[OpenSIPS-Devel] [NEW] Simplify your script with script_helper module
Hello all, OpenSIPS 1.11 has a new addition, the script_helper module . The purpose of this new module is to help you *start in an easy way with the OpenSIPS script*. *The learning curve gets milder*as the beginners can start with a simplified format of the script - the idea is to shift the main focus on routing the SIP initial requests (which define the service logic), while transparently handling a lot of standard SIP scripting logic, in order to both save script coding time and mitigate potential SIP scripting errors. This tutorial offers both a brief overview on the features of the new module and an example script. The current features of the module: 1) *transparent handling of SIP sequential requests* * sequential requests are properly and silently handled - they will not hit the script at all anymore for initial SIP requests, the module performs record-routing (i.e. OpenSIPS adds itself in the signaling path using a Record-Route header), after which the main route is triggered, just as before * for sequential SIP requests, the module is automatically doing loose_route() for proper routing. * optionally a special route may be run just before all sequential requests are routed out 2) *automatic dialog support* * automatically create dialog support for the INVITE-based sessions * for sequential requests within a dialog, a complete dialog matching (Call-ID, From-tag, To-tag) will be attempted if the quick 'did'-based Record-Route parameter matching fails A nice tutorial with script examples can be found at: http://www.opensips.org/Documentation/Tutorials-ScriptHelper-1-11 The module documentation can be found at: http://opensips.org/html/docs/modules/1.11.x/script_helper.html As future directions for the module, we are looking to include support for: * auto authentication * auto NAT traversal * auto registration handling Enjoy, -- Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [opensips] [RFC] An initial attempt of porting rtpproxy-ng module from your twin project to OpenSIPS. (#152)
Bogdan, about 10 days ago the mediaproxy-ng project was renamed into rtpengine. What about to add new module with this name? As for me, I'd like to use rtpengine for a long time, but I've never had a module. And I'm unhappy that we will have this module only in trunk, but not in 1.11. -- Nick 2014-02-21 19:33 GMT+04:00 Bogdan Andrei IANCU notificati...@github.com: @lemenkov https://github.com/lemenkov thanks for the update...I'm a but confused and puzzled. .The media engine from sipwise is called mediaproxy-ng, but has noting to do to with mediaproxy from AG-projects (not even a fork) - how comes this ??? Further more, why to name the module in OpenSIPS rtpproxy-ng, while it has nothing to do with the RTPproxy media relay ?? It is a lot of confusion and IMHO we should try to clear it up before considering any code. Regards, Bogdan — Reply to this email directly or view it on GitHubhttps://github.com/OpenSIPS/opensips/pull/152#issuecomment-35740773 . ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [opensips] opensips 1.11 crash on startup (#192)
If rtpproxy_engage() is called without any parameters, opensips crashes:
#0 fixup_spve (param=0x0) at mod_fix.c:840
s = {s = 0x0, len = 0}
gp = 0x7fbfff3067d8
__FUNCTION__ = fixup_spve
If there are no params, the fixup function should not be called.
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[OpenSIPS-Devel] [opensips] opensips 1.11 seg fault (#193)
Having a problem with a segmentation fault in avp.
Version info
version: opensips 1.11.0beta-notls (x86_64/linux)
flags: STATS: On, USE_TCP, DISABLE_NAGLE, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, DBG_QM_MALLOC, DBG_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: fad3491
@(#) $Id$
main.c compiled on 15:01:32 Apr 7 2014 with gcc 4.1.2
--- BT FULL
Program terminated with signal 11, Segmentation fault.
#0 destroy_avp_list_unsafe (list=0x2b85c692eb68) at usr_avp.c:438
438 avp = avp-next;
(gdb) bt full
#0 destroy_avp_list_unsafe (list=0x2b85c692eb68) at usr_avp.c:438
avp = 0x4030303033353332
__FUNCTION__ = destroy_avp_list_unsafe
#1 0x2b85c355fad1 in mi_tm_uac_dlg (cmd_tree=0x0, param=value optimized
out) at mi.c:457
rpl_tree = value optimized out
node = 0x80ae60
pruri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len =
-274025480}, host = {s = 0x7fffefaab588 \001Ϫ\357\377\177, len = 4400917},
port = {s = 0x87b6678b Address 0x87b6678b out of bounds, len = -974857576},
params = {s = 0x2b85c5b9e420 \003, len = 0}, headers = {
s = 0x0, len = -274025088}, port_no = 10794, proto = 0, type =
ERROR_URI_T, transport = {
s = 0x43666d
\211D$$\351h\337\377\377\220\220\220\220\220\220\220\220\220\220H\213G\020\303H\205\377u\006\270\377\377\377\377\303H\205\366t\365H\211~\bH\213G\bH\211\006H\205\300u\005\353\vH\211\006H\213,
len = -274025354}, ttl = {s = 0x7fffefaab477 , len = 0},
user_param = {s = 0xbdaa010 \230$\255, incomplete sequence \373,
len = 5571712}, maddr = {s = 0x40f89b H\203\304\b\303\377\065B\t9, len = 1},
method = {s = 0x7fffefaacf62 , len = -2018089077}, lr = {s = 0x3705a1cbc0 ,
len = 0}, r2 = {s = 0x0, len = -274025088},
gr = {s = 0x0, len = 0}, transport_val = {s = 0x3705c1d994
\211\307\350uZ\001, len = 4260240}, ttl_val = {s = 0x7fffefaab588
\001Ϫ\357\377\177, len = 0}, user_param_val = {s = 0x43342b
AWAVAUATUSH\203\354HA\211\376I\211\367\211=\307S, len = 94489536}, maddr_val
= {
s = 0xab5d2627fddc67cb Address 0xab5d2627fddc67cb out of bounds,
len = 0}, method_val = {s = 0x7fffefaab580 \t, len = 0}, lr_val = {s = 0x0,
len = 309777275}, r2_val = {s = 0xab5d2610f81dbe84 Address 0xab5d2610f81dbe84
out of bounds, len = 0}, gr_val = {s = 0x0,
len = 0}, u_name = {{s = 0x550480
L\211d$\340L\211l$\350L\215%o\373$, len = -274025080}, {s = 0x9 Address 0x9
out of bounds, len = 0}, {s = 0x0, len = 4260240}, {s = 0x7fffefaab580 \t,
len = 0}, {s = 0x4101b9 \364\220\220H\203\354\bH\213\005\021,
len = -274025096}}, u_val = {{s = 0x0, len = 9}, {s =
0x7fffefaacf01 /usr/local/sbin/opensips, len = -274018534}, {s =
0x7fffefaacf1d /var/run/opensips.pid, len = -274018509}, {s = 0x7fffefaacf36
/usr/local/etc/opensips/opensips.cfg, len = -274018469}, {
s = 0x7fffefaacf5e 4096, len = -274018461}}, u_params_no =
53094}
pnexthop = {user = {s = 0x6a8 Address 0x6a8 out of bounds, len =
-1017699944}, passwd = {s = 0x400 Address 0x400 out of bounds, len =
-1017701989}, host = {s = 0x10002 Address 0x10002 out of bounds, len
= -1015415552}, port = {s = 0x7 Address 0x7 out of bounds,
len = -977668636}, params = {s = 0x2b85c5b9e420 \003, len =
-963456288}, headers = {s = 0x2b85c63e8148 нPƅ+, len = 8000888}, port_no =
34400, proto = 127, type = ERROR_URI_T, transport = {s = 0x2b85c357361c
\377\017\204, incomplete sequence \375,
len = -1015415552}, ttl = {s = 0x2b85c6427d48 \367\001, len =
8000900}, user_param = {s = 0x7f8660 , len = -274025848}, maddr = {s =
0x7a1580 , len = 8000900}, method = {s = 0x7fffefaab288 , len = 8000896},
lr = {s = 0x7fffefaab288 , len = 0}, r2 = {
s = 0x205812992 Address 0x205812992 out of bounds, len = 1430},
gr = {s = 0x0, len = 0}, transport_val = {s = 0x0, len = -1294834576}, ttl_val
= {s = 0x0, len = 0}, user_param_val = {s = 0x553ed0a0 Address 0x553ed0a0 out
of bounds, len = 1430}, maddr_val = {
s = 0x4af83f H\213\025\t1, len = 99963304}, method_val = {s =
0x8 Address 0x8 out of bounds, len = -974857448}, lr_val = {s =
0x2b85b2d26418 \001, len = -1294627440}, r2_val = {s = 0xa000a Address
0xa000a out of bounds, len = 5662019}, gr_val = {
s = 0x49002c
Re: [OpenSIPS-Devel] [opensips] RTP Proxy C (Copy/Fork) audio (#189)
can anyone help? i have issued the command FORKAUDIOTOSERVER C 3c37a825927c-25qfdoh64vxg@snom360-000413239ECB 192.168.5.170:57313 12ul3kjwoe;1 i receive a response of (0) zero suggesting everything is ok. while running tcpdump to monitor the rtp streams during a call, I can see the rtp stream from my phone to the destination and back. I can also see the RTP file being created on the box but I do not see any attempt to sent rtp to the forked server. --- Reply to this email directly or view it on GitHub: https://github.com/OpenSIPS/opensips/issues/189#issuecomment-39812441___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
