10.1.1.2 type=friend context=callingcard qualify=no insecure=very fromdomain=10.1.1.2 Route: http://pastebin.com/mLgpXiNx Can someone help me on this? Thanks Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips + asterisk 1.4
what contex hit invite from opensips ? On Wed, Jul 17, 2013 at 1:24 PM, Willian Mazzardo - SYSSVOIP will...@syssvoip.com.br wrote: Hi Dani ... thanks ... i have for now insecure=very ... my asterisk version is 1.4... and this type of setting is for 1.6+ Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Dani Popa dani.p...@gmail.com set opensips peer to insecure=port,invite On Wed, Jul 17, 2013 at 1:12 PM, Willian Mazzardo - SYSSVOIP will...@syssvoip.com.br wrote: Hi Stephens... how do I do this? Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Stephen Vigus svi...@gmail.com Hi Willian You most likely need to configure Asterisk to not authenticate SIP requests coming from Opensips. Regards Stephen On Wed, Jul 17, 2013 at 3:32 AM, Willian Mazzardo - SYSSVOIP will...@syssvoip.com.br wrote: Hi all.. I know this is a very simple scenario, all PSTN calls be routed to asterisk to do the billing job, but im having some problems, this is my scenario: Sip Client (10.0.0.3) Opensips (10.1.1.2) Asterisk (10.1.1.247) . PSTN Calls between sip clients on Opensips are working, but when I try to call over Asterisk, I have Proxy authentication problem. Here is my logs: Opensips: http://pastebin.com/SWpuRHku Asterisk: http://pastebin.com/6jp50LSS [opensips] host=10.1.1.2 type=friend context=callingcard qualify=no insecure=very fromdomain=10.1.1.2 Route: http://pastebin.com/mLgpXiNx Can someone help me on this? Thanks Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips + asterisk 1.4
when you send a call in asterisk, do you see in asterisj cli that call hit you callingcard context or it hit default context ? On Wed, Jul 17, 2013 at 1:55 PM, Willian Mazzardo - SYSSVOIP will...@syssvoip.com.br wrote: My a2billing context [callingcard] exten = _X.,1,DeadAGI(a2billing.php) Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Dani Popa dani.p...@gmail.com what contex hit invite from opensips ? On Wed, Jul 17, 2013 at 1:24 PM, Willian Mazzardo - SYSSVOIP will...@syssvoip.com.br wrote: Hi Dani ... thanks ... i have for now insecure=very ... my asterisk version is 1.4... and this type of setting is for 1.6+ Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Dani Popa dani.p...@gmail.com set opensips peer to insecure=port,invite On Wed, Jul 17, 2013 at 1:12 PM, Willian Mazzardo - SYSSVOIP will...@syssvoip.com.br wrote: Hi Stephens... how do I do this? Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Stephen Vigus svi...@gmail.com Hi Willian You most likely need to configure Asterisk to not authenticate SIP requests coming from Opensips. Regards Stephen On Wed, Jul 17, 2013 at 3:32 AM, Willian Mazzardo - SYSSVOIP will...@syssvoip.com.br wrote: Hi all.. I know this is a very simple scenario, all PSTN calls be routed to asterisk to do the billing job, but im having some problems, this is my scenario: Sip Client (10.0.0.3) Opensips (10.1.1.2) Asterisk (10.1.1.247) . PSTN Calls between sip clients on Opensips are working, but when I try to call over Asterisk, I have Proxy authentication problem. Here is my logs: Opensips: http://pastebin.com/SWpuRHku Asterisk: http://pastebin.com/6jp50LSS [opensips] host=10.1.1.2 type=friend context=callingcard qualify=no insecure=very fromdomain=10.1.1.2 Route: http://pastebin.com/mLgpXiNx Can someone help me on this? Thanks Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] radius acc in local_route on dialog timeout
Hi,
Thanks for response, but for INVITE and BYE transaction, from called party
or calling party the things are right and i got radius requests. Also
db_requests in all cases, including when calls is disconnected by
Opensips are working fine.
Dani
On Mon, Jun 24, 2013 at 8:31 AM, qasimak...@gmail.com
qasimak...@gmail.comwrote:
Hi Dani,
You most probably don't have correct dictionary files placed. You can turn
debug=6 and it then see if you have any dictionary items missing. Every
time i install a new opensips with radius accounting i end up missing
dictionary file in one or more places and opensips does not show it to you
unless you have debug on.
Regards,
Qasim
On Thu, Jun 20, 2013 at 11:25 PM, Dani Popa dani.p...@gmail.com wrote:
any ideea ?
On Tue, Jun 18, 2013 at 7:10 PM, Dani Popa dani.p...@gmail.com wrote:
Hi all,
I use acc with radius and when i set accountig flag in local_route i
dont receive any accountig request on radius server. As I see local_route
was hit twice on dialog timeout and i dont understand when and how many
request should i receive on accounting if should i receive accounting
request. Or should i user radius_send_acc in this case.
this is my local_route
local_route {
xloglocal route);
if (is_method(BYE)) {
xlog(acc 1);
setflag(ACCOUNTING_FLAG);
#acc_db_request(200 Dialog Timeout, acc);
}
}
Thanks,
--
Dani Popa
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] radius acc in local_route on dialog timeout
Ok,
Thanks,
Dani
On Mon, Jun 24, 2013 at 5:36 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote:
**
Hi Dani,
Unfortunately the flag-based acc does not work for local route; the
flag-based acc is designed for proxies transactions, not for UAC-like
transactions (generated by OpenSIPS).
I suggest you to use the manual accounting in this case.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 06/18/2013 07:10 PM, Dani Popa wrote:
Hi all,
I use acc with radius and when i set accountig flag in local_route i
dont receive any accountig request on radius server. As I see local_route
was hit twice on dialog timeout and i dont understand when and how many
request should i receive on accounting if should i receive accounting
request. Or should i user radius_send_acc in this case.
this is my local_route
local_route {
xloglocal route);
if (is_method(BYE)) {
xlog(acc 1);
setflag(ACCOUNTING_FLAG);
#acc_db_request(200 Dialog Timeout, acc);
}
}
Thanks,
--
Dani Popa
___
Users mailing
listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] radius acc in local_route on dialog timeout
any ideea ?
On Tue, Jun 18, 2013 at 7:10 PM, Dani Popa dani.p...@gmail.com wrote:
Hi all,
I use acc with radius and when i set accountig flag in local_route i
dont receive any accountig request on radius server. As I see local_route
was hit twice on dialog timeout and i dont understand when and how many
request should i receive on accounting if should i receive accounting
request. Or should i user radius_send_acc in this case.
this is my local_route
local_route {
xloglocal route);
if (is_method(BYE)) {
xlog(acc 1);
setflag(ACCOUNTING_FLAG);
#acc_db_request(200 Dialog Timeout, acc);
}
}
Thanks,
--
Dani Popa
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] radius acc in local_route on dialog timeout
Hi all,
I use acc with radius and when i set accountig flag in local_route i dont
receive any accountig request on radius server. As I see local_route was
hit twice on dialog timeout and i dont understand when and how many
request should i receive on accounting if should i receive accounting
request. Or should i user radius_send_acc in this case.
this is my local_route
local_route {
xloglocal route);
if (is_method(BYE)) {
xlog(acc 1);
setflag(ACCOUNTING_FLAG);
#acc_db_request(200 Dialog Timeout, acc);
}
}
Thanks,
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] rtpproxy
Hi all, I know this is not Opensips question, but maybe some of you know the answer... I want to use rtpproxy as media relay but i need stream information at the end of call, information like: used codec, in/out packets, in/out IP traffic and if it's possible: RoundTripDelay, EarlyPackets, LatePackets, LostPackets. I know, some of you will recomand mediaproxy and it's not good for me, because i chosed to use rtpproxy because, i can insert and record media in curent stream. So the question is: there is any way to have such information at the end of call? Thanks, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] (no subject)
Hi again, Can how can insert from opensips, without any other pbx, b2b ..., 183 Progress with sdp. The ideea is, that sometime i need to insert in media stream at begining, some audio, using rtpproxy and for this i should have sdp and rtpproxy already used in call. As application, for example i want to change the ringback tone with opensips and rtpproxy from clasic ringing to some music or message. Thanks in advance for those who will tell me that opensips is sip proxy and has nothing to do with relayed media. the call trace case is next one: (A)invite -opensips-invite(B) (A)trying -opensips -trying(B) (A)ringing -opensips -ringing(B) (A)progress -opensips (A)200ok -opensips -200OK(B) (A) ACK -opensips -ACK(B) -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] insert 183 Progress with SDP in call dialog
Hi again, Can how can insert from opensips, without any other pbx, b2b ..., 183 Progress with sdp. The ideea is, that sometime i need to insert in media stream at begining, some audio, using rtpproxy and for this i should have sdp and rtpproxy already used in call. As application, for example i want to change the ringback tone with opensips and rtpproxy from clasic ringing to some music or message. Thanks in advance for those who will tell me that opensips is sip proxy and has nothing to do with relayed media. the call trace case is next one: (A)invite -opensips-invite(B) (A)trying -opensips -trying(B) (A)ringing -opensips -ringing(B) (A)progress -opensips (A)200ok -opensips -200OK(B) (A) ACK -opensips -ACK(B) -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] media stream extra info after hangup
Hi all, I know this is not Opensips question, but maybe some of you know the answer... I want to use rtpproxy as media relay but i need stream information at the end of call, information like: used codec, in/out packets, in/out IP traffic and if it's possible: RoundTripDelay, EarlyPackets, LatePackets, LostPackets. I know, some of you will recomand mediaproxy and it's not good for me, because i chosed to use rtpproxy because, i can insert and record media in curent stream. So the question is: there is any way to have such information at the end of call? Thanks, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Too many RFCs ????
:) nice On Mon, Apr 29, 2013 at 2:55 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: Trying to access : http://tools.ietf.org/html/**rfc5626http://tools.ietf.org/html/rfc5626 You get: % args) IOError: [Errno 28] No space left on device Maybe IETF has too many RFCs and DRAFTs.sorting them out by quality would be an option ;) Regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.**com http://www.opensips-solutions.com __**_ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-**bin/mailman/listinfo/usershttp://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.9 - Radius accounting
this should be enough http://cdrtool.ag-projects.com/projects/cdrtool/wiki/Installation_Guide Dani On Tue, Apr 9, 2013 at 11:59 AM, leo uzcud...@yahoo.it wrote: Hello. Were anyone able to setup opensips 1.9 with radius accounting? Unfortunately the tutorial is based on an old version of Opensips were the ACC module was different. Of course i can have accounting in the opensips database (acc table) but tcpdump-ing on radius ports i don't see any traffic (if i try with the radtest it works and i can see the traffic with tcpdump). Thanks, Leo. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-9-Radius-accounting-tp7585735.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.9 - Radius accounting
can you post your config ? There are 2 different things, aaa_radius and acc with aaa_url defined. With acc you just set account flag for acc event, but using aaa_radius you should use radius_send_acct. Dani On Wed, Apr 10, 2013 at 12:31 AM, Leonardo Uzcudun uzcud...@yahoo.itwrote: Hello: Activating opensips debug and grepping for radius i could find the following messages: Apr 9 23:10:52 sip-dev /usr/sbin/opensips[20471]: ERROR:aaa_radius:rad_avp_add: failure Apr 9 23:10:52 sip-dev /usr/sbin/opensips[20471]: ERROR:acc:acc_dlg_callback: Cannot create radius accounting Apr 9 23:10:52 sip-dev /usr/sbin/opensips[20472]: ERROR:aaa_radius:rad_avp_add: failure Some topics in the forum mention to check that the dictionary.opensips is included in radiusclient-ng dictionary, and it is. Any help on this? Opensips 1.9 Server OS Debian Squeeze Freeradius:2.1.10 Radiusclient-ng: 0.5.6 Thanks. -- *Da:* leo uzcud...@yahoo.it *A:* users@lists.opensips.org *Inviato:* Martedì 9 Aprile 2013 10:59 *Oggetto:* [OpenSIPS-Users] Opensips 1.9 - Radius accounting Hello. Were anyone able to setup opensips 1.9 with radius accounting? Unfortunately the tutorial is based on an old version of Opensips were the ACC module was different. Of course i can have accounting in the opensips database (acc table) but tcpdump-ing on radius ports i don't see any traffic (if i try with the radtest it works and i can see the traffic with tcpdump). Thanks, Leo. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-1-9-Radius-accounting-tp7585735.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] msrp relay
Hi, Thanks for your replay. @nguyen khue: It's look like boghe not support msrprelay(authnetication), it support just msrp switch and peer2peer http://code.google.com/p/doubango/issues/detail?id=78 https://groups.google.com/forum/#!msg/doubango/jYxwWKKCZ60/3I7za6ChGpgJ Dani On Mon, Feb 18, 2013 at 4:50 PM, Saúl Ibarra Corretgé s...@ag-projects.comwrote: On Feb 18, 2013, at 2:26 PM, Dani Popa wrote: Hi, I think it's more helpful if you can give us calltrace in case of using msrp, sipproxy and of course 2 sip clients. Msrprelay it's act as a mediaproxy or the sip client should connect first to msrprelay, stream info from msrp and use them in invite to the sip proxy ? I think it's not. You need to understand how protocols work with each other if you are building a service with them. You don't necessarily need to read tons of specifications, MSRP for example is just 2 RFCs, one for the core protocol and another one for using relays for NAT traversal. Here are some examples on how the relays are inserted on the SDP: https://tools.ietf.org/html/rfc4976#section-11 We also have different scenario usage diagrams on msrprelay.org. Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] msrp relay
: sip:1001@192.168.103.107:1059;transport=tcp Call-ID: 5538bbe8-b226-77d1-f14e-8b9ff44f0afd CSeq: 6081 INVITE Content-Type: application/sdp Content-Length: 315 Via: SIP/2.0/TCP 192.168.0.35:55150 ;rport=55150;received=113.160.24.110;branch=z9hG4bK891608784 Record-Route: sip:123.30.188.104;transport=tcp;lr=on;did=817.c172;nat=yes Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE v=0 o=doubango 1983 678901 IN IP4 192.168.103.107 s=- c=IN IP4 192.168.103.107 t=0 0 m=message 1060 TCP/MSRP * c=IN IP4 192.168.103.107 a=path:msrp://192.168.103.107:1060/1838028;tcp a=connection:new a=setup:active a=accept-types:message/CPIM a=accept-wrapped-types:application/octet-stream a=recvonly ACK sip:1001@192.168.103.107:1059;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 123.30.188.104;branch=z9hG4bKcydzigwkX;i=2 Via: SIP/2.0/TCP 192.168.0.35:55150 ;received=113.160.24.110;branch=z9hG4bK891589177;rport=55150 From: sip:1004@123.30.188.104;tag=891604370 To: sip:1001@123.30.188.104;tag=1839963 Contact: sip:1004@192.168.0.35:55150 ;alias=113.160.24.110~55150~2;transport=tcp;+g.oma.sip-im;language=en,fr Call-ID: 5538bbe8-b226-77d1-f14e-8b9ff44f0afd CSeq: 6081 ACK Content-Length: 0 Max-Forwards: 16 Subject: FIXME Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: IM-client/OMA1.0 Boghe/v2.0.130.804 BYE sip:1001@192.168.103.107:1059;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 123.30.188.104;branch=z9hG4bK92d7.e0554b27.0;i=2 Via: SIP/2.0/TCP 192.168.0.35:55150 ;received=113.160.24.110;branch=z9hG4bK891605340;rport=55150 From: sip:1004@123.30.188.104;tag=891604370 To: sip:1001@123.30.188.104;tag=1839963 Call-ID: 5538bbe8-b226-77d1-f14e-8b9ff44f0afd CSeq: 6083 BYE Content-Length: 0 Max-Forwards: 16 Accept-Contact: *;+g.oma.sip-im Accept-Contact: *;language=en,fr Subject: FIXME Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: IM-client/OMA1.0 Boghe/v2.0.130.804 P-Preferred-Identity: sip:1004@123.30.188.104 SIP/2.0 200 OK Via: SIP/2.0/TCP 123.30.188.104;branch=z9hG4bK92d7.e0554b27.0;i=2 From: sip:1004@123.30.188.104;tag=891604370 To: sip:1001@123.30.188.104;tag=1839963 Contact: sip:1001@192.168.103.107:1059;transport=tcp Call-ID: 5538bbe8-b226-77d1-f14e-8b9ff44f0afd CSeq: 6083 BYE Content-Length: 0 Via: SIP/2.0/TCP 192.168.0.35:55150 ;rport=55150;received=113.160.24.110;branch=z9hG4bK891605340 I run msrprelay and SIP server on same physical server with IP: 123.30.188.104. From: Saúl Ibarra Corretgé s...@ag-projects.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Monday, February 18, 2013 3:10 PM Subject: Re: [OpenSIPS-Users] msrp relay On Feb 18, 2013, at 5:03 AM, nguyen khue wrote: Hi all, How I can integrates msrprelay (msrprelay.org) with opensips to make File Transfer session between SIP end-points located behind NAT?. Please guide me. I tested file transfer between two SIP end-points in LAN and it worked successful. What problems did you ran into? Did you follow the installation guide http://msrprelay.org/projects/msrprelay/wiki/InstallationGuide ? Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] msilo on failure_route
Hi,
Regarding msilo module and example from the documentation, one simple
question:
if i have 5 clients already registered and non of them know IM(message sip
method), on next failure_route(it was taken from modules documentation).
haw many times the message is stored in database ? In fact the question is,
because of sip forking, how many times the IM message is stored in db with
m_store on failure_route.
Thanks,
Dani
failure_route[1] {
# forwarding failed -- check if the request was a MESSAGE
if (!method==MESSAGE)
{
exit;
};
log(1,MSILO:the downstream UA doesn't support MESSAGEs\n);
# we have changed the R-URI with the contact address, ignore it now
if (m_store($ou))
{
log(MSILO: offline message stored\n);
t_reply(202, Accepted);
}else{
log(MSILO: offline message NOT stored\n);
t_reply(503, Service Unavailable);
};
}
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] msilo on failure_route
Five SIP clients with the same username.
Dani
On Fri, Feb 15, 2013 at 12:19 PM, Dani Popa dani.p...@gmail.com wrote:
Hi,
Regarding msilo module and example from the documentation, one simple
question:
if i have 5 clients already registered and non of them know IM(message sip
method), on next failure_route(it was taken from modules documentation).
haw many times the message is stored in database ? In fact the question is,
because of sip forking, how many times the IM message is stored in db with
m_store on failure_route.
Thanks,
Dani
failure_route[1] {
# forwarding failed -- check if the request was a MESSAGE
if (!method==MESSAGE)
{
exit;
};
log(1,MSILO:the downstream UA doesn't support MESSAGEs\n);
# we have changed the R-URI with the contact address, ignore it now
if (m_store($ou))
{
log(MSILO: offline message stored\n);
t_reply(202, Accepted);
}else{
log(MSILO: offline message NOT stored\n);
t_reply(503, Service Unavailable);
};
}
--
Dani Popa
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] msilo on failure_route
Thanks,
Dani
On Fri, Feb 15, 2013 at 1:22 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote:
**
Hi Dani,
If you do parallel forking to all 5 registrations of the client and all
fails, the failure route is triggered only once (as all branches belong to
the same transaction and failure route is triggered when the transaction
fails).
So the final answer - one time.
Regards
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 02/15/2013 12:22 PM, Dani Popa wrote:
Five SIP clients with the same username.
Dani
On Fri, Feb 15, 2013 at 12:19 PM, Dani Popa dani.p...@gmail.com wrote:
Hi,
Regarding msilo module and example from the documentation, one simple
question:
if i have 5 clients already registered and non of them know IM(message
sip method), on next failure_route(it was taken from modules
documentation). haw many times the message is stored in database ? In fact
the question is, because of sip forking, how many times the IM message is
stored in db with m_store on failure_route.
Thanks,
Dani
failure_route[1] {
# forwarding failed -- check if the request was a MESSAGE
if (!method==MESSAGE)
{
exit;
};
log(1,MSILO:the downstream UA doesn't support MESSAGEs\n);
# we have changed the R-URI with the contact address, ignore it now
if (m_store($ou))
{
log(MSILO: offline message stored\n);
t_reply(202, Accepted);
}else{
log(MSILO: offline message NOT stored\n);
t_reply(503, Service Unavailable);
};
}
--
Dani Popa
--
Dani Popa
___
Users mailing
listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] add sdp on 180 Ringing and play file as ringback tone
Hi, I wondering if it posiible to add sdp on 180 ringing in order to play some ringing tone. The ideea si that i want to play from rtpproxy with rtpproxy_stream2uac/rtpproxy_stream2uas some music as ringback tone to calling party if it's online. -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] add sdp on 180 Ringing and play file as ringback tone
I just want to play media on replay route in case of 18[013] reply, so i'm sure the user was alerted if i got one of them, i'm pretty sure is not the case from the link below and also inserted media is not a fake ringback. Thanks anyway! Dani Popa On Feb 13, 2013, at 0:56, Daniel Goepp d...@goepp.net wrote: Although I do not believe it is technically a violation of the RFC, it is not recommended best practice, and would be a rare implementation. The most common way to support ringback (early media) is with a 183 w/SDP session progress. For a little more information: http://wiki.freeswitch.org/wiki/180_vs._183_vs._Early_Media Of course some googling will give you tons more opinions about ring back and early media. -dg On Tue, Feb 12, 2013 at 10:04 AM, Dani Popa dani.p...@gmail.com wrote: Hi, I wondering if it posiible to add sdp on 180 ringing in order to play some ringing tone. The ideea si that i want to play from rtpproxy with rtpproxy_stream2uac/rtpproxy_stream2uas some music as ringback tone to calling party if it's online. -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] add sdp on 180 Ringing and play file as ringback tone
Thank you, Dani On Wed, Feb 13, 2013 at 2:00 AM, Daniel Goepp d...@goepp.net wrote: I know there is a perl script that does the opposite, take a 183, and convert to a 180: http://www.opensips.org/Resources/DocsTutPerl183to180 Perhaps you could do something like this to take a 180, and convert to a 183 w/SDP? The link I sent was really just conversational, not intended to be an example of how to do it, just some info on the difference, and why it's not recommended to do a 180 w/SDP, as that removes local custom ringtones. I do think that what you want to do is a 183 with early media, not just append an SDP to a 180. Good luck though:) -dg On Tue, Feb 12, 2013 at 3:28 PM, Dani Popa dani.p...@gmail.com wrote: I just want to play media on replay route in case of 18[013] reply, so i'm sure the user was alerted if i got one of them, i'm pretty sure is not the case from the link below and also inserted media is not a fake ringback. Thanks anyway! Dani Popa On Feb 13, 2013, at 0:56, Daniel Goepp d...@goepp.net wrote: Although I do not believe it is technically a violation of the RFC, it is not recommended best practice, and would be a rare implementation. The most common way to support ringback (early media) is with a 183 w/SDP session progress. For a little more information: http://wiki.freeswitch.org/wiki/180_vs._183_vs._Early_Media Of course some googling will give you tons more opinions about ring back and early media. -dg On Tue, Feb 12, 2013 at 10:04 AM, Dani Popa dani.p...@gmail.com wrote: Hi, I wondering if it posiible to add sdp on 180 ringing in order to play some ringing tone. The ideea si that i want to play from rtpproxy with rtpproxy_stream2uac/rtpproxy_stream2uas some music as ringback tone to calling party if it's online. -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Send 487 request terminbated while a cancel recieved from UAC
As far as I know, opensips send 487, when receiving 200ok, when forking On Dec 5, 2012 8:19 AM, M.Khaled W Chehab kche...@icucall.com wrote: Dears , How to send a 487 request terminated and drop the call directly if the UA send a cancel ,since now I am sending 200 canceling to UA and send a cancel for the Trunk and wait for his reply . Regards Khaled Chehab Senior NGN Engineer Operations Office - Lebanon Office: +961 1 515155 ext 300 Mobile : +961 3 045212 E-mail: kche...@icucall.com MSN ID :khalidche...@hotmail.com Skype: k_chehab Web Site: http://www.icucall.com http://www.allohi.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Modify Via header
none, I think you want and need to use topology_hiding()http://www.opensips.org/html/docs/modules/devel/dialog.html#id294845 from dialog module. Dani On Tue, Jan 10, 2012 at 1:21 PM, Maciej Bylica mb...@gazeta.pl wrote: Hello, What is the best way to replace or modify Via header of incoming INVITE? I need to change private ip address with $si. Oryginal header is Via: SIP/2.0/UDP 10.10.10.128:5060 ;branch=z9hG4bK-680826 Is it subst? What is your advice? Regards, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] delay for first invite
Hi, I know it's a weird question, but still, it is possible to add a delay (let's say 5 seconds) for the first invite(somehow to increase post dial delay with 5 seconds). Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] delay for first invite
thanks, Dani On 11/22/11 16:09, Bogdan-Andrei Iancu wrote: Hi all, Using sleep() functions in script is really dangerous as actually you block an opensips process in doing the sleep. So if you have 8 processes and you have 8 calls in sleep for 5 secs, you will end up blocking your entire opensips for all SIP traffic. A not simple approach, but more efficient is to first set fr_timer to 5 and send the invite to a destination that does not exists / answer - in 5 seconds you will end up in failure route and you can resume the processing there.and there is no blocking in opensips. Regards, Bogdan On 11/22/2011 03:46 PM, Sammy Govind wrote: We can add delay for a particular host, add error, packet drop and packet reordering in network layer but for just first invite !! ummm...yes in configuration where you detect _first_ INVITE put a sleep in there but then it won't be true network latency simulation. On Tue, Nov 22, 2011 at 6:26 PM, Dani Popa dani.p...@gmail.com mailto:dani.p...@gmail.com wrote: Hi, I know it's a weird question, but still, it is possible to add a delay (let's say 5 seconds) for the first invite(somehow to increase post dial delay with 5 seconds). Thanks, Dani ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] changing dialog timeout value on the fly
hi, thank you for response. I'm not sure that i get you :). How can i change the dialog timeout for fa estabilished dialog, when the second dialog is initialized. Can you give me a short example ? Thanks, Dani On Tue, Nov 1, 2011 at 11:14 AM, Vlad Paiu vladp...@opensips.org wrote: Hello, From the script, you can alter the timeout_avp value at any time after the dialog is established. Unfortunately, you cannot modify AVP values through FIFO. What you can do is use avp_db_load [1] so that you can set the value of an AVP from what you have in DB. So your external APP would have to insert into a DB the new timeout_avp value. [1] http://www.opensips.org/html/**docs/modules/devel/avpops.** html#id250328http://www.opensips.org/html/docs/modules/devel/avpops.html#id250328 Regards, Vlad Paiu OpenSIPS Developer On 10/31/2011 07:02 PM, Dani Popa wrote: hi, it is possible somehow to change/update the dialog timeout_avp(value of it) on the fly. Meaning, after the dialog is established, to change it somehow from fifo ? I want to use It for simultaneous prepaid calls. Thanks, Dani __**_ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-**bin/mailman/listinfo/usershttp://lists.opensips.org/cgi-bin/mailman/listinfo/users __**_ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-**bin/mailman/listinfo/usershttp://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] changing dialog timeout value on the fly
hi, it is possible somehow to change/update the dialog timeout_avp(value of it) on the fly. Meaning, after the dialog is established, to change it somehow from fifo ? I want to use It for simultaneous prepaid calls. Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MWI indicator when integrating with Asterisk
hi,
load module presence_mwi
and
if(is_method(SUBSCRIBE)) {
if (!has_totag()) {
if (avp_check($hdr(Event), eq/message-summary/i)) {
rewritehostport(asterisk.host);
record_route();
if (!t_relay()) {
t_reply(500, Server internal error);
}
exit;
}
}
}
Dani
On 10/27/11 17:34, Schneur Rosenberg wrote:
We have a Opensips server that is used to load balance a few asterisk
servers, the opensips also handles registration, but asterisk handles
everything else, everything works fine but I don't know how to get MWI
indicator to work, I tried rewritehostport for the SUBSCRIBE but it
did not work, can anyone please help me with this.
Thanks
S. Rosenebrg
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No voice and No video, But I can register.
Hi, You have a lot of invite there, and it's hard to follow a single call trace. Can you post a single call trace? Do you make nat detection and fix nat , do you use mediaproxy or nathelper to pass media behaind nat ? Dani On 10/27/11 09:53, Nick wrote: Hello It's my network idoubs on iphone -- NAT -- opensips -- NAT -- linphone on andriod idoubs call linphone or linphone call idoubs. It's OK. But No voice and No video. It's my ngrep log. please give me a suggest. Thanks Nick U 2011/10/27 14:42:30.595631 192.168.20.118:5060 - 111.250.252.241:57344 SIP/2.0 200 OK. Via: SIP/2.0/UDP 111.250.252.241:57344;branch=z9hG4bK1663566969;rport=57344. From: sip:0939723377@220.130.6.180;tag=1116164971. To: sip:0939723377@220.130.6.180;tag=c97b4d1cb1f3d0da549e06a8d482ef63.8257. Call-ID: 9fd9f4fb-8a3e-1614-a9ab-2657b0def8fb. CSeq: 1361788763 REGISTER. Contact: sip:0939723377@192.168.20.139:57685;transport=udp;expires=51;received=sip:220.130.6.180:57685, sip:0939723377@111.250.252.241:61938;transport=udp;expires=1560, sip:0939723377@111.250.252.241:64091;transport=udp;expires=1807, sip:0939723377@192.168.20.139:49289;transport=udp;expires=1909;received=sip:220.130.6.180:49289, sip:0939723377@111.250.252.241:57344;transport=udp;expires=3200. Server: OpenSIPS (1.7.0-tls (i386/linux)). Content-Length: 0. . # U 2011/10/27 14:42:30.599525 111.235.230.93:2339 - 192.168.20.118:5060 jaK... # U 2011/10/27 14:42:42.758246 111.250.252.241:57344 - 192.168.20.118:5060 INVITE sip:09@220.130.6.180 SIP/2.0. Via: SIP/2.0/UDP 111.250.252.241:57344;branch=z9hG4bK409732603;rport. From: sip:0939723377@220.130.6.180;tag=1535502360. To: sip:09@220.130.6.180. Contact: sip:0939723377@111.250.252.241:57344;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel. Call-ID: 330b0d11-fcbb-1606-6536-5727b505bf9d. CSeq: 2031179596 INVITE. Content-Type: application/sdp. Content-Length: 802. Max-Forwards: 70. Route: sip:220.130.6.180:5060;lr;transport=udp. Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel. P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel. Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER. Privacy: none. P-Access-Network-Info: ADSL;utran-cell-id-3gpp=. User-Agent: IM-client/OMA1.0 ios-ngn-stack/v00 (doubango r000). P-Preferred-Identity: sip:0939723377@220.130.6.180. Supported: 100rel. . v=0. o=doubango 1983 678901 IN IP4 111.250.252.241. s=-. c=IN IP4 111.250.252.241. t=0 0. m=audio 34928 RTP/AVP 101 8 0 9 18. a=ptime:20. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=rtpmap:8 PCMA/8000/1. a=rtpmap:0 PCMU/8000/1. a=rtpmap:9 G722/8000/1. a=rtpmap:18 g729/8000/1. a=fmtp:18 annexb=yes. m=video 6632 RTP/AVP 125 106 105 104 121. a=rtpmap:125 VP8/9. a=fmtp:125 CIF=2;QCIF=2;SQCIF=2. a=rtpmap:106 H264/9. a=fmtp:106 profile-level-id=42e01e; packetization-mode # U 2011/10/27 14:42:42.761909 192.168.20.118:5060 - 111.250.252.241:57344 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 111.250.252.241:57344;branch=z9hG4bK409732603;rport=57344. From: sip:0939723377@220.130.6.180;tag=1535502360. To: sip:09@220.130.6.180. Call-ID: 330b0d11-fcbb-1606-6536-5727b505bf9d. CSeq: 2031179596 INVITE. Server: OpenSIPS (1.7.0-tls (i386/linux)). Content-Length: 0. . # U 2011/10/27 14:42:42.762554 192.168.20.118:5060 - 220.130.6.180:2339 INVITE sip:09@192.168.20.149:2339;line=5597fee1c733567 SIP/2.0. Record-Route: sip:192.168.20.118;lr. Via: SIP/2.0/UDP 192.168.20.118;branch=z9hG4bKbda9.9754d403.0. Via: SIP/2.0/UDP 111.250.252.241:57344;received=111.250.252.241;branch=z9hG4bK409732603;rport=57344. From: sip:0939723377@220.130.6.180;tag=1535502360. To: sip:09@220.130.6.180. Contact: sip:0939723377@111.250.252.241:57344;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel. Call-ID: 330b0d11-fcbb-1606-6536-5727b505bf9d. CSeq: 2031179596 INVITE. Content-Type: application/sdp. Content-Length: 802. Max-Forwards: 69. Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel. P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel. Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER. Privacy: none. P-Access-Network-Info: ADSL;utran-cell-id-3gpp=. User-Agent: IM-client/OMA1.0 ios-ngn-stack/v00 (doubango r000). P-Preferred-Identity: sip:0939723377@220.130.6.180. Supported: 100rel. . v=0. o=doubango 1983 678901 IN IP4 111.250.252.241. s=-. c=IN IP4 111.250.252.241. t=0 0. m=audio 34928 RTP/AVP 101 8 0 9 18. a=ptime:20. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=rtpmap:8 PCMA/8000/1. a=rtpmap:0 PCMU/8000/1. a=rtpmap:9 G722/8000/1. a=rtpmap:18 g729/8000/1. a=fmtp:18 annexb=yes. m=video 6632 RTP/AVP 125 106 105 104 121. a=rtpmap:125 VP8/9. a= # U 2011/10/27 14:42:42.762770 192.168.20.118:5060 - 61.220.124.37:2339 INVITE
Re: [OpenSIPS-Users] ASR ACD Monitoring
Him if you look for asterisk tools, i think you should ask on asterisk mailing list, not opensips. Dani On 10/24/11 11:54, Faisal Rehman wrote: Hi I am in search of an opensource/paid tool for the monitoring and analysis of ASR ACD from Master.csv (of Asterisk), before that Sammy recommended me a software for this but it was too expensive for me, so any ideas about a little cheap recommended tool for the sake of above purpose. Regards, Faisal Rehman ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] IM authorize by xcap
Thanks Adrian, Dani On 09/23/11 21:52, Adrian Georgescu wrote: IM based on SIP can be done poorly by using MESSAGE method or properly by using MSRP media plane. Policy to allow or deny incoming requests is up to the end-points. XCAP typically stores Presence related policy. Nobody stops you to extend such presence rules to session requests and implement an OpenSIPS module to query them but it would be s a stretch of imagination. Adrian On Sep 23, 2011, at 3:05 PM, Dani Popa wrote: Hi all, Does opensips have implemented something like authorize_messages to authorize IM by xcap ? Thanks, Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] IM authorize by xcap
Hi all, Does opensips have implemented something like authorize_messages to authorize IM by xcap ? Thanks, Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] pua_dialoginfo and publish method
Hi, Maybe i was lame in my explanation. The Publish for dialog event is hadled by pua_dialoginfo, but if i make a call, on first invite pua_dialoginfo will Publish dialog info and after that, my softphone send Publish with my status. The presence for my user is updated with new Publish from my status and overide status from pua_dialoginfo. I'll make more tests and let you know. Thanks, Dani On 09/19/11 18:57, Anca Vamanu wrote: Hi Dani, Does your phone actually send Publish for dialog event? I never saw this, what phone are you using? Anyhow, the Publish from the phone can not delete the information that opensips has published with pua_dialoginfo because each record is identified by a ETAG and when updating/inserting a match against this ETAG is done. Please look closer in presentity table. Regards, Anca On Thu, Sep 15, 2011 at 4:09 PM, Dani Popa dani.p...@gmail.com mailto:dani.p...@gmail.com wrote: Hi, I'm using pua_dialoginfo to publish dialog info. My problem is that if in the middle of call, my softphone will send PUBLISH, it will overwrite the publish from dialog info, and i don't want this. Can you give me a hint how should i avoid this overwriting, if it possible ? Thanks, Dani ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] pua_dialoginfo and publish method
Hi, I'm using pua_dialoginfo to publish dialog info. My problem is that if in the middle of call, my softphone will send PUBLISH, it will overwrite the publish from dialog info, and i don't want this. Can you give me a hint how should i avoid this overwriting, if it possible ? Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] methods= from register contact
Thanks,
Dani
On 09/14/11 11:25, Vlad Paiu wrote:
Hello,
Upon registration, each UA can supply a list of allowed/supported methods.
OpenSIPS saves this information by using an integer, and bitmask flags.
The enum used is :
enum request_method {
METHOD_UNDEF=0, /* 0 - --- */
METHOD_INVITE=1, /* 1 - 2^0 */
METHOD_CANCEL=2, /* 2 - 2^1 */
METHOD_ACK=4, /* 3 - 2^2 */
METHOD_BYE=8, /* 4 - 2^3 */
METHOD_INFO=16, /* 5 - 2^4 */
METHOD_OPTIONS=32,/* 6 - 2^5 */
METHOD_UPDATE=64, /* 7 - 2^6 */
METHOD_REGISTER=128, /* 8 - 2^7 */
METHOD_MESSAGE=256, /* 9 - 2^8 */
METHOD_SUBSCRIBE=512, /* 10 - 2^9 */
METHOD_NOTIFY=1024, /* 11 - 2^10 */
METHOD_PRACK=2048,/* 12 - 2^11 */
METHOD_REFER=4096,/* 13 - 2^12 */
METHOD_PUBLISH=8192, /* 14 - 2^13 */
METHOD_OTHER=16384/* 15 - 2^14 */
};
0x1F6F = 8047 = 4096 + 2048 + 1024 + 512 + 256 + 64 + 32 + 8 + 4 + 2 + 1 ,
Basically, using the methods integer and the above enumeration, you
can tell what methods the registering UA supports.
Regards,
Vlad Paiu
OpenSIPS Developer
On 09/13/2011 10:42 PM, Dani Popa wrote:
Hi all,
What does it mean methods=0x1F6F from register contact when i see it
with opensipsctl ul show, and how can i decode it ?
Contact::
sip:@x.x.x.x:xxx;transport=UDP;ob;q=;expires=525;flags=0x0;cflags=0x0;socket=udp:y.y.y.y:;methods=0x1F6F;user_agent=CSipSimple
r801 / SGH-I897-7
Thanks,
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] methods= from register contact
Hi,
Based on this mehods value, opensips will relay sip messages to the
contact ? I mean, if methods value is matching bitwise with
METHOD_MESSAGE value, opensips will sent sip MESSAGES method to the
contact, otherwise, it will not send it ?
Thanks,
Dani
On 09/14/11 11:25, Vlad Paiu wrote:
Hello,
Upon registration, each UA can supply a list of allowed/supported methods.
OpenSIPS saves this information by using an integer, and bitmask flags.
The enum used is :
enum request_method {
METHOD_UNDEF=0, /* 0 - --- */
METHOD_INVITE=1, /* 1 - 2^0 */
METHOD_CANCEL=2, /* 2 - 2^1 */
METHOD_ACK=4, /* 3 - 2^2 */
METHOD_BYE=8, /* 4 - 2^3 */
METHOD_INFO=16, /* 5 - 2^4 */
METHOD_OPTIONS=32,/* 6 - 2^5 */
METHOD_UPDATE=64, /* 7 - 2^6 */
METHOD_REGISTER=128, /* 8 - 2^7 */
METHOD_MESSAGE=256, /* 9 - 2^8 */
METHOD_SUBSCRIBE=512, /* 10 - 2^9 */
METHOD_NOTIFY=1024, /* 11 - 2^10 */
METHOD_PRACK=2048,/* 12 - 2^11 */
METHOD_REFER=4096,/* 13 - 2^12 */
METHOD_PUBLISH=8192, /* 14 - 2^13 */
METHOD_OTHER=16384/* 15 - 2^14 */
};
0x1F6F = 8047 = 4096 + 2048 + 1024 + 512 + 256 + 64 + 32 + 8 + 4 + 2 + 1 ,
Basically, using the methods integer and the above enumeration, you
can tell what methods the registering UA supports.
Regards,
Vlad Paiu
OpenSIPS Developer
On 09/13/2011 10:42 PM, Dani Popa wrote:
Hi all,
What does it mean methods=0x1F6F from register contact when i see it
with opensipsctl ul show, and how can i decode it ?
Contact::
sip:@x.x.x.x:xxx;transport=UDP;ob;q=;expires=525;flags=0x0;cflags=0x0;socket=udp:y.y.y.y:;methods=0x1F6F;user_agent=CSipSimple
r801 / SGH-I897-7
Thanks,
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips 1.6.4 core/bug
Hi, My opensips used for presence stoped with Segmentation fault. root@test:/home# gdb opensips_1_6/opensips core GNU gdb (GDB) 7.3-debian Copyright (C) 2011 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i486-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /home/opensips_1_6/opensips...done. [New LWP 14277] warning: Can't read pathname for load map: Input/output error. [Thread debugging using libthread_db enabled] Core was generated by `/usr/local/sbin/opensips -P /var/run/opensips_presence/opensips_presence.pid -m'. Program terminated with signal 11, Segmentation fault. #0 0xb6c2c3ee in get_rules_doc (user=0xbfb603ac, domain=0xbfb603bc, type=2, rules_doc=0xb6c6a2a4) at xcap_auth.c:527 527 body.len = strlen(body.s); (gdb) bt #0 0xb6c2c3ee in get_rules_doc (user=0xbfb603ac, domain=0xbfb603bc, type=2, rules_doc=0xb6c6a2a4) at xcap_auth.c:527 #1 0x081f6bfc in mem_pool () Backtrace stopped: Not enough registers or memory available to unwind further The core file has 256M, if you need it, i'll post it on web, but please let me know. Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.6.4 core/bug
Hi, (gdb) frame 0 #0 0xb6c2c3ee in get_rules_doc (user=0xbfb603ac, domain=0xbfb603bc, type=2, rules_doc=0xb6c6a2a4) at xcap_auth.c:527 527 body.len = strlen(body.s); (gdb) p row_vals[0] value has been optimized out (gdb) On 09/14/11 15:43, Vlad Paiu wrote: Hello, In frame 0, could you please do p row_vals[0] and paste here the output ? Regards, Vlad Paiu OpenSIPS Developer On 09/14/2011 02:53 PM, Dani Popa wrote: Hi, My opensips used for presence stoped with Segmentation fault. root@test:/home# gdb opensips_1_6/opensips core GNU gdb (GDB) 7.3-debian Copyright (C) 2011 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i486-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /home/opensips_1_6/opensips...done. [New LWP 14277] warning: Can't read pathname for load map: Input/output error. [Thread debugging using libthread_db enabled] Core was generated by `/usr/local/sbin/opensips -P /var/run/opensips_presence/opensips_presence.pid -m'. Program terminated with signal 11, Segmentation fault. #0 0xb6c2c3ee in get_rules_doc (user=0xbfb603ac, domain=0xbfb603bc, type=2, rules_doc=0xb6c6a2a4) at xcap_auth.c:527 527 body.len = strlen(body.s); (gdb) bt #0 0xb6c2c3ee in get_rules_doc (user=0xbfb603ac, domain=0xbfb603bc, type=2, rules_doc=0xb6c6a2a4) at xcap_auth.c:527 #1 0x081f6bfc in mem_pool () Backtrace stopped: Not enough registers or memory available to unwind further The core file has 256M, if you need it, i'll post it on web, but please let me know. Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips 1.6.4 core/bug
hi, i imagine that. I patched and start opensips with new patch and now waiting. Thanks, Dani On 09/14/11 16:29, Vlad Paiu wrote: Hello, Ok, that wasn't of really much help :) Please try the attached patch and let me know if OpenSIPS still crashes. Regards, Vlad Paiu OpenSIPS Developer On 09/14/2011 03:53 PM, Dani Popa wrote: Hi, (gdb) frame 0 #0 0xb6c2c3ee in get_rules_doc (user=0xbfb603ac, domain=0xbfb603bc, type=2, rules_doc=0xb6c6a2a4) at xcap_auth.c:527 527 body.len = strlen(body.s); (gdb) p row_vals[0] value has been optimized out (gdb) On 09/14/11 15:43, Vlad Paiu wrote: Hello, In frame 0, could you please do p row_vals[0] and paste here the output ? Regards, Vlad Paiu OpenSIPS Developer On 09/14/2011 02:53 PM, Dani Popa wrote: Hi, My opensips used for presence stoped with Segmentation fault. root@test:/home# gdb opensips_1_6/opensips core GNU gdb (GDB) 7.3-debian Copyright (C) 2011 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i486-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /home/opensips_1_6/opensips...done. [New LWP 14277] warning: Can't read pathname for load map: Input/output error. [Thread debugging using libthread_db enabled] Core was generated by `/usr/local/sbin/opensips -P /var/run/opensips_presence/opensips_presence.pid -m'. Program terminated with signal 11, Segmentation fault. #0 0xb6c2c3ee in get_rules_doc (user=0xbfb603ac, domain=0xbfb603bc, type=2, rules_doc=0xb6c6a2a4) at xcap_auth.c:527 527 body.len = strlen(body.s); (gdb) bt #0 0xb6c2c3ee in get_rules_doc (user=0xbfb603ac, domain=0xbfb603bc, type=2, rules_doc=0xb6c6a2a4) at xcap_auth.c:527 #1 0x081f6bfc in mem_pool () Backtrace stopped: Not enough registers or memory available to unwind further The core file has 256M, if you need it, i'll post it on web, but please let me know. Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] methods= from register contact
Hi all, What does it mean methods=0x1F6F from register contact when i see it with opensipsctl ul show, and how can i decode it ? Contact:: sip:@x.x.x.x:xxx;transport=UDP;ob;q=;expires=525;flags=0x0;cflags=0x0;socket=udp:y.y.y.y:;methods=0x1F6F;user_agent=CSipSimple r801 / SGH-I897-7 Thanks, -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 1.7.0 Topology Hiding and NAT Traversal
Hi,
make two sessions of opensips config. One that handle NAT and another for
topology hiding.
Dani
On Tue, Sep 13, 2011 at 7:30 PM, Jeremy Childs jere...@ssimicro.com wrote:
I'm having a problem with the dialog module's topology hiding when a UA
is behind a NAT.
If I call
if (nat_uac_test()) {
fix_nated_contact();
}
topology_hiding();
The Contact header is rewritten twice - once by fix_nated_contact() and
again by topology_hiding(). This results in an invalid contact header.
Is there an obvious way I'm missing that could make these two modules
coexist, or is the best solution to add NAT knowledge to dlg_tophiding.c?
This seems like a lot of code to duplicate.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] a good document with examples for presence with xcap
Hi, I found, i think, a good document about integrating xcap with presence. Maybe some of you need this: http://download.oracle.com/docs/cd/E17667_01/doc.50/e17669/cpt_concepts.htm Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Callcontrol never returns duplicate callid error code
are you sure that is not handled as retrasmision ? Do you see the times that invite hit call_control ? dani On 09/07/11 14:00, Mino Haluz wrote: Hi, I'm using kamailio+callcontrol2.0.14 , and when kamailio receives identical 3 INVITES, the callcontrol function never returns -3 (return value for duplicate callid). What is the purpose of this return value then ? Thanks, M ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
Hi again, A new mem error, maybe you are interested Aug 24 06:59:50 test /usr/local/sbin/opensips[1324]: SUBSCRIBE : SUBSCRIBE -- b6874def55c7f5cf8e4d61b4cc4095c9-f985 -- e170995a-- Aug 24 06:59:50 test /usr/local/sbin/opensips[1324]: ERROR:presence:send_2XX_reply: No more pkg memory Aug 24 06:59:50 test /usr/local/sbin/opensips[1324]: CRITICAL:core:qm_free: bad pointer (nil) (out of memory block!) - aborting Dani On 08/19/11 17:31, Bogdan-Andrei Iancu wrote: Hi Dani, You can not have comments in multi-line assignments So, instead of DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ you should do : DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ -DDBG_QM_MALLOC \ #-DF_MALLOC \ #-DDBG_F_MALLOC \ BTW, once you compiled in the DBG support, set: memlog=6 memdump=1 in order to get only the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs) \ . . . . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ opensips will not be compiled with -DDBG_QM_MALLOC I'll come back with goodies, Thanks, Dani On 08/19/11 12:01, Vlad Paiu wrote: Hello, Thanks for the reply. Unfortunately, It seems that you have not compiled OpenSIPS with -DDBG_QM_MALLOC, so please review my last email and see that I also suggested editing Makefiles.defs and uncommenting that particular line, while commenting -DF_MALLOC. If you succesfully do this, opensips -V should show something like : . SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC . Please do this and return with the logs at shutdown. Thank you. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
Hi, Thanks for your response. Right now PKG_MEM_POOL_SIZE is 8*1024*1024 and i have 33 users online using presence(it's right that any expire timers regarding publish and notify are 60 seconds instead 3600 as it is in documentation) . What value should i use for let's say, 100k users using presence ? I will configure a separate presence server to check if the problem is related to presence. Dani On 08/24/11 16:40, Vlad Paiu wrote: Hello Dani, About the memory log that you sent.. I see no obvious memory leak. So as far as I can see, there are two possibilities - something really needs that much PKG. To make sure that this isn't the case, please increase your PKG_MEM_POOL_SIZE to a higher value and let us know if you still experience the same problem. - at shutdown, that extra bogus memory is freed up and doesn't show up anymore in the memory dump. Please try first increasing the memory and see if OpenSIPS is still reporting out of mem problems. About the second issue with bad pointer (nil) (out of memory block!) - aborting , there was a bug in OpenSIPS which lead to the calling of free(0), in a no more pkg situation. This has been fixed in trunk, 1.7 and 1.6 branches. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
Hi, I think now the logs should be fine and you can find them at: http://92.55.132.13/loca7.log_v1.tar.gz root@test:/var/log# opensips -V version: opensips 1.6.4-2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, USE_SCTP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT 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. svnrevision: 2:8256M @(#) $Id: main.c 7530 2010-12-13 19:07:53Z bogdan_iancu $ main.c compiled on 05:48:37 Aug 19 2011 with gcc 4.5.2 Thanks, Dani Popa On 08/19/11 17:31, Bogdan-Andrei Iancu wrote: Hi Dani, You can not have comments in multi-line assignments So, instead of DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ you should do : DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ -DDBG_QM_MALLOC \ #-DF_MALLOC \ #-DDBG_F_MALLOC \ BTW, once you compiled in the DBG support, set: memlog=6 memdump=1 in order to get only the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs) \ . . . . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ opensips will not be compiled with -DDBG_QM_MALLOC I'll come back with goodies, Thanks, Dani On 08/19/11 12:01, Vlad Paiu wrote: Hello, Thanks for the reply. Unfortunately, It seems that you have not compiled OpenSIPS with -DDBG_QM_MALLOC, so please review my last email and see that I also suggested editing Makefiles.defs and uncommenting that particular line, while commenting -DF_MALLOC. If you succesfully do this, opensips -V should show something like : . SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC . Please do this and return with the logs at shutdown. Thank you. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] msilo and server_header
Thanks, Dani On 08/19/11 18:01, Bogdan-Andrei Iancu wrote: Hi Dani, In your case opensips will act as UAC (not server), so you need to define your custom user_agent_header: http://www.opensips.org/Resources/DocsCoreFcn17#toc96 Regards, Bogdan On 08/16/2011 03:12 PM, Dani Popa wrote: Hi, When using m_store($ru) the SIP messages sent back to sender have default server_header and not the one i rewrite it. Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Ability to tell active calls per customer
Hi, I think you could use dialog profile, but not sure. Dani On 08/19/11 23:17, Robert Thomas wrote: Hi, I have a load balancer module to distribute calls among my Gateways. I can use the lb_list command to see the active calls per gw, but I would like something similar to graph my customer amount of active calls. I was thinking creating another set of resources on the load balancer, but this would be messy. Or somehow use the dialog module for this. Ideally if the variable could be exposed via snmp I could use cacti to graph each customer. Has anyone tried this, and what would be the best way? Sent from my iPhone ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
Hi, Where should i find memory dump ? I have something in logs about memory. I'll attach an file. Please let me know if this is what you need. I also increased PKG_MEM_POOL_SIZE = 8 *1024 * 1024, and shared mem to 256, and also updated opensips 1.6.4 to latest svn revision, i think. root@test:/home/danip/opensips_1_6# opensipsctl fifo get_statistics all | grep shmem shmem:total_size = 268435456 shmem:used_size = 2084688 shmem:real_used_size = 2221616 shmem:max_used_size = 3095312 shmem:free_size = 266213840 shmem:fragments = 796 root@test:/home/danip/opensips_1_6# opensips -V version: opensips 1.6.4-2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, USE_SCTP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT 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. svnrevision: 2:8256M @(#) $Id: main.c 7530 2010-12-13 19:07:53Z bogdan_iancu $ main.c compiled on 13:13:09 Aug 17 2011 with gcc 4.5.2 Please let me know how i can help more. logs about memory you can find at http://92.55.132.13/local7.log.mem.tar.gz, i tried to send on list but the attach was too big. Thanks, Dani On 08/04/11 18:26, Vlad Paiu wrote: Hello, Is it possible that you upgrade to 1.7 ? It is possible that this issue was fixed in the latest OpenSIPS version. If not, go to Makefile.defs, uncomment the line with -DDBG_QM_MALLOC \ and comment the line with -DF_MALLOC \ and then recompile OpenSIPS. Also set memlog=1 in your OpenSIPS cfg, and when the memory get's filled up you can either shutdown the proxy or send a SIGUSR1 signal to the attendant to get a memory dump. Please return with the memory dump and we will try to help. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
Hi again, i also saw that i compiled opensips with libxmlrpc-c3-dev and libxmlrpc-c3 and i was warned somewhere that i'll compile it on my own risk. Now i removed libxmlrpc-c3-dev and libxmlrpc-c3 and i compiled with libxmlrpc-c++4-dev without warnings. Let's see what we will get! Thanks, Dani On 08/19/11 12:01, Vlad Paiu wrote: Hello, Thanks for the reply. Unfortunately, It seems that you have not compiled OpenSIPS with -DDBG_QM_MALLOC, so please review my last email and see that I also suggested editing Makefiles.defs and uncommenting that particular line, while commenting -DF_MALLOC. If you succesfully do this, opensips -V should show something like : . SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC . Please do this and return with the logs at shutdown. Thank you. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs) \ . . . . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ opensips will not be compiled with -DDBG_QM_MALLOC I'll come back with goodies, Thanks, Dani On 08/19/11 12:01, Vlad Paiu wrote: Hello, Thanks for the reply. Unfortunately, It seems that you have not compiled OpenSIPS with -DDBG_QM_MALLOC, so please review my last email and see that I also suggested editing Makefiles.defs and uncommenting that particular line, while commenting -DF_MALLOC. If you succesfully do this, opensips -V should show something like : . SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC . Please do this and return with the logs at shutdown. Thank you. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
Hi, what about this logs: http://92.55.132.13/memdump.log.tar.gz thanks, Dani On 08/19/11 17:31, Bogdan-Andrei Iancu wrote: Hi Dani, You can not have comments in multi-line assignments So, instead of DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ you should do : DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ -DDBG_QM_MALLOC \ #-DF_MALLOC \ #-DDBG_F_MALLOC \ BTW, once you compiled in the DBG support, set: memlog=6 memdump=1 in order to get only the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs) \ . . . . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ opensips will not be compiled with -DDBG_QM_MALLOC I'll come back with goodies, Thanks, Dani On 08/19/11 12:01, Vlad Paiu wrote: Hello, Thanks for the reply. Unfortunately, It seems that you have not compiled OpenSIPS with -DDBG_QM_MALLOC, so please review my last email and see that I also suggested editing Makefiles.defs and uncommenting that particular line, while commenting -DF_MALLOC. If you succesfully do this, opensips -V should show something like : . SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC . Please do this and return with the logs at shutdown. Thank you. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
Hi Bogdan, I did and solved compliling opensips with -DDBG_QM_MALLOC. Thanks, Dani On 08/19/11 17:31, Bogdan-Andrei Iancu wrote: Hi Dani, You can not have comments in multi-line assignments So, instead of DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ you should do : DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ -DDBG_QM_MALLOC \ #-DF_MALLOC \ #-DDBG_F_MALLOC \ BTW, once you compiled in the DBG support, set: memlog=6 memdump=1 in order to get only the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs) \ . . . . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ opensips will not be compiled with -DDBG_QM_MALLOC I'll come back with goodies, Thanks, Dani On 08/19/11 12:01, Vlad Paiu wrote: Hello, Thanks for the reply. Unfortunately, It seems that you have not compiled OpenSIPS with -DDBG_QM_MALLOC, so please review my last email and see that I also suggested editing Makefiles.defs and uncommenting that particular line, while commenting -DF_MALLOC. If you succesfully do this, opensips -V should show something like : . SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC . Please do this and return with the logs at shutdown. Thank you. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
the logs are also ok if opensips crash instead stop it ? Thanks, Dani On 08/19/11 17:31, Bogdan-Andrei Iancu wrote: Hi Dani, You can not have comments in multi-line assignments So, instead of DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ you should do : DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ -DDBG_QM_MALLOC \ #-DF_MALLOC \ #-DDBG_F_MALLOC \ BTW, once you compiled in the DBG support, set: memlog=6 memdump=1 in order to get only the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs) \ . . . . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ opensips will not be compiled with -DDBG_QM_MALLOC I'll come back with goodies, Thanks, Dani On 08/19/11 12:01, Vlad Paiu wrote: Hello, Thanks for the reply. Unfortunately, It seems that you have not compiled OpenSIPS with -DDBG_QM_MALLOC, so please review my last email and see that I also suggested editing Makefiles.defs and uncommenting that particular line, while commenting -DF_MALLOC. If you succesfully do this, opensips -V should show something like : . SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC . Please do this and return with the logs at shutdown. Thank you. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
sorry because i waste your time. the logs i sent to you are wrong. If i'ii replicate the case i'll send you good logs. Thanks, Dani On 08/19/11 17:31, Bogdan-Andrei Iancu wrote: Hi Dani, You can not have comments in multi-line assignments So, instead of DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ you should do : DEFS+= $(extra_defs) \ . -DCHANGEABLE_DEBUG_LEVEL \ -DDBG_QM_MALLOC \ #-DF_MALLOC \ #-DDBG_F_MALLOC \ BTW, once you compiled in the DBG support, set: memlog=6 memdump=1 in order to get only the memory dump without all runtime logs from mem debugger. Regards, Bogdan On 08/19/2011 01:13 PM, Dani Popa wrote: Hi, True, i changed wrong the Makefiles.defs. I dont know if you need this: if i change Makefile.defs as: DEFS+= $(extra_defs) \ . . . . -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ opensips will not be compiled with -DDBG_QM_MALLOC I'll come back with goodies, Thanks, Dani On 08/19/11 12:01, Vlad Paiu wrote: Hello, Thanks for the reply. Unfortunately, It seems that you have not compiled OpenSIPS with -DDBG_QM_MALLOC, so please review my last email and see that I also suggested editing Makefiles.defs and uncommenting that particular line, while commenting -DF_MALLOC. If you succesfully do this, opensips -V should show something like : . SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC . Please do this and return with the logs at shutdown. Thank you. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] subscribe non case sensitive user from sip uri
OK, Thanks, Dani On 08/08/11 22:38, Bogdan-Andrei Iancu wrote: Hi Dani, This option is obsolete and inherited from the early years of SIP, when 50% of the UAC were not able to be case sensitive. The switch was added ~8 years ago just cope with broken UAC, but this does not mean that it RFC compliant. Regards, Bogdan On 08/05/2011 07:30 PM, Dani Popa wrote: Hi, Ok, but also, registrar module support non case sensitive sip username. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] msilo and server_header
Hi, When using m_store($ru) the SIP messages sent back to sender have default server_header and not the one i rewrite it. Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] subscribe non case sensitive user from sip uri
Hi, Ok, but also, registrar module support non case sensitive sip username. -- Dani Popa On 8/5/11 11:40 AM, Vlad Paiu wrote: Hello, What you're asking for is against the RFC 3261 URI comparison rules, which states that comparison of the userinfo part of the URI should be done case sensitive. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] all sip body headers regarding video removed
Hi all, How can i remove all sip video body headers regardin video. Should i remove any line from body after m=video, or how. Please give me a hint, if you have. Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] all sip body headers regarding video removed
Hi, In fact, i have some problems with one of my pstn gw's that send 400 Incorrect content length, i think, because of too long sip packet. So, because it is pstn, i want to remove video capability(many lines in first invite packet). Dani On 08/04/11 17:02, Razvan Crainea wrote: Hi Dani, Why would you do that? If you don't want to allow video, you can simply replace the video port in the m= line with 0. Regards, Razvan Crainea OpenSIPS Developer On 04.08.2011 16:58, Dani Popa wrote: Hi all, How can i remove all sip video body headers regardin video. Should i remove any line from body after m=video, or how. Please give me a hint, if you have. Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
Hi, I know that, but how can i check why :) root@test:~# opensips -V version: opensips 1.6.4-2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, USE_SCTP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT 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. svnrevision: 2:7861M @(#) $Id: main.c 7530 2010-12-13 19:07:53Z bogdan_iancu $ main.c compiled on 05:57:05 May 24 2011 with gcc 4.5.2 root@test:~# free total used free sharedbuffers cached Mem: 41358163924440 211376 0 1874963253496 -/+ buffers/cache: 4834483652368 Swap: 60651483006064848 I have 50 users online that use presence and IM Dani On 08/04/11 17:04, Razvan Crainea wrote: Hi Dani, It seems you are out of memory. What version of OpenSIPS are you using? Regards, Razvan Crainea OpenSIPS Developer On 04.08.2011 16:07, Dani Popa wrote: Hi, How can i solve this kind of problems ? Opensips doesn't crash, but it not respond to any sip requests. Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]: WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]: ERROR:core:build_res_buf_from_sip_req: out of pkg memory ; needs 496 Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]: ERROR:sl:sl_send_reply_helper: response building failed Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]: ERROR:signaling:sig_send_reply_mod: failed to send reply with sl module Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]: ERROR:auth:challenge: failed to send the response Aug 3 07:36:48 test /usr/local/sbin/opensips[29085]: WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Aug 3 07:36:48 test /usr/local/sbin/opensips[29085]: ERROR:core:build_res_buf_from_sip_req: out of pkg memory ; needs 529 Aug 3 07:36:48 test /usr/local/sbin/opensips[29085]: ERROR:sl:sl_send_reply_helper: response building failed Aug 3 07:36:48 test /usr/local/sbin/opensips[29085]: ERROR:signaling:sig_send_reply_mod: failed to send reply with sl module Aug 3 07:36:48 test /usr/local/sbin/opensips[29085]: ERROR:auth:challenge: failed to send the response Aug 3 07:36:50 test /usr/local/sbin/opensips[29091]: WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Aug 3 07:36:50 test /usr/local/sbin/opensips[29091]: ERROR:core:build_res_buf_from_sip_req: out of pkg memory ; needs 480 Aug 3 07:36:50 test /usr/local/sbin/opensips[29091]: ERROR:sl:sl_send_reply_helper: response building failed Aug 3 07:36:50 test /usr/local/sbin/opensips[29091]: ERROR:signaling:sig_send_reply_mod: failed to send reply with sl module Aug 3 07:36:50 test /usr/local/sbin/opensips[29091]: ERROR:options:opt_reply: failed to send 200 via send_reply Aug 3 07:36:50 test /usr/local/sbin/opensips[29087]: WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Aug 3 07:36:50 test /usr/local/sbin/opensips[29087]: ERROR:core:build_res_buf_from_sip_req: out of pkg memory ; needs 503 Aug 3 07:36:50 test /usr/local/sbin/opensips[29087]: ERROR:sl:sl_send_reply_helper: response building failed Aug 3 07:36:50 test /usr/local/sbin/opensips[29087]: ERROR:signaling:sig_send_reply_mod: failed to send reply with sl module Aug 3 07:36:50 test /usr/local/sbin/opensips[29087]: ERROR:auth:challenge: failed to send the response Aug 3 07:36:51 test /usr/local/sbin/opensips[29086]: WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Aug 3 07:36:51 test /usr/local/sbin/opensips[29086]: ERROR:core:build_res_buf_from_sip_req: out of pkg memory ; needs 503 Aug 3 07:36:51 test /usr/local/sbin/opensips[29086]: ERROR:sl:sl_send_reply_helper: response building failed Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory, will atempt defragmenation
Ok, thanks for quick response. Dani On 08/04/11 18:26, Vlad Paiu wrote: Hello, Is it possible that you upgrade to 1.7 ? It is possible that this issue was fixed in the latest OpenSIPS version. If not, go to Makefile.defs, uncomment the line with -DDBG_QM_MALLOC \ and comment the line with -DF_MALLOC \ and then recompile OpenSIPS. Also set memlog=1 in your OpenSIPS cfg, and when the memory get's filled up you can either shutdown the proxy or send a SIGUSR1 signal to the attendant to get a memory dump. Please return with the memory dump and we will try to help. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] all sip body headers regarding video removed
Thanks, I already did that. Dani On 08/04/11 18:19, Razvan Crainea wrote: Hi Dani, You can try by deleting the most common video codecs (like H261, H263, H264). You can do that using the codec_delete[1] functions from the textops module. I think you should also replace the video port with 0. [1] http://www.opensips.org/html/docs/modules/devel/textops.html#id293910 Regards, Razvan Crainea OpenSIPS Developer On 04.08.2011 18:03, Dani Popa wrote: Hi, In fact, i have some problems with one of my pstn gw's that send 400 Incorrect content length, i think, because of too long sip packet. So, because it is pstn, i want to remove video capability(many lines in first invite packet). Dani On 08/04/11 17:02, Razvan Crainea wrote: Hi Dani, Why would you do that? If you don't want to allow video, you can simply replace the video port in the m= line with 0. Regards, Razvan Crainea OpenSIPS Developer On 04.08.2011 16:58, Dani Popa wrote: Hi all, How can i remove all sip video body headers regardin video. Should i remove any line from body after m=video, or how. Please give me a hint, if you have. Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] subscribe non case sensitive user from sip uri
Hi, it is somehow that username from sip uri to be non case sensitive when we talk about presence and xcap storage? I mean, if userA add userB, in his contact list, i need userA to be able to add userB even he add him(type) as USERB. Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to limit calls to specific number
HI, first aaa_radius_auth and specific sql procedure in sql server. the second asterisk/freeswitch load balncing Dani On 07/12/11 17:06, duane.lar...@gmail.com wrote: For your first question would this work? http://www.ag-projects.com/projects-products-96/535-call-control For your second question I hear that SEMS has better performance than Asterisk or Freeswitch, but I think you have to put a lot of work into it because it isn't as easy to work with as Asterisk. http://www.iptel.org/sems If you can't figure SEMS out then maybe your best bet for an IVR that can handle 1000 calls would be Asterisk Clustering. On Jul 12, 2011 8:03am, Akib Sayyed akibsay...@gmail.com wrote: hello guys i am creating billing system for premium number portal here i need to allow specific number of minutes to a DID. how can i do that any idea's also i need to handle 1000 call i know Opensips can handle it but i want to route those calls to IVR server tell me best server for IVR which can handle 1000 concurrent calls for ivr also server hardware needed if its asterisk -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with siptrace module
the
following line if you want opensips to bind on a specific
interface/port/proto (default bind on all available) */listen=udp:
192.168.2.154:5060 # -- module loading
--mpath=/usr/local/lib/opensips/modules/loadmodule
maxfwd.soloadmodule sl.soloadmodule tm.soloadmodule
dispatcher.soloadmodule mi_fifo.soloadmodule signaling.soloadmodule
options.soloadmodule textops.soloadmodule db_mysql.soloadmodule
siptrace.so#loadmodule acc.so# - setting module-specific
parameters ---# -- dispatcher params --modparam(mi_fifo,
fifo_name, /tmp/opensips_fifo) modparam(dispatcher, ds_ping_from,
sip:proxy@192.1.8.2.154)modparam(dispatcher, ds_ping_interval,
30)modparam(dispatcher, ds_probing_threshhold, 2)modparam(dispatcher,
ds_probing_mode, 1)modparam(dispatcher, list_file,
/usr/local/etc/opensips/dispatcher.list)# modparam(dispatcher,
force_dst, 1) modparam(siptrace, db_url,
mysql://root:Viamonte1621@localhost/opensips)modparam(siptrace,
enable_ack_trace, 1)modparam(siptrace, trace_on,
1)modparam(siptrace, table, sip_trace)modparam(siptrace,
trace_flag, 22) #modparam(acc, log_level, 1)#modparam(acc,
log_flag, 1)#modparam(acc, db_url,
mysql://root:Viamonte1621@localhost/opensips) route{
setflag(22);setbflag(22);sip_trace();if (
!mf_process_maxfwd_header(10) ){
sl_send_reply(483,To Many Hops); drop();};
if (is_method(OPTIONS)) {options_reply();
exit;}ds_select_dst(1, 0);if
($retcodexlog([Redmond] Service full\n);
sl_send_reply(500,Service full);exit;}
forward();#t_relay();#sip_trace();} failure_route[1]
{if (t_check_status((408)|(5[0-9][0-9]))) {
ds_mark_dst();if (ds_select_dst(1, 0))
{forward();} else
{ t_reply(503, Service
Unavailable);}}} onreply_route {
setflag(22);setflag(22);sip_trace();} ThanksDiego
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to check active calls
))
##route(2);
if (is_method(PUBLISH))
{
sl_send_reply(503, Service Unavailable);
exit;
}
if (is_method(REGISTER))
{
# authenticate the REGISTER requests (uncomment to enable auth)
##if (!www_authorize(, subscriber))
##{
##www_challenge(, 0);
##exit;
##}
##
##if (!db_check_to())
##{
##sl_send_reply(403,Forbidden auth ID);
##exit;
##}
if (!save(location))
sl_reply_error();
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
# apply DB based aliases (uncomment to enable)
##alias_db_lookup(dbaliases);
# do lookup with method filtering
if (!lookup(location,m)) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply(404, Not Found);
exit;
case -2:
sl_send_reply(405, Method Not Allowed);
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}
route[1] {
# for INVITEs enable some additional helper routes
if (is_method(INVITE)) {
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
# Presence route
/* uncomment the whole following route for enabling presence
NOTE: do not forget to enable the call of this route from the main
route */
##route[2]
##{
##if (!t_newtran())
##{
##sl_reply_error();
##exit;
##};
##
##if(is_method(PUBLISH))
##{
##handle_publish();
##}
##else
##if( is_method(SUBSCRIBE))
##{
##handle_subscribe();
##}
##
##exit;
##}
branch_route[2] {
xlog(new branch at $ru\n);
}
onreply_route[2] {
xlog(incoming reply\n);
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status(3[0-9][0-9])) {
##t_reply(404,Not found);
##exit;
##}
# uncomment the following lines if you want to redirect the failed
# calls to a different new destination
##if (t_check_status(486|408)) {
##sethostport(192.168.2.100:5060);
### do not set the missed call flag again
##t_relay();
##}
}
--
Akib Sayyed
Matrix-Shell
akibsay...@gmail.com
akibsay...@matrixshell.com
Mob:- +91-966-514-2243
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] one way media stream
Hi, As far as i know it's hard to insert media from other sources in proxy mode for situation like call hold or in call media insert. If you find a solution, please let me know. Dani On 06/29/11 10:06, Barsan Liviu wrote: Hi, Yes, exactly. And obviously for this we want just one way stream. Thanks, Liviu *From:* Saúl Ibarra Corretgé s...@ag-projects.com *To:* OpenSIPS users mailling list users@lists.opensips.org *Sent:* Tue, June 28, 2011 6:55:27 PM *Subject:* Re: [OpenSIPS-Users] one way media stream Hi, On Jun 28, 2011, at 4:41 PM, Barsan Liviu wrote: Hello, We have an OpenSIPs-MediaProxy solution for audio and IM, now we would like to add a functionality, one way audio (e.g. music broadcast from one source to many destinations) in certain situations. Do you think it is possible by modifying the reply-route and cut use_media_proxy()? How would be possible to use a 'switch', which can set one way/two way media streams? The client will be written by us using pjsip.org http://pjsip.org I didn't quite get what you are trying to achieve, but I'll guess: are you trying to implement some kind of music on hold functionality? -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 30x redirect for register
Hi, Thanks Bogdan for confirmation, for list acknowledge, till now, Linksys support 30x redirect for register(an also for invite) and grandstream not for REGISTER(but support for INVITE) . Also x-lite,bria,eyebeam and jitsi don't support 30x for REGISTER(i didn't test for INVITE and other sip methods). If someone have other results, please let me know. Dani On 06/21/11 12:26, Bogdan-Andrei Iancu wrote: Hi Dani, Theoretically yes - I mean according to RFC 3261 is perfectly make sense. On the other hand, some SIP UA implementations do not properly handle a redirect for REGISTER.Probably you need to explicitly test with the UACs you want use. Regards, Bogdan On 06/20/2011 07:08 PM, Dani Popa wrote: Hi all, It is viable solution to use 30(1|2|5) redirect for REGISTER sip messages ? Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] 30x redirect for register
Hi all, It is viable solution to use 30(1|2|5) redirect for REGISTER sip messages ? Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] moh or in dialog media insertion with opensips as sip proxy
Hi all, I looked on the internet for MOH with opensips as sip proxy(not b2b) and other media servers (sems,asterisk,etc). The answers on internet was that is not possible because SIP implementation and because sems,asterisk are full implemented sip servers(invite from opensips to media server for on hold with to-tag and from-tag will be recognized as reinvite without initial invite). Anyone managed to implement the MOH with Opensips as SIP proxy ? Also for features like in call recharge, when the customer go to low credit, i want to announce him that is get low credit, and that he can recharge pressing some confirmation digits. For this features i don't have any solution how can i implement it . Can someone give me a hint ? Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] moh or in dialog media insertion with opensips as sip proxy
Hi, I thought so, but I needed confirmation. Thanks Adrian, Dani On 06/16/11 15:46, Adrian Georgescu wrote: You cannot do this reliably unless you insert a B2BUA in the call flow. Adrian On Jun 16, 2011, at 2:11 PM, Dani Popa wrote: Hi all, I looked on the internet for MOH with opensips as sip proxy(not b2b) and other media servers (sems,asterisk,etc). The answers on internet was that is not possible because SIP implementation and because sems,asterisk are full implemented sip servers(invite from opensips to media server for on hold with to-tag and from-tag will be recognized as reinvite without initial invite). Anyone managed to implement the MOH with Opensips as SIP proxy ? Also for features like in call recharge, when the customer go to low credit, i want to announce him that is get low credit, and that he can recharge pressing some confirmation digits. For this features i don't have any solution how can i implement it . Can someone give me a hint ? Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenXCAP - failed to create OpenXCAP 2.0.0: Document is empty
when start openxcap, it try to take schema from www.w3.org/2001/xml.xsd and www.w3.org doesn't responde. I changed schemaLocation in /usr/local/pymodules/python2.6/xcap/appusage/xml-schemas/xcap-directory.xsd and pointed to local file. Dani On 06/06/11 03:01, duane.lar...@gmail.com wrote: That was missing from the file on the first line, but it still crashes. Here is what it looks like now /usr/share/pyshared/xcap/appusage/xml-schemas/xcap-directory.xsd ?xml version=1.0 encoding=UTF-8? xs:schema targetNamespace=urn:oma:xml:xdm:xcap-directory xmlns=urn:oma:xml:xdm:xcap-directory xmlns:xs=http://www.w3.org/2001/XMLSchema; elementFormDefault=qualified attributeFormDefault=unqualified xs:import namespace=http://www.w3.org/XML/1998/namespace; schemaLocation=http://www.w3.org/2001/xml.xsd/ xs:element name=xcap-directory xs:complexType xs:sequence minOccurs=0 maxOccurs=unbounded xs:element name=folder xs:complexType xs:choice xs:sequence minOccurs=0 maxOccurs=unbounded xs:element name=entry xs:complexType xs:attribute name=uri type=xs:anyURI use=required/ xs:attribute name=etag type=xs:string use=required/ xs:attribute name=last-modified type=xs:dateTime/ xs:attribute name=size type=xs:nonNegativeInteger/ xs:anyAttribute processContents=lax/ /xs:complexType /xs:element /xs:sequence xs:element name=error-code type=xs:string/ /xs:choice xs:attribute name=auid type=xs:string use=required/ /xs:complexType /xs:element /xs:sequence /xs:complexType /xs:element /xs:schema And here is what the syslog says when it crashes. Jun 5 18:58:15 xcap01 openxcap[5705]: fatal error: failed to create OpenXCAP 2.0.0: Document is empty, line 1, column 1 Jun 5 18:58:15 xcap01 openxcap[5705]: Traceback (most recent call last): Jun 5 18:58:15 xcap01 openxcap[5705]: File /usr/bin/openxcap, line 64, in module Jun 5 18:58:15 xcap01 openxcap[5705]: from xcap.server import XCAPServer Jun 5 18:58:15 xcap01 openxcap[5705]: File /usr/lib/pymodules/python2.6/xcap/server.py, line 20, in module Jun 5 18:58:15 xcap01 openxcap[5705]: from xcap import authentication Jun 5 18:58:15 xcap01 openxcap[5705]: File /usr/lib/pymodules/python2.6/xcap/authentication.py, line 28, in module Jun 5 18:58:15 xcap01 openxcap[5705]: from xcap.appusage import getApplicationForURI, namespaces, public_get_applications Jun 5 18:58:15 xcap01 openxcap[5705]: File /usr/lib/pymodules/python2.6/xcap/appusage/__init__.py, line 357, in module Jun 5 18:58:15 xcap01 openxcap[5705]: XCAPDirectoryApplication.id: XCAPDirectoryApplication(storage) Jun 5 18:58:15 xcap01 openxcap[5705]: File /usr/lib/pymodules/python2.6/xcap/appusage/__init__.py, line 59, in __init__ Jun 5 18:58:15 xcap01 openxcap[5705]: self.xml_schema = etree.XMLSchema(xml_schema_doc) Jun 5 18:58:15 xcap01 openxcap[5705]: File xmlschema.pxi, line 105, in lxml.etree.XMLSchema.__init__ (src/lxml/lxml.etree.c:128508) Jun 5 18:58:15 xcap01 openxcap[5705]: XMLSchemaParseError: Document is empty, line 1, column 1 On Jun 3, 2011 5:33am, Saúl Ibarra Corretgé s...@ag-projects.com wrote: Hi, On May 25, 2011, at 5:28 PM, duane.lar...@gmail.com wrote: I am trying to start up openxcap on a server that it used to work on without issue. It starts up and then after 20 seconds or so it crashes. In syslog I am seeing the following error May 25 10:18:00 xcap01 openxcap[701]: Starting OpenXCAP 2.0.0 May 25 10:18:01 xcap01 openxcap[701]: using set_wakeup_fd May 25 10:18:32 xcap01 openxcap[701]: fatal error: failed to create OpenXCAP 2.0.0: Document is empty, line 1, column 1 May 25 10:18:32 xcap01 openxcap[701]: Traceback (most recent call last): May 25 10:18:32 xcap01 openxcap[701]: File /usr/bin/openxcap, line 64, in May 25 10:18:32 xcap01 openxcap[701]: from xcap.server import XCAPServer May 25 10:18:32 xcap01 openxcap[701]: File /usr/lib/pymodules/python2.6/xcap/server.py, line 20, in May 25 10:18:32 xcap01 openxcap[701]: from xcap import authentication May 25 10:18:32 xcap01 openxcap[701]: File /usr/lib/pymodules/python2.6/xcap/authentication.py, line 28, in May 25 10:18:32 xcap01 openxcap[701]: from xcap.appusage import getApplicationForURI, namespaces, public_get_applications May 25 10:18:32 xcap01 openxcap[701]: File /usr/lib/pymodules/python2.6/xcap/appusage/__init__.py, line 357, in May 25 10:18:32 xcap01 openxcap[701]: XCAPDirectoryApplication.id: XCAPDirectoryApplication(storage) May 25 10:18:32 xcap01 openxcap[701]: File /usr/lib/pymodules/python2.6/xcap/appusage/__init__.py, line 59, in __init__ May 25 10:18:32 xcap01 openxcap[701]: self.xml_schema = etree.XMLSchema(xml_schema_doc) May 25 10:18:32 xcap01 openxcap[701]: File xmlschema.pxi, line 105, in lxml.etree.XMLSchema.__init__ (src/lxml/lxml.etree.c:128508) May 25 10:18:32 xcap01 openxcap[701]: XMLSchemaParseError: Document is empty, line 1, column Looks like xcap-directory.xsd schema file lacks the initial
Re: [OpenSIPS-Users] media-dispatcher and media relay connection problem
Hi Liviu, What kernel do you have on running media-relay machine ? Thanks, Dani On 05/26/11 11:14, Barsan Liviu wrote: Hi, With the python-gnutls update to 1.2.1 the mediaproxy works fine. A suggestion: would be welcome a minimal install guide for Ubuntu/Debian, for example I spent several days until I find out that iptables should be loaded before starting media-relay. Thank you again, Liviu *From:* Saúl Ibarra Corretgé s...@ag-projects.com *To:* OpenSIPS users mailling list users@lists.opensips.org *Sent:* Wed, May 25, 2011 6:55:00 PM *Subject:* Re: [OpenSIPS-Users] media-dispatcher and media relay connection problem Hi, On May 25, 2011, at 5:42 PM, Barsan Liviu wrote: Hi, I installed python-gnutls 1.2.1 from sources and got a little better, I restarted the server and was able to call and speak from one Blink client to another. But trying second (and several) time failed with same errors as I wrote: May 25 18:38:04 P4025 media-dispatcher[1755]: debug: Connection from relay at 80.97.161.39 May 25 18:38:04 P4025 media-dispatcher[1755]: debug: Issuing sessions command to relay at 80.97.161.39 May 25 18:38:04 P4025 media-dispatcher[1755]: error: Connection with relay at 80.97.161.39 was lost: Rehandshake was requested by the peer. May 25 18:38:04 P4025 media-dispatcher[1755]: error: Unhandled error in Deferred: May 25 18:38:04 P4025 media-relay[1758]: error: Could not connect to dispatcher at 80.97.161.39:25060 (retrying in 10 seconds): A TLS packet with unexpected length was received. May 25 18:38:04 P4025 media-dispatcher[1755]: RelayError: Relay at 80.97.161.39 disconnected Is the python-gnutls instable? What can I do to have a stable far-end NAT traversal server? Initially I intended to install rtpproxy, but client asked mediaproxy... If you are running different machines, did you upgrade python-gnutls in all of them? You can verify what version of python-gnutls you are running by typing the following: $ python import gnutls print gnutls.__version__ It should say 1.2.1. Also, completely stop and start MediaProxy, after this. -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips 1_6_X tls crash opensips
root@test:/opensips_1_6# opensips -V version: opensips 1.6.4-2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT 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. svnrevision: unknown @(#) $Id: main.c 7530 2010-12-13 19:07:53Z bogdan_iancu $ main.c compiled on 06:23:09 Apr 20 2011 with gcc 4.5.2 root@test:/opensips_1_6# gdb opensips /var/run/opensips/core GNU gdb (GDB) 7.0.1-debian Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i486-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /home/danip/opensips_1_6/opensips...done. [New Thread 13625] [New Thread 12825] [New Thread 13629] [New Thread 13627] [New Thread 13631] [New Thread 13628] [New Thread 13632] [New Thread 13653] [New Thread 13655] [New Thread 13656] [New Thread 13658] Core was generated by `/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid -m 64 -u opensips -g'. Program terminated with signal 11, Segmentation fault. #0 0x08156f5e in tls_connect (c=0x2e323862, poll_events=value optimized out) at tls/tls_server.c:331 331 LM_DBG(sending socket: %s:%d \n, (gdb) bt #0 0x08156f5e in tls_connect (c=0x2e323862, poll_events=value optimized out) at tls/tls_server.c:331 #1 0x081574e4 in ip_addr2a (c=0x2e323862, poll_events=value optimized out) at tls/../ip_addr.h:428 #2 tls_connect (c=0x2e323862, poll_events=value optimized out) at tls/tls_server.c:331 #3 0xb6fdceab in ?? () #4 0xb6f81598 in ?? () #5 0xb6f816c1 in ?? () #6 0xb6f89226 in ?? () #7 0xb6f79781 in ?? () #8 0xb6f73410 in ?? () #9 0xb6f7c392 in ?? () #10 0xb729a955 in ?? () #11 0xb75d1e7e in ?? () Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â
Hi, do you have news about this mediaproxy issues ? Thanks, Dani On 05/03/11 11:52, Dani Popa wrote: Ok, Thanks, Dani On Tue, May 3, 2011 at 10:00 AM, Saúl Ibarra Corretgé s...@ag-projects.com mailto:s...@ag-projects.com wrote: On 05/02/2011 10:58 PM, Dani Popa wrote: Hi, Do you have any news with this issues ? Unfortunately not. I didn't have time to go and fix this yet, sorry. -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â
Ok, Thanks, Dani On Tue, May 3, 2011 at 10:00 AM, Saúl Ibarra Corretgé s...@ag-projects.comwrote: On 05/02/2011 10:58 PM, Dani Popa wrote: Hi, Do you have any news with this issues ? Unfortunately not. I didn't have time to go and fix this yet, sorry. -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â
Hi,
Do you have any news with this issues ?
Thanks,
Dani
On Thu, Apr 21, 2011 at 3:31 PM, Dani Popa dani.p...@gmail.com wrote:
OK,
Thanks,
Dani
On 04/21/11 15:14, Saúl Ibarra Corretgé wrote:
I'm not talking abut binding ports for streams, i'm talking about stream
packets and bytes info on telnet localhost 25060.
I meant the statisticas that get printed in syslog after the call is
closed.
[{from_tag: 4fc7812b, start_time: 1303386789.09, call_id:
f233072fb063d5c554bebffe80248eba@0:0:0:0:0:0:0:0, duration: 24,
streams: [{status: active, caller_codec: G711u,
post_dial_delay: 3.49981117249, callee_codec: G711u,
caller_bytes: 0, start_time: 0, callee_packets: 0, callee_bytes:
0, caller_packets: 0, callee_remote: X.X.X.X:8752, end_time: 24,
caller_remote: X.X.X.X:5014, media_type: audio, callee_local:
X.X.X.X:50006, timeout_wait: 0, caller_local: X.X.X.X:50004}],
to_tag: a94c095b773be1dd6e8d668a785a9c848e314110, to_uri:
123456...@gigi.ro, caller_ua:
Jitsi1.0-beta1-nightly.build.3408Linux, callee_ua: Cantata,
from_uri: dani.p...@gigi.ro}]
And also, when mediaproxy send radius acounting request, it send with :
caller_bytes: 0, callee_packets: 0, callee_bytes: 0,
caller_packets: 0
This could be due to the bug with the netfilter integration which I need
to look into.
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] b2b_init_request('top hiding')
ok, Thanks, Dani On 04/21/11 11:10, Anca Vamanu wrote: Hi Dani, As Ovidiu said - b2b doesn't work with nat_traversal. The main limitation comes from the fact that nat_traversal uses dialog module and b2b does not work with dialog module yet. The reason is that dialog module was designed to store proxied dialog, not the ones where opensips is an endpoint. Indeed there isn't a clear note in documentation about this limitation - I will add it now. Regards, -- Anca Vamanu OpenSIPS Developer On 04/20/2011 06:26 PM, Anca Vamanu wrote: Hi Dani, Seems similar to something that we also hit.. but still not the same. Can you please paste the output of 'bt'** http://opensips.svn.sourceforge.net/viewvc/opensips/branches/1.6/modules/tm/uac.c?revision=7747view=markup in gdb? Regards, -- Anca Vamanu OpenSIPS Developer On 04/20/2011 03:11 PM, Dani Popa wrote: Hi, I have a problem using b2b_init_request with top hiding. When i receive 200 ok for invite, opensips crash with ERROR:nat_traversal:__dialog_confirmed: FAKED reply - exit. In core dump this is where opensips crash: #0 get_source_uri (dlg=0xb2b4bc84, type=8, _params=0xb70b3c20) at nat_traversal.c:968 968 snprintf(uri, 64, sip:%s:%d, ip_addr2a(msg-rcv.src_ip), msg-rcv.src_port); opensips info: version: opensips 1.6.4-2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT 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. svnrevision: unknown @(#) $Id: main.c 7530 2010-12-13 19:07:53Z bogdan_iancu $ main.c compiled on 06:23:09 Apr 20 2011 with gcc 4.5.2 Can someone give me a hint? Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â
Hi, yes, i was able to install it and run it, but i have some issues. I dont have stream statistics: caller_bytes,callee_bytes,caller_packets and callee_packets. Also, if i'm not sure if media timeout is working, because i tried to simulate a hang call (in the middle of call, i restart my hardphone) and call was not terminated with timeout after stream_timeout seconds. Indeed, i saw rtp packets in one way. Dani On 04/21/11 13:37, Saúl Ibarra Corretgé wrote: Hi, Thanks for the report, I was able to reproduce this on a Squeeze system. We need to adapt to changes in latest libnetfilter-conntrack. Did you try to install the Debian package from our repository? Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â
test kernel: [ 209.839147] ---[ end trace 8883c080e30a8b9c ]--- Thanks, Dani On 04/21/11 13:51, Saúl Ibarra Corretgé wrote: On 04/21/2011 12:44 PM, Dani Popa wrote: Hi, yes, i was able to install it and run it, but i have some issues. I dont have stream statistics: caller_bytes,callee_bytes,caller_packets and callee_packets. Also, if i'm not sure if media timeout is working, because i tried to simulate a hang call (in the middle of call, i restart my hardphone) and call was not terminated with timeout after stream_timeout seconds. Indeed, i saw rtp packets in one way. MediaProxy will not terminate the session is there are RTP packets flowing in at least one direction. Stream timeout only kicks in if RTP stops from both sides. About the statistics, can you paste what you get on syslog after a successful call with RTP? Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â
On 04/21/11 14:13, Saúl Ibarra Corretgé wrote:
On 04/21/2011 01:06 PM, Dani Popa wrote:
sure,
Apr 21 06:06:41 test media-relay[4903]:
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50012
Apr 21 06:06:41 test media-relay[4903]:
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50013
Apr 21 06:06:41 test media-relay[4903]:
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50014
Apr 21 06:06:41 test media-relay[4903]:
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50015
Apr 21 06:06:50 test media-relay[4903]: (Port 50012 Closed)
Apr 21 06:06:50 test media-relay[4903]: (Port 50013 Closed)
Apr 21 06:06:50 test media-relay[4903]: (Port 50014 Closed)
Apr 21 06:06:50 test media-relay[4903]: (Port 50015 Closed)
Don't you get teh statistics printed here?
I'm not talking abut binding ports for streams, i'm talking about stream
packets and bytes info on telnet localhost 25060.
[{from_tag: 4fc7812b, start_time: 1303386789.09, call_id:
f233072fb063d5c554bebffe80248eba@0:0:0:0:0:0:0:0, duration: 24,
streams: [{status: active, caller_codec: G711u,
post_dial_delay: 3.49981117249, callee_codec: G711u,
caller_bytes: 0, start_time: 0, callee_packets: 0, callee_bytes:
0, caller_packets: 0, callee_remote: X.X.X.X:8752, end_time: 24,
caller_remote: X.X.X.X:5014, media_type: audio, callee_local:
X.X.X.X:50006, timeout_wait: 0, caller_local: X.X.X.X:50004}],
to_tag: a94c095b773be1dd6e8d668a785a9c848e314110, to_uri:
123456...@gigi.ro, caller_ua:
Jitsi1.0-beta1-nightly.build.3408Linux, callee_ua: Cantata,
from_uri: dani.p...@gigi.ro}]
And also, when mediaproxy send radius acounting request, it send with :
caller_bytes: 0, callee_packets: 0, callee_bytes: 0,
caller_packets: 0
s
Also, i'm not very sure when(i'll make more tests and i'll come with
updates) and what conditions i get next kernel errors:
Apr 19 05:53:36 test kernel: [ 209.838816] ctnetlink v0.93: registering
with nfnetlink.
Apr 19 05:53:36 test kernel: [ 209.838957] [ cut here
]
Apr 19 05:53:36 test kernel: [ 209.838971] WARNING: at
/build/buildd-linux-2.6_2.6.37-1-i386-vxjyZA/linux-2.6-2.6.37/debian/build/source_i38
6_none/mm/page_alloc.c:1990 __alloc_pages_nodemask+0x17c/0x5ff()
Apr 19 05:53:36 test kernel: [ 209.838978] Hardware name: PowerEdge R310
Apr 19 05:53:36 test kernel: [ 209.838980] Modules linked in:
nf_conntrack_netlink xt_NOTRACK xt_tcpudp iptable_raw nfnetlink
iptable_nat nf_nat nf_conntrack_ipv4 nf_conntrack nf_defrag_ipv4
ip_tables x_tables loop snd_pcm snd_timer snd soundcore snd_page_alloc
evdev button tpm_tis tpm processor pcspkr dcdbas ghes thermal_sys hed
tpm_bios power_meter ext3 jbd mbcache sd_mod crc_t10dif sg sr_mod cdrom
ata_generic ata_piix libata mptsas mptscsih mptbase ehci_hcd
scsi_transport_sas usbcore scsi_mod bnx2 nls_base [last unloaded:
scsi_wait_scan]
Apr 19 05:53:36 test kernel: [ 209.839030] Pid: 2521, comm: media-relay
Not tainted 2.6.37-1-686-bigmem #1
Apr 19 05:53:36 test kernel: [ 209.839033] Call Trace:
Apr 19 05:53:36 test kernel: [ 209.839041] [c1036005] ?
warn_slowpath_common+0x6a/0x7b
Apr 19 05:53:36 test kernel: [ 209.839047] [c10986c8] ?
__alloc_pages_nodemask+0x17c/0x5ff
Apr 19 05:53:36 test kernel: [ 209.839052] [c1036023] ?
warn_slowpath_null+0xd/0x10
Apr 19 05:53:36 test kernel: [ 209.839058] [c10986c8] ?
__alloc_pages_nodemask+0x17c/0x5ff
Apr 19 05:53:36 test kernel: [ 209.839068] [c102cd47] ?
select_task_rq_fair+0x326/0x604
Apr 19 05:53:36 test kernel: [ 209.839071] [c1098b57] ?
__get_free_pages+0xc/0x17
Apr 19 05:53:36 test kernel: [ 209.839074] [c10be2c7] ?
__kmalloc_track_caller+0x32/0x127
Apr 19 05:53:36 test kernel: [ 209.839077] [f8d6c4da] ?
nlmsg_new+0xf/0x11 [nf_conntrack_netlink]
Apr 19 05:53:36 test kernel: [ 209.839080] [c11fbb07] ?
__alloc_skb+0x4c/0xda
Apr 19 05:53:36 test kernel: [ 209.839082] [f8d6c4da] ?
nlmsg_new+0xf/0x11 [nf_conntrack_netlink]
Apr 19 05:53:36 test kernel: [ 209.839085] [f8d6e228] ?
ctnetlink_conntrack_event+0x11e/0x3f2 [nf_conntrack_netlink]
Apr 19 05:53:36 test kernel: [ 209.839087] [f8d6c303] ?
nf_conntrack_eventmask_report+0x98/0xfb [nf_conntrack_netlink]
Apr 19 05:53:36 test kernel: [ 209.839090] [f8d6c468] ?
ctnetlink_del_conntrack+0xee/0x142 [nf_conntrack_netlink]
Apr 19 05:53:36 test kernel: [ 209.839094] [f8d36222] ?
nfnetlink_rcv_msg+0x12b/0x15c [nfnetlink]
Apr 19 05:53:36 test kernel: [ 209.839097] [f8d360f7] ?
nfnetlink_rcv_msg+0x0/0x15c [nfnetlink]
Apr 19 05:53:36 test kernel: [ 209.839102] [c121bd58] ?
netlink_rcv_skb+0x2d/0x72
Apr 19 05:53:36 test kernel: [ 209.839105] [f8d360f1] ?
nfnetlink_rcv+0x18/0x1e [nfnetlink]
Apr 19 05:53:36 test kernel: [ 209.839107] [c121bbac] ?
netlink_unicast+0xba/0x10e
Apr 19 05:53:36 test kernel: [ 209.839109] [c121c6b0] ?
netlink_sendmsg+0x23d/0x256
Apr 19 05:53:36 test kernel: [ 209.839112] [c11f5326] ?
__sock_sendmsg+0x48/0x4e
Apr 19 05:53:36 test kernel: [ 209.839114] [c11f558f] ?
sock_sendmsg+0x78/0x8f
Apr 19 05:53:36 test kernel
Re: [OpenSIPS-Users] mediaproxy build fail: libiptc.c:63:8: error: redefinition of â
OK,
Thanks,
Dani
On 04/21/11 15:14, Saúl Ibarra Corretgé wrote:
I'm not talking abut binding ports for streams, i'm talking about stream
packets and bytes info on telnet localhost 25060.
I meant the statisticas that get printed in syslog after the call is
closed.
[{from_tag: 4fc7812b, start_time: 1303386789.09, call_id:
f233072fb063d5c554bebffe80248eba@0:0:0:0:0:0:0:0, duration: 24,
streams: [{status: active, caller_codec: G711u,
post_dial_delay: 3.49981117249, callee_codec: G711u,
caller_bytes: 0, start_time: 0, callee_packets: 0, callee_bytes:
0, caller_packets: 0, callee_remote: X.X.X.X:8752, end_time: 24,
caller_remote: X.X.X.X:5014, media_type: audio, callee_local:
X.X.X.X:50006, timeout_wait: 0, caller_local: X.X.X.X:50004}],
to_tag: a94c095b773be1dd6e8d668a785a9c848e314110, to_uri:
123456...@gigi.ro, caller_ua:
Jitsi1.0-beta1-nightly.build.3408Linux, callee_ua: Cantata,
from_uri: dani.p...@gigi.ro}]
And also, when mediaproxy send radius acounting request, it send with :
caller_bytes: 0, callee_packets: 0, callee_bytes: 0,
caller_packets: 0
This could be due to the bug with the netfilter integration which I
need to look into.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] b2b_init_request('top hiding')
Hi, I have a problem using b2b_init_request with top hiding. When i receive 200 ok for invite, opensips crash with ERROR:nat_traversal:__dialog_confirmed: FAKED reply - exit. In core dump this is where opensips crash: #0 get_source_uri (dlg=0xb2b4bc84, type=8, _params=0xb70b3c20) at nat_traversal.c:968 968 snprintf(uri, 64, sip:%s:%d, ip_addr2a(msg-rcv.src_ip), msg-rcv.src_port); opensips info: version: opensips 1.6.4-2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT 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. svnrevision: unknown @(#) $Id: main.c 7530 2010-12-13 19:07:53Z bogdan_iancu $ main.c compiled on 06:23:09 Apr 20 2011 with gcc 4.5.2 Can someone give me a hint? Thanks, Dani ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy KBIn and KBOut
Hi,
As you can see, callee_bytes and caller_bytes are 0 during the call.
Media relay version is media-relay 2.4.4. network hardware is Ethernet
controller: Broadcom Corporation NetXtreme II BCM5716 Gigabit Ethernet
(rev 20). The same issue i have with media-relay 2.4.3 on the same
machine. I wondering if is a network card driver issue or kernel
issue(if so, i'm dont know how to make troubleshooting, where should i
see the callee_bytes and caller_bytes in kernel stats).
Dani
On 04/18/11 10:43, Saúl Ibarra Corretgé wrote:
On 04/15/2011 02:42 PM, Dani Popa wrote:
Hi,
Mediaproxy radius request does not populate Kbin and Kbout. Also i tried
to see sessions on port 25061 and also there callee_bytes and
caller_bytes are 0.
opensips:~# telnet localhost 25061
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
sessions
[]
sessions
[{from_tag: 2a476a89, start_time: 1302867244.13, call_id:
f2c9ca5f8ffe0e5ea422b57d7b93aa28@0:0:0:0:0:0:0:0, duration: 5,
streams: [{status: active, caller_codec: G711u,
post_dial_delay: 3.43099498749, callee_codec: G711u,
caller_bytes: 0, start_time: 0, callee_packets: 0, callee_bytes:
0, caller_packets: 0, callee_remote: 192.5.32.12:8152, end_time:
5, caller_remote: 189.8.7.161:5105, media_type: audio,
callee_local: 167.14.19.35:50010, timeout_wait: 0, caller_local:
167.14.19.35:50008}], to_tag:
a94c095b773be1dd6e8d668a785a9c8498c8c4db, to_uri:
cal...@domain.org, caller_ua:
Jitsi1.0-beta1-nightly.build.3408Linux, callee_ua: Cisco,
from_uri: cal...@domain.org}]
[{from_tag: 2a476a89, start_time: 1302867244.13, call_id:
f2c9ca5f8ffe0e5ea422b57d7b93aa28@0:0:0:0:0:0:0:0, duration: 11,
streams: [{status: active, caller_codec: G711u,
post_dial_delay: 3.43099498749, callee_codec: G711u,
caller_bytes: 0, start_time: 0, callee_packets: 0, callee_bytes:
0, caller_packets: 0, callee_remote: 192.5.32.12:8152, end_time:
11, caller_remote: 189.8.7.161:5105, media_type: audio,
callee_local: 167.14.19.35:50010, timeout_wait: 0, caller_local:
167.14.19.35:50008}], to_tag:
a94c095b773be1dd6e8d668a785a9c8498c8c4db, to_uri:
cal...@domain.org, caller_ua:
Jitsi1.0-beta1-nightly.build.3408Linux, callee_ua: Cisco,
from_uri: cal...@domain.org}]
Can someone give me a hint ?
Thanks,
Dani
Hi Dani,
What versions of the software are you using? KBIn and KBOut were
renamed quite some time ago. CDRTool will render Acct-Input-Octets and
Acct-Output-Octets fields, which are populated by MediaProxy using the
caller_bytes and callee_bytes attributes which you can see on the
statistics.
I just ran a quick test and it works here. Do the calls on the trace
you pasted have audio at all?
Regards,
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] mediaproxy KBIn and KBOut
Hi,
Mediaproxy radius request does not populate Kbin and Kbout. Also i tried
to see sessions on port 25061 and also there callee_bytes and
caller_bytes are 0.
opensips:~# telnet localhost 25061
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
sessions
[]
sessions
[{from_tag: 2a476a89, start_time: 1302867244.13, call_id:
f2c9ca5f8ffe0e5ea422b57d7b93aa28@0:0:0:0:0:0:0:0, duration: 5,
streams: [{status: active, caller_codec: G711u,
post_dial_delay: 3.43099498749, callee_codec: G711u,
caller_bytes: 0, start_time: 0, callee_packets: 0, callee_bytes:
0, caller_packets: 0, callee_remote: 192.5.32.12:8152, end_time:
5, caller_remote: 189.8.7.161:5105, media_type: audio,
callee_local: 167.14.19.35:50010, timeout_wait: 0, caller_local:
167.14.19.35:50008}], to_tag:
a94c095b773be1dd6e8d668a785a9c8498c8c4db, to_uri:
cal...@domain.org, caller_ua:
Jitsi1.0-beta1-nightly.build.3408Linux, callee_ua: Cisco,
from_uri: cal...@domain.org}]
[{from_tag: 2a476a89, start_time: 1302867244.13, call_id:
f2c9ca5f8ffe0e5ea422b57d7b93aa28@0:0:0:0:0:0:0:0, duration: 11,
streams: [{status: active, caller_codec: G711u,
post_dial_delay: 3.43099498749, callee_codec: G711u,
caller_bytes: 0, start_time: 0, callee_packets: 0, callee_bytes:
0, caller_packets: 0, callee_remote: 192.5.32.12:8152, end_time:
11, caller_remote: 189.8.7.161:5105, media_type: audio,
callee_local: 167.14.19.35:50010, timeout_wait: 0, caller_local:
167.14.19.35:50008}], to_tag:
a94c095b773be1dd6e8d668a785a9c8498c8c4db, to_uri:
cal...@domain.org, caller_ua:
Jitsi1.0-beta1-nightly.build.3408Linux, callee_ua: Cisco,
from_uri: cal...@domain.org}]
Can someone give me a hint ?
Thanks,
Dani
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
