Re: [OpenSIPS-Users] My mediaproxy and callcontrol do not output any log on Debian.
On Feb 7, 2014, at 3:06 AM, hutx wrote: Hi All, I do not know how to make my mediaproxy and callcontrol output log. I use Debian3.2.0. The default log level for MediaProxy is debug, so you should see it starting. Don't you see any entries on syslog? What happens if you run the binaries (media-dispatcher and media-relay) manually? -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Crash of Opensips
Perfect Sean ! The crash was also fixed in OpenSIPS by Liviu - even if config is wrong, it should not crash at all ! Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.02.2014 03:03, hutx wrote: Hi, Yes, I modified my configuration accordingly. This problem disappears. Thank you so much for your great help. Sean On Thursday, February 6, 2014 10:09 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hello, Based on the BT, I guess you are using the acc_db_request() function from the local_route, right ? If so, have you defined a db_url for the acc module ? I see the table param is NULL and this may happen only if you do not set the db_url for the module. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 06.02.2014 15:26, hutx wrote: Hi All, I am playing with opensips + call control + cdrtool. When I set MaximumSession=60 and the opensips tries to hang up two call legs upon timeout, opensips experiences crash. The bt of core file is below. Any help and opinion are appreciated. Best Regards, Sean /bt of Core file/ Program terminated with signal 11, Segmentation fault. #0 0xb6b7bc96 in w_acc_db_request (rq=0xb7156740, comment=0xb7185360, table=0x0) at acc_logic.c:213 213 int table_len = strlen(table); (gdb) bt #0 0xb6b7bc96 in w_acc_db_request (rq=0xb7156740, comment=0xb7185360, table=0x0) at acc_logic.c:213 #1 0x0805f5b5 in do_action (a=0xb7182718, msg=0xb7156740) at action.c:1715 #2 0x08064ea4 in run_action_list (a=0xb7182718, msg=0xb7156740) at action.c:171 #3 0x08062851 in do_action (a=0xb7182784, msg=0xb7156740) at action.c:1010 #4 0x08064891 in run_action_list (a=Unhandled dwarf expression opcode 0xf3 ) at action.c:171 #5 run_actions (a=Unhandled dwarf expression opcode 0xf3 ) at action.c:136 #6 0x08065276 in run_actions (a=0xb7182784, msg=0xb7156740) at action.c:129 #7 run_top_route (a=0xb7182784, msg=0xb7156740) at action.c:211 #8 0xb713672c in t_uac (method=0xbfc942d8, headers=0xbfc942d0, body=0x0, dialog=0xb7195c64, cb=0xb6adf606 bye_reply_cb, cbp=0xb2bd6e94, release_func=0) at uac.c:313 #9 0xb7137dd3 in req_within (method=0xbfc942d8, headers=0xbfc942d0, body=0x0, dialog=0xb7195c64, completion_cb=0xb6adf606 bye_reply_cb, cbp=0xb2bd6e94, release_func=0) at uac.c:476 #10 0xb6ae081b in send_leg_bye (dlg=0xb2bd6e94, extra_hdrs=0x0) at dlg_req_within.c:352 #11 dlg_end_dlg (dlg=0xb2bd6e94, extra_hdrs=0x0) at dlg_req_within.c:418 #12 0xb6ae1b05 in mi_terminate_dlg (cmd_tree=0xb7195bac, param=0x0) at dlg_req_within.c:475 #13 0xb6a6ea75 in run_mi_cmd (rx_sock=8, tx_sock=9) at ../../mi/mi.h:109 #14 mi_datagram_server (rx_sock=8, tx_sock=9) at datagram_fnc.c:520 #15 0xb6a6fda4 in datagram_process (rank=0) at mi_datagram.c:311 #16 0x080e41d8 in start_module_procs () at sr_module.c:586 #17 0x0805b6e0 in main_loop (argc=13, argv=0xbfc946a4) at main.c:840 #18 main (argc=13, argv=0xbfc946a4) at main.c:1598 (gdb) /// ___ 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] use branches with uac_replace_from?
Hello,
If you plan to change the FROM in different ways for each branch, you
need to do the change in a branch_route and not in the request route -
whatever changes you do in request route will be propagate in all
branches. The changes you do in branch route will be vizible only in
that branch. See:
http://www.opensips.org/Documentation/Script-Routes-1-10#toc2
So, in your script store the new FROM into an AVP variable and arm the
branch route - in that branch, if AVP is set, use it as input for
uac_replace_from().
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 27.01.2014 14:05, Edwin Haselhoff wrote:
Hello Everyone,
I want to use 'uac_replace_from' multiple times in my script (to use
for a failover situation). This is a simple example of what I want to do:
route[1]
{
if (method==INVITE)
{
record_route();
uac_replace_from(,sip:anonymous@anonymous.invalid);
append_hf(Remote-Party-ID:
sip:anonymous@anonymous.invalid;party=calling;screen=yes;privacy=full\r\n);
append_hf(P-Asserted-Identity:
sip:anonymous@anonymous.invalid\r\n);
append_hf(Privacy: id\r\n);
t_on_failure(1);
t_relay();
}
}
route[2]
{
if (method==INVITE)
{
record_route();
uac_replace_from(,sip:a...@xx.xxx.xx);
append_hf(Remote-Party-ID:
sip:a...@xx.xxx.xx;party=calling;screen=no;privacy=off\r\n);
append_hf(P-Asserted-Identity: sip:a...@xx.xxx.xx\r\n);
t_on_failure(1);
t_relay();
exit;
}
}
failure_route[1]
{
route(2);
}
Reading the docs and forums I tried to use branches but I think I
don't understand the explanation / examples of the branches well
enough :)
Can anyone explain / give me a hint how to change my script so I can
drop the first 'uac_replace_from' and 'append_hf'?
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Re: [OpenSIPS-Users] Fwd: Account expiration notification
Hello, There is no such mechanism - as said, subscribing to a user's presentity (and the whole presence stuff) is completely independent from the registration mechanism. There is no need to register in order to have presence working. The subscriptions are time based (they have an expire, exactly like registration), so , even if the UA crashed, the subscription will timeout after some time (exactly like the registration, but independent and maybe with different timeout value). So, the crashed UA will go offline at some point (because of the subscribe expire) - there is no need for a parallel mechanism (the registration expire). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.02.2014 09:12, Bogdan Chifor wrote: Yes, this is correct. I would like to know if there is a mechanism to notify all the subscribed users when an account expires. Thank you, Chifor Bogdan On Thu, Feb 6, 2014 at 10:15 AM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hello, You are mentioning 2 different things here : (1) presence for notifications and (2) registration status of the user. Presence and registration are not related at all (unless using the pua_usrloc module). The closest I can understand is: you have presence between UAs and one UA crashes - the other UAs are still seeing it online (and not offline) - is this correct ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 30.01.2014 14:52, Bogdan Chifor wrote: Hello, I have a presence enabled Opensips server and am facing the following issue. My SIP client sometimes stops in an unfriendly manner (crashes etc..) and the account remains active on the registrar side (until the registration period expires, of course). I would like to notify all the users which are subscribed to this user when the account expired. Any help would be appreciated. Thank you, Bogdan Chifor. ___ 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
[OpenSIPS-Users] Registrar - max_contacts and 503 logging
Hello, In the registrar module I set the max_contacts parameter. Save function work as expected, but I did not manage to log the 503 status when max contacts is reached. It seems configuration after save function is not used by Opensips in this case. Is this an expected behavior and could I log the 503 in this way or another ? PS: I found an appropriate log at INFO level, but at the moment I would rather keep NOTICE level. Indeed I could use INFO level if there is no other way to check save function's return. PS2: Opensips version is 1.8.0. Regards, Adrien Martin signature.asc Description: OpenPGP digital signature ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Manipulating Contact header in locally generated 200 OK for a REGISTER
Hello, The SIP RFC3261 says that only the registered URI contact (along with the URI params) are to be stored by the registrar server. The header params are not to be stored. So, should I understand the that GSMA specs may conflict with the SIP specs ??? BTW, do you have an example of such a Contact hdr ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.01.2014 08:00, Java Jalwa wrote: Hello experts, I am using the registrar module and the save() function automatically sends a 200 OK for the REGISTER. However the Contact header built by the registrar module removes all the feature tags such as +g.oma.sip-im, +sip.instance etc. To be spec compliant with certain GSMA specs I need to keep these feature tags / parameters in the Contact header of 200 response, while at the same time want the registrar module to add expires and received= params. append_to_reply() is not something that is useful for achieving this since it adds a new header. Is there anyway to achieve this without having to hack the registrar module ? Using the r flag in save method, would force the registrar module to not send a reply, however if I use sl_send_reply(200, OK) after calling save(location,fr); the 200 OK does not contain the Contact header at all. Any suggestions as to how I can achieve this ? Thanks. ___ 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] Initializing SIP messages from routing
Hello, No sure what CRBT stands for, but it looks to me that you need to use B2B module - what you are trying to do is something more than simply proxying a call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.02.2014 19:00, H Yavari wrote: Hi Bogdan, thanks for your answer. I want to implement CRBT. For this I want when the invite received,I send an invite to media server and play something. I can do this with B2BUA? I write a module for this or do with script? script running has side effect on performance when load is high? Regards, H.Yavari Hello, Typically you process in OpenSIPS script an incoming request (and you fwd or reply it). It is unusual to generate a new request while processing another one. May I ask about your scenario ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com http://www.opensips-solutions.com/ On 28.01.2014 10:02, H Yavari wrote: Hi to all openSIPSer, I want to initialize a sip message from routing, is it possible? If only way that do this is writing script, is this efficient when load on openSIPS is high? If answer is yes, can you give me some example to how do this? Regards, H.Yavari ___ 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
Re: [OpenSIPS-Users] Registrar - max_contacts and 503 logging
Hello, Not sure a bug or an undocumented feature, but save() function, in case of internal error, is stopping the script. Assuming we change that (to continue the script execution), you will failure from save() without actually knowing the error itself. I assume you are looking the error code/reason, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 07.02.2014 17:38, Adrien Martin wrote: Hello, In the registrar module I set the max_contacts parameter. Save function work as expected, but I did not manage to log the 503 status when max contacts is reached. It seems configuration after save function is not used by Opensips in this case. Is this an expected behavior and could I log the 503 in this way or another ? PS: I found an appropriate log at INFO level, but at the moment I would rather keep NOTICE level. Indeed I could use INFO level if there is no other way to check save function's return. PS2: Opensips version is 1.8.0. Regards, Adrien Martin ___ 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] topology hiding
The traffic goes from the server making the call to the opensips proxy to the
carrier. The /304 doesn't exist in the original invite from the server that
makes the call. Do you want to see the packets from the calling server to the
opensips box to the carrier?
But most importantly, I'm still having trouble getting topology hiding to work.
I made the change you suggested, but it appears that opensips is throwing away
all responses it's receiving from the carrier, which means that opensips thinks
the call never got set up and the invite was never accepted, and then it
eventually responds 408 to the original calling server. In fact the invite WAS
accepted, and the carrier sent back a 200 ok, but opensips threw it away and
never acknowledged it. The outgoing invite from the opensips server looks
correct -- opensips has modified the route and contact headers to its own IP,
but obviously it doesn't know what to do with the responses that are coming
back to it.
The /304 issue is odd and could be causing us issues with certain carriers, but
right now I'd like to get topology hiding working with even ONE carrier, and I
can't do that yet. All I did was add the section you showed in your response --
if (loose_route() || match_dialog()) {
if ($DLG_status==NULL) {
xlog( cannot match request to a dialog \n);
# something wrong - might want to drop such requests
}
... as a replacement to ...
if (loose_route()) {
And I added --
if (is_method(INVITE)) {
topology_hiding();
}
... right after --
# account only INVITEs
if (is_method(INVITE)) {
setflag(ACC_DO); # do accounting
}
All this is from the stock config file, with only my listen IPs and a couple
aliases added, and obviously the dialog module loaded as well to allow for
topology hiding.
Thanks!
-BJ
- Original Message -
From: Vlad Paiu vladp...@opensips.org
To: users@lists.opensips.org
Sent: Wednesday, February 5, 2014 6:19:41 AM
Subject: Re: [OpenSIPS-Users] topology hiding
Hello,
The sequential processing part is a little bit wrong - you should have
if (loose_route() || match_dialog()) {
if ($DLG_status==NULL) {
xlog( cannot match request to a dialog \n);
# something wrong - might want to drop such requests
}
Can you please also post a trace of the traffic flow when the Route
header gets that bogus \304 header ? Trying to replicate this on my side
and see what's wrong.
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 03.02.2014 20:36, BJ Quinn wrote:
Oh and the only manual manipulation of the route headers was an attempt to
get rid of that \304 in the header.
I think the \304 thing may be a red herring for now. I still can't get the
topology hiding to work. Below is my config file. It's literally the default
config file with nothing changed but I've put in my IP address on the listen
line, added a couple of aliases, added UAC module to try to change the from
header (that works) and the dialog module and a couple of modifications to
the route to make topology hiding work (not working for me).
Am I putting this in the wrong part of the route?
Thx
-BJ Quinn
---
debug=3
log_stderror=no
log_facility=LOG_LOCAL0
fork=yes
children=4
auto_aliases=no
listen=udp:xx.xx.xx.9:5060
disable_tcp=yes
disable_tls=yes
alias=xx.xx.xx.76:5060
alias=xx.xx.xx.77:5060
mpath=/usr/lib64/opensips/modules
loadmodule signaling.so
loadmodule sl.so
loadmodule tm.so
modparam(tm, fr_timer, 5)
modparam(tm, fr_inv_timer, 30)
modparam(tm, restart_fr_on_each_reply, 0)
modparam(tm, onreply_avp_mode, 1)
loadmodule rr.so
modparam(rr, append_fromtag, 0)
loadmodule maxfwd.so
loadmodule sipmsgops.so
loadmodule mi_fifo.so
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
modparam(mi_fifo, fifo_mode, 0666)
loadmodule uri.so
modparam(uri, use_uri_table, 0)
loadmodule usrloc.so
modparam(usrloc, nat_bflag, NAT)
modparam(usrloc, db_mode, 0)
loadmodule registrar.so
modparam(registrar, tcp_persistent_flag, TCP_PERSISTENT)
loadmodule acc.so
modparam(acc, early_media, 0)
modparam(acc, report_cancels, 0)
modparam(acc, detect_direction, 0)
modparam(acc, failed_transaction_flag, ACC_FAILED)
modparam(acc, log_flag, ACC_DO)
modparam(acc, log_missed_flag, ACC_MISSED)
# added to rewrite from header
loadmodule uac.so
loadmodule uac_auth.so
modparam(uac,restore_mode,manual)
#added for topology hiding
loadmodule dialog.so
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
if (has_totag()) {
if (loose_route()) {
# added for topology hiding
if ($DLG_status==NULL !match_dialog() ) {
xlog( cannot match request to a dialog \n);
}
#/added for topology hiding
if (is_method(BYE)) {
setflag(ACC_DO);
setflag(ACC_FAILED);
} else if (is_method(INVITE)) {
record_route();
}
