[OpenSIPS-Users] CDRTool install
Hi everyone, I've started configuring CDRTool 8.0.10, FreeRadius 2.1.10, CallControl 2.0.8 and OpenSIPS 1.6 but I've encountered some problems: all the servers have installed successfully (I can start them) but after I've followed the instructions from http://cdrtool.ag-projects.com/wiki/Install I've realized that they refer to older versions of FreeRadius and OpenSIPS. The documentation of FreeRadius for MySQL's configuration (http://wiki.freeradius.org/SQL_HOWTO) seems to be different than the one from CDRTool's page (the section that describes the installation of FreeRadius). The patch CDRTool/setup/radius/FreeRadius/freeradius.patch doesn't seem to fit on all files of the latest version of FreeRadius. Also, the settings for OpenSIPS seem to be for 1.4. Which versions of servers should I use, those from CDRTool's page or the latest ones ? Can I bypass FreeRadius and use only OpenSIPS for authentication ? Are CDRTool and Call Control built to work this way ? I'm not concerned about security and I'd prefer not to use FreeRadius in the beginning if possible. Thanks, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] receive port during call process
Hello! I have a such problem. Opensips using 2 ports One – 5068 for client which must register on Opensips Second – 5060 for all other clients. 1) Client А registering on Opensips (socket: udp:2.2.2.2:5068). Client A is behind NAT. 2) Client А receives incoming call (via lookup() function). 3) Call has such way Cisco (source port 1 sends INVITE to port 5060 of Opensips) – Opensips (receives INVITE from Cisco to port 5060 and sends the INVITE to client from source port 5068 to some client`s port) – Client A. Everything work fine until client А answers. Then Client A sends to Opensips (port 5068) 200 OK, Opensips retransmit it to Cisco (from port 5060 to 1), Cisco sends to Opensips ACK (from port 1 to 5060) AND Opensips retransmit this ACK to client`s port FROM PORT 5060, BUT NOT 5068. This ACK didn`t reach client A because of NAT. Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] TEXTOPS module
Thank you Bogdan, that is working -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Monday, December 20, 2010 5:27 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] TEXTOPS module Hi Denis, the best way to do it is via branch_route (http://www.opensips.org/Resources/DocsCoreRoutes16#toc2) . Whatever changes you do there will be applied only for that particular branch and not for all branches. When you do changes in the request route, the changes will be applied to all future branches ! So, do something like this: - in request route (first time) put the RPID, PAI and FROM (new vals ) in 2 different AVPS (according to the first selected GW) - arm a branch route and failure route - do t_relay() - this will trigger the branch route and you can do the changes to the messages ( as you do it now) - if you end up in failure route - set new values for the 3 AVPs, reflecting the new destination - do t_relay() - triggers branch route, etc.. Regards, Bogdan Denis Putyato wrote: Thank you Bogdan for your answer. Now I understood that apply changes is a bad idea. But during process a call I have to make some changes to INVITE message. For example, I need to add Remote-Party-ID (RPI) and/or P-Asserted-ID (PAI) and make uac_replace_from(). If I make it for the first time everything fine. But if I need then change these fields (via subst or uac_replace_from() again)(for example, some gateways fails and cannot accepts call, I use use_next_gw() of d_routing module and MUST change callerid information) then my tests show that during, for example, second time call of uac_replace_from() there are two uri in From: header field (as you understand that is wrong), or if I make subst() of RPI or PAI then second header RPI and PAI appear in addition of first headers which I added (or subst) before. And to avoid this I make signaling loop. New INVITE process as a new message with modified early headers, so I can change it again. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Monday, December 20, 2010 4:07 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] TEXTOPS module Hello Denis, So far there is no good arguments for such a function, but there are a lot of performance penalties while using such a function. Basically, to apply the change to a message, opensips/kamilio has to 1) take the received buffer and the changes and to generate a new buffer with the whole message (including the changes) and to 2) take the newly generate buffer and to parse it as a SIP content in order to be able to use internally it. Bottom line, each time you use such a function you double the processing effort for parsing and generating SIP messages. And if you check the code profiling we did (see http://www.opensips.org/Resources/TestsProfiling), these operations are ~50% from the total CPU usage (cumulate the PARSE and BUILD times). Now, in most of the cases (99% of the case) you do not really need to apply changes in realtime - there are a lot of simple tricks to avoid it. If you describe the problem you have, I can help you in putting some extra logic in the script to avoid the need to apply changes. Using a smart approach is more efficient than a brute force approach - the idea is that you are aware of the changes you do in script and you remember (in script) these changes, so you can take them into account in your later processing even if they are not actually applied on the SIPS message. Regards, Bogdan Denis Putyato wrote: Hello! In kamailio project there is a function |msg_apply_changes() ||in textops module for applying changes (for example add or subst some header field) in SIP messages. Is there some way on opensips for doing such operation? Now I need make signaling “loop” for change header fields which I, for example, add during call process.| | | |Opensips 1.6.3| | | |Thank you || | ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ 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-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out
Hi. I see you removed diaplan from default modules. Could you explain why? WBR, Anton Zagorskiy VoIP Developer, Oyster Telecom Phone.: +7 812 601-0666 Fax: +7 812 601-0593 a.zagors...@oyster-telecom.ru www.oyster-telecom.ru -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 21, 2010 1:10 AM To: users@lists.opensips.org Subject: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out Hello all, OpenSIPS 1.6.4 is a major release and it is the third release following the new release policy . 1.6.4 release brings both new features / enhancement and a lot of fixes. The most important additions : - CDR support in ACC module - Media timeout and call termination with nathelper and dialog module - new dialog Presence Call Info - B2BUA API The listing with all additions and fixes is available under http://www.opensips.org/Main/Ver164. Migration documentation (from 1.6.3 to 1.6.4) can be found under http://www.opensips.org/Resources/DocsMigration163to164 OpenSIPS 1.6.4 is now ready for download on project web site and SF download system. The full ChangeLog is available under http://opensips.org/pub/opensips/1.6.4/src/ChangeLog To get the OpenSIPS 1.6.4 version, see the Downloads page - http://www.opensips.org/Resources/Downloads Enjoy, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ 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-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out
On 12/21/2010 12:51 PM, Anton Zagorskiy wrote: Hi. I see you removed diaplan from default modules. Could you explain why? Becase it depends on libpcre-dev. WBR, Anton Zagorskiy VoIP Developer, Oyster Telecom Phone.: +7 812 601-0666 Fax: +7 812 601-0593 a.zagors...@oyster-telecom.ru www.oyster-telecom.ru -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 21, 2010 1:10 AM To: users@lists.opensips.org Subject: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out Hello all, OpenSIPS 1.6.4 is a major release and it is the third release following the new release policy . 1.6.4 release brings both new features / enhancement and a lot of fixes. The most important additions : - CDR support in ACC module - Media timeout and call termination with nathelper and dialog module - new dialog Presence Call Info - B2BUA API The listing with all additions and fixes is available under http://www.opensips.org/Main/Ver164. Migration documentation (from 1.6.3 to 1.6.4) can be found under http://www.opensips.org/Resources/DocsMigration163to164 OpenSIPS 1.6.4 is now ready for download on project web site and SF download system. The full ChangeLog is available under http://opensips.org/pub/opensips/1.6.4/src/ChangeLog To get the OpenSIPS 1.6.4 version, see the Downloads page - http://www.opensips.org/Resources/Downloads Enjoy, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ 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 -- Anca Vamanu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out
Hello In scripts/mysql_update_1_6_4.sh there is such string run_query - Adding new 'attrs' field in DR_GATEWAYS table ALTER TABLE ast_dr_gateways ADD COLUMN attrs CHAR(255) DEFAULT NULL But there is already attrs fields in DR_GATEWAYS. May be in DR_RULES? -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 21, 2010 1:10 AM To: users@lists.opensips.org Subject: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out Hello all, OpenSIPS 1.6.4 is a major release and it is the third release following the new release policy . 1.6.4 release brings both new features / enhancement and a lot of fixes. The most important additions : - CDR support in ACC module - Media timeout and call termination with nathelper and dialog module - new dialog Presence Call Info - B2BUA API The listing with all additions and fixes is available under http://www.opensips.org/Main/Ver164. Migration documentation (from 1.6.3 to 1.6.4) can be found under http://www.opensips.org/Resources/DocsMigration163to164 OpenSIPS 1.6.4 is now ready for download on project web site and SF download system. The full ChangeLog is available under http://opensips.org/pub/opensips/1.6.4/src/ChangeLog To get the OpenSIPS 1.6.4 version, see the Downloads page - http://www.opensips.org/Resources/Downloads Enjoy, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ 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-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out
drouting module: The migration script alters column 'groupid' in the table 'dr_rules' to the type int, but drouting module checks it as DB_STRING (see dr_load.c line 536) WBR, Anton Zagorskiy VoIP Developer, Oyster Telecom Phone.: +7 812 601-0666 Fax: +7 812 601-0593 a.zagors...@oyster-telecom.ru www.oyster-telecom.ru -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 21, 2010 1:10 AM To: users@lists.opensips.org Subject: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out Hello all, OpenSIPS 1.6.4 is a major release and it is the third release following the new release policy . 1.6.4 release brings both new features / enhancement and a lot of fixes. The most important additions : - CDR support in ACC module - Media timeout and call termination with nathelper and dialog module - new dialog Presence Call Info - B2BUA API The listing with all additions and fixes is available under http://www.opensips.org/Main/Ver164. Migration documentation (from 1.6.3 to 1.6.4) can be found under http://www.opensips.org/Resources/DocsMigration163to164 OpenSIPS 1.6.4 is now ready for download on project web site and SF download system. The full ChangeLog is available under http://opensips.org/pub/opensips/1.6.4/src/ChangeLog To get the OpenSIPS 1.6.4 version, see the Downloads page - http://www.opensips.org/Resources/Downloads Enjoy, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ 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-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out
Hello, Yes, you are right, the new attrs field should be added in the DR_RULES table and groupid field should be modified in the DR_GROUPS table. Thanks for the notice and sorry for this mistake. We will fix this and regenerate the tars. On 12/21/2010 02:21 PM, Anton Zagorskiy wrote: drouting module: The migration script alters column 'groupid' in the table 'dr_rules' to the type int, but drouting module checks it as DB_STRING (see dr_load.c line 536) WBR, Anton Zagorskiy VoIP Developer, Oyster Telecom Phone.: +7 812 601-0666 Fax: +7 812 601-0593 a.zagors...@oyster-telecom.ru www.oyster-telecom.ru -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 21, 2010 1:10 AM To: users@lists.opensips.org Subject: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out Hello all, OpenSIPS 1.6.4 is a major release and it is the third release following the new release policy . 1.6.4 release brings both new features / enhancement and a lot of fixes. The most important additions : - CDR support in ACC module - Media timeout and call termination with nathelper and dialog module - new dialog Presence Call Info - B2BUA API The listing with all additions and fixes is available under http://www.opensips.org/Main/Ver164. Migration documentation (from 1.6.3 to 1.6.4) can be found under http://www.opensips.org/Resources/DocsMigration163to164 OpenSIPS 1.6.4 is now ready for download on project web site and SF download system. The full ChangeLog is available under http://opensips.org/pub/opensips/1.6.4/src/ChangeLog To get the OpenSIPS 1.6.4 version, see the Downloads page - http://www.opensips.org/Resources/Downloads Enjoy, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ 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 -- Vlad Paiu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out
Tarball regenerated both on opensips.org and SourceForge ! Enjoy, Bogdan Vlad Paiu wrote: Hello, Yes, you are right, the new attrs field should be added in the DR_RULES table and groupid field should be modified in the DR_GROUPS table. Thanks for the notice and sorry for this mistake. We will fix this and regenerate the tars. On 12/21/2010 02:21 PM, Anton Zagorskiy wrote: drouting module: The migration script alters column 'groupid' in the table 'dr_rules' to the type int, but drouting module checks it as DB_STRING (see dr_load.c line 536) WBR, Anton Zagorskiy VoIP Developer, Oyster Telecom Phone.: +7 812 601-0666 Fax: +7 812 601-0593 a.zagors...@oyster-telecom.ru www.oyster-telecom.ru -Original Message- From: users-boun...@lists.opensips.org [mailto:users- boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 21, 2010 1:10 AM To: users@lists.opensips.org Subject: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out Hello all, OpenSIPS 1.6.4 is a major release and it is the third release following the new release policy . 1.6.4 release brings both new features / enhancement and a lot of fixes. The most important additions : - CDR support in ACC module - Media timeout and call termination with nathelper and dialog module - new dialog Presence Call Info - B2BUA API The listing with all additions and fixes is available under http://www.opensips.org/Main/Ver164. Migration documentation (from 1.6.3 to 1.6.4) can be found under http://www.opensips.org/Resources/DocsMigration163to164 OpenSIPS 1.6.4 is now ready for download on project web site and SF download system. The full ChangeLog is available under http://opensips.org/pub/opensips/1.6.4/src/ChangeLog To get the OpenSIPS 1.6.4 version, see the Downloads page - http://www.opensips.org/Resources/Downloads Enjoy, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ 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 -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] segfault in codecs.c during call back
Hi Bobby,
Could you confirm that the latest fixes on trunk (for codec ops) solved
your problem ?
Regards,
Bogdan
Bobby Smith wrote:
The crash has happened again twice after this posting, in exactly the
same place. Here's the bt of the latest core.
I can make two observations:
a) In the seven times we've experienced it so far, it's happened on
either a NOTIFY (with message summary) or an OPTIONS (with no body)
that hits the failure_route[softswitch] in the routing script here:
http://pastebin.com/RmVVW8N7
b) The text ops call we're using that seems to cause this is the
codec_delete_except_re() function -- which we invoke twice, once
initially, and once again in the failure route as it recurses back
through the main route setup.
1.
failure_route[softswitch] {
2.
xlog(failure route[$ci] -- softswitch -- reached -- load
balancing $rm to a different destination, current destination is
$dd for $ci);
3.
if(t_was_cancelled()) {
4.
exit;
5.
}
6.
7.
if (t_check_status([4-5][0-9][0-9])) {
8.
lb_disable();
9.
xlog(failure route[$ci] -- load balancer picked new destination
for $rm, destination is $dd);
10.
route(main);
11.
exit;
12.
}
13.
}
1.
2.
route[filter_codecs] {
3.
if(has_body(application/sdp))
4.
{
5.
codec_delete_except_re(PCMA|PCMU|telephone-event);
6.
}
7.
}
#0 backup () at codecs.c:104
104 int n = old-len;
(gdb) bt full
#0 backup () at codecs.c:104
l = 0x7a1df0
old = 0x0
n = value optimized out
len = 1
i = 0
__FUNCTION__ = backup
#1 0x2aeb40aea0e1 in pre_route_callback (msg=0x2aeb406c5140,
param=0x7a2c80) at codecs.c:169
No locals.
#2 0x0046d9be in exec_post_cb (msg=0x2aeb406c5140) at
script_cb.c:198
No locals.
#3 exec_pre_route_cb (msg=0x2aeb406c5140) at script_cb.c:231
No locals.
#4 0x00412153 in run_top_route (a=0x798070,
msg=0x2aeb406c5140) at action.c:181
bk_action_flags = 0
bk_rec_lev = 0
#5 0x2aeb404b178f in run_failure_handlers (Trans=0x2aeb4250b070,
new_code=value optimized out, branch=value optimized out,
should_store=0x7fff50e2f7b8, should_relay=0x7fff50e2f7bc,
cancel_bitmap=value optimized out,
reply=0x79f740) at t_reply.c:613
faked_req = {id = 1024, first_line = {type = 1, len = 53, u = {request
= {method = {
s = 0x2aeb4252f5a8 OPTIONS sip:1...@internal-sip.mysipserver.com
mailto:sip%3a...@internal-sip.mysipserver.com
SIP/2.0\r\nRecord-Route:
sip:4.2.2.245;lr=on;ftag=6364386232656635313363340131333138353238393233\r\nVia:
SIP/2.0/UDP 4.2.2.245;branch=z9hG4bK4721.cf..., len = 7}, uri = {
s = 0x2aeb4252f5b0 sip:1...@internal-sip.mysipserver.com
mailto:sip%3a...@internal-sip.mysipserver.com
SIP/2.0\r\nRecord-Route:
sip:4.2.2.245;lr=on;ftag=6364386232656635313363340131333138353238393233\r\nVia:
SIP/2.0/UDP 4.2.2.245;branch=z9hG4bK4721.cf01ae25.0..., len = 35},
version = {
s = 0x2aeb4252f5d4 SIP/2.0\r\nRecord-Route:
sip:4.2.2.245;lr=on;ftag=6364386232656635313363340131333138353238393233\r\nVia:
SIP/2.0/UDP 4.2.2.245;branch=z9hG4bK4721.cf01ae25.0\r\nVia:
SIP/2.0/UDP 217.73.175.1:5060 http://217.73.175.1:5060..., len =
7}, method_value = 32}, reply = {version = {
s = 0x2aeb4252f5a8 OPTIONS sip:1...@internal-sip.mysipserver.com
mailto:sip%3a...@internal-sip.mysipserver.com
SIP/2.0\r\nRecord-Route:
sip:4.2.2.245;lr=on;ftag=6364386232656635313363340131333138353238393233\r\nVia:
SIP/2.0/UDP 4.2.2.245;branch=z9hG4bK4721.cf..., len = 7}, status = {
s = 0x2aeb4252f5b0 sip:1...@internal-sip.mysipserver.com
mailto:sip%3a...@internal-sip.mysipserver.com
SIP/2.0\r\nRecord-Route:
sip:4.2.2.245;lr=on;ftag=6364386232656635313363340131333138353238393233\r\nVia:
SIP/2.0/UDP 4.2.2.245;branch=z9hG4bK4721.cf01ae25.0..., len = 35},
reason = {
s = 0x2aeb4252f5d4 SIP/2.0\r\nRecord-Route:
sip:4.2.2.245;lr=on;ftag=6364386232656635313363340131333138353238393233\r\nVia:
SIP/2.0/UDP 4.2.2.245;branch=z9hG4bK4721.cf01ae25.0\r\nVia:
SIP/2.0/UDP 217.73.175.1:5060 http://217.73.175.1:5060..., len =
7}, statuscode = 32}}}, via1 = 0x2aeb4252f8a0, via2 = 0x2aeb4252fa18,
headers = 0x2aeb4252f810, last_header = 0x2aeb4252ffd8, parsed_flag =
18446744073709551615, h_via1 = 0x2aeb4252f858, h_via2 = 0x2aeb4252f9d0,
callid = 0x2aeb4252ff90, to = 0x2aeb4252fce8, cseq = 0x2aeb4252ff18,
from = 0x2aeb4252fc10, contact = 0x2aeb4252fed0, maxforwards =
0x2aeb4252ffd8, route = 0x0, record_route = 0x2aeb4252f810, path =
0x0, content_type = 0x0,
content_length = 0x2aeb4252fbc8, authorization = 0x0, expires = 0x0,
proxy_auth = 0x0, supported = 0x0, proxy_require = 0x0, unsupported =
0x0, allow = 0x0, event = 0x0, accept = 0x2aeb4252fc58,
accept_language = 0x0,
organization = 0x0, priority = 0x0, subject = 0x0, user_agent =
0x2aeb4252fca0, content_disposition = 0x0, accept_disposition = 0x0,
diversion = 0x0, rpid = 0x0,
Re: [OpenSIPS-Users] just one quick question
Hi Vic, No example, but I can try to help you. have you follow the tutorial for freeradius with opensips ? http://www.opensips.org/Resources/DocsTutorials#toc11 http://www.opensips.org/Resources/DocsTutorials#toc12 Regards, Bogdan Vic Jolin wrote: can you give an example on how to do it? really sorry but Im new in this thanks On Tue, Dec 7, 2010 at 6:36 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Vic, if you read the RADIUS Accounting RFC, you see that acc has nothing to do with credit or time duration checks. Actually, with RADIUS, you need to use an Auth/Access Request to get information about how long the call should be allowed. Typically, with RADIUS, there is no cost involved. The proxy does an Access Request to the RADIUS server (when the call is received) to see for how long the call should be allowed. Then the proxy will allow the call only for that maximum period and it will generate the START and STOP accounting events to the RADIUS server. Regards, Bogdan Vic Jolin wrote: If I were to use freeradius for accounting, where should I put user list? and also I'd like to check credit as well before calling. Thanks ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro http://www.voice-system.ro ___ 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 Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] terminating early dialogs with BYE
Hi Taisto, I agree on what you said, still from a dialog point of view , in early state, CANCEL and BYE are parallel transaction that do not directly terminate the INVITE on proxy, but they are handled by end clients which is responsible the rejecting the INVITE due when receiving CANCEL/BYE in early state. Regards, Bogdan Taisto Qvist (WM) wrote: I stumbled onto a problem with BYE in early dialog last week, during forking, but its not an core opensips issue. Since my OpenSIPs setup is dialog-agnostic, any client that sends BYE in an early dialog, will only manage to trigger the UAS to send a 487 or simlar to the specific INVITE transaction. A transaction stateful, forking proxy, wether working in parallell or serial, has no way of matching the BYE to the other transactions, so any other parallell branches will keep going just as normal, possibly returning a 2xx response. In a serial forking case, the proxy will just rollOver to the next branch, which in turn could complete the call with a 2xx. So the BYE from the client results in very different behavior than a CANCEL which i *assume* is the expected behavior. UACs should use CANCEL if they want to terminate the *session* and BYE if they want to terminate a *specific* early dialog. I've never worked with the dialog-module but I assume that if you wanted to handle this case as if it was cancel, you would in some way have to find the unfinished invite transaction, and generate cancel on it. A ::cancelMatchingInvite(SIP_ByeRequest *) sortof... Regards Taisto On Wed, 08 Dec 2010 12:04:57 +0200, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Andrew, Do you refer to the dialog module not terminating an early dialog if a bye is received ? I know the RFC allows it, but never be able to actually test with something like that. On the other hand, I do not think we need to do anything special to support BYE in early state. In such a case, even if the dialog state machine will not be triggered by the BYE request, the BYE will get to the callee and callee will terminate the INVITE transaction with a negative reply (like 487 when cancelling)...So, the failed INVITE transaction will trigger the dialog termination.. Or maybe I fail to understand how this BYE for early dialogs worksCan you reproduce this scenario ? Regards, Bogdan Andrew Pogrebennyk wrote: Hi, RFC3261 paragraph 15 Terminating a Session says: When a BYE is received on a dialog, any session associated with that dialog SHOULD terminate. A UA MUST NOT send a BYE outside of a dialog. The caller's UA MAY send a BYE for either confirmed or early dialogs, and the callee's UA MAY send a BYE on confirmed dialogs, but MUST NOT send a BYE on early dialogs. However early dialog termination with BYE appears to be not supported in OpenSIPS. If there any known solution to that? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 477 send failed with TCP
Hi Anshuman,
for such call, could you check if before sending the BYE (or any other
in-dialog request) the TCP connection is still up (check with netstat) -
it may be the case where the TCP conn (opened by INVITE) is terminated
and opensips cannot open a connection behind NAT back to the client.
Regards,
Bogdan
Anshuman S. Rawat wrote:
Thanks Bogdan. That fixed most of the problems but not all.
I have a scenario where I make calls without registering (transport is
TCP) so the contact in INVITE carries local/private IP address.
Result: I cannot receive in-dailog requests after call in setup. It
tried to fix this by doing this -
if (method == INVITE) {
# Required for TCP - Anshuman
xlog(L_INFO, INVITE: Check for NAT);
if (nat_uac_test(1)) {
xlog(L_INFO, NAT: FIXING CONTACT in
INVITE);
fix_nated_contact();
};
};
This is followed by processing by the lcr module -
if ((method == INVITE) !(to_uri=~paral...@.+)){
# do lcr
# xlog(Message :$tu);
if (!load_contacts()){
sl_send_reply(500,Unable to load
contacts);
exit;
}else{
if (next_contacts()){
t_on_failure(1);
}#else{
# sl_send_reply(404,Not found );
# exit;
# }
}
append_hf(P-hint: lcr applied\r\n);
}else{
append_hf(P-hint: usrloc applied\r\n);
}
The above does not fix the 'Contact' header in the INVITE (i.e.
Contact still carries private IP address).
Can anyone help me to identify what's wrong with this picture?
Regards,
Anshuman
- Original Message - From: Bogdan-Andrei Iancu
bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Sent: Thursday, December 09, 2010 5:30 PM
Subject: Re: [OpenSIPS-Users] 477 send failed with TCP
Hi Anshuman,
do fix_nated_register() at registration time (before save(location))
if uac_nat_test() function returns true (if nat was detected) - see the
nathelper module for more details:
http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html
Regards,
Bogdan
Anshuman S. Rawat wrote:
Hi,
I am trying to use TCP on OpenSIPS 1.6 (with PJSIP as client). Client
successfully registers and sends keep-alives to keep the TCP
connection open. However, all call attempts (or any other request)
fail with a '477 Send Failed'. The logs show this -
Dec 8 06:58:18 ip-208-109-177-46 /usr/local/sbin/opensips[15463]:
ERROR:core:tcp_blocking_connect: timeout 10 s elapsed from 10 s
Dec 8 06:58:18 ip-208-109-177-46 /usr/local/sbin/opensips[15463]:
ERROR:core:tcpconn_connect: tcp_blocking_connect failed
Dec 8 06:58:18 ip-208-109-177-46 /usr/local/sbin/opensips[15463]:
ERROR:core:tcp_send: connect failed
Dec 8 06:58:18 ip-208-109-177-46 /usr/local/sbin/opensips[15463]:
ERROR:tm:msg_send: tcp_send failed
Client is behind a NAT so registers with contact containing the
private IP address. It seems that OpenSIPS is trying to connect to the
private IP address contained in the REGISTER request rather than
reusing the connection the request arrived on.
Is this a bug? Or is there a way to configure it to use the
established connection?
Thanks,
Anshuman
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.872 / Virus Database: 271.1.1/3304 - Release Date:
12/08/10 13:04:00
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Bogdan-Andrei Iancu
OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami, USA
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] random segfaults without generating core file
Hi Bobby, what is the status on that? have you managed to get the core files? Regards, Bogdan Bobby Smith wrote: Greetings, We're encountering an issue on 1.6 trunk where we're receiving random segmentation faults that DO NOT generate cores. I've had a couple of segfaults (which I've posted to the list) that have resulted in core dumps, but doing a dmesg: opensips[28467]: segfault at 0010 rip 2b0ea5b92e79 rsp 7fffa7dd7080 error 4 opensips[8587]: segfault at 0010 rip 2ae8797c3e79 rsp 7fffb83cc080 error 4 opensips[13853]: segfault at 0010 rip 2b147aadee79 rsp 7fff37fc2c90 error 4 opensips[23692]: segfault at 0010 rip 2b7c7d9bce79 rsp 7fff71b0cea0 error 4 opensips[26753]: segfault at 0010 rip 2abd2f30be79 rsp 7ed97870 error 4 opensips[27697]: segfault at 0010 rip 2aae52261e79 rsp 7fffa90ebd30 error 4 opensips[13233]: segfault at 0010 rip 2ad495261e79 rsp 7fff224559b0 error 4 opensips[2168]: segfault at 0010 rip 2ab01f40be79 rsp 7fff622a54f0 error 4 opensips[9084]: segfault at 0010 rip 2b4bba9f7e79 rsp 7fffa884f4d0 error 4 opensips[21966]: segfault at 0010 rip 2b823ff9ae79 rsp 7fff6e81b370 error 4 This seems to happening fairly regularly (once a day). The latest occurrence was on a machine that was under 0% load (once a minute periodic OPTIONS health checks from sipsak, and that's it). http://pastebin.com/RmVVW8N7 Any ideas of how to track this issue down? Thanks, Bobby Smith ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] destroy dialog on transaction timeout and load balancer
Hi Bobby, Form dialog state machine point of view, an dialog without 200OK ACK is ok, it will not timeout it. So, I see 2 future solutions: 1) we change the code so that (optional) timeout and dialog termination will be done for missing ACK also 2) make a small shell script that takes the list of dialogs (opensipsctl fifo dlg_list), greps for the dialogs in CONFIRMED BUT NOT ACKED state and if lifetime longer than X sec, it terminates the dialog via opensipsctl fifo dlg_end_dlg Regards, Bogdan Bobby Smith wrote: Is there an easy way/example from a scripting perspective to check if a dialog's status is 3 (not received an ACK yet), after a certain period of time, time out ONLY dialogs in this state? I've identified a situation where, if the UAC goes unresponsive, and never sends an ACK, we will eventually stop retransmitting the 200 OK to this but the dialog will remain established in memory. Because of this, the load balancer resources list becomes highly inaccurate over time. I'm hesitant to set the dialog default timeout flag, because there are certain situations where a long call (6 hours +) could occur, and we need resource accuracy to be 1 hour ish. Thanks, Bobby Smith ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Need some help with NAT/rtpproxy
Hi James,
when using fix_nated_register(), opensips saves as contact (in user
location) the received contact (which is private) and saves as outbound
proxy the public IP of the NAT (source IP and port).
So, after a lookup(), when sending a request to the nated client, the
RURI will have the saved contact (the private one) and the $du
(destination URI) will point to the NAT public address...So, your test
is bogus, as the RURI will be indeed private.
Better test like : if $du is present, test $du if private; if no $du,
test $ru.
Regards,
Bogdan
James Lamanna wrote:
Hi,
I'm having some issues getting a correct NAT configuration going.
The problem I'm having is
I get a 479 We don't forward to private IP addresses back when
receiving a call to a phone from Asterisk, presumably because the
location table has private IPs in it for some reason.
This seems to be related to my failed attempt to use fix_nated_register().
Removing the call to fix_nated_register and just using
fix_nated_contact allows calls to go through, but then I get no audio
on either side...
Config follows.
Thanks.
-- James
debug=3 # debug level (cmd line: -dd)
fork=yes
log_stderror=no # (cmd line: -E)
log_facility=LOG_LOCAL0
tos=0x60
# Uncomment these lines to enter debugging mode
#fork=no
#log_stderror=yes
#debug=6
check_via=no# (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
listen=udp:208.xxx.xxx.6:5060
listen=udp:208.xxx.xxx.6:5061
# -- module loading --
#set module path
#mpath=/usr/local/lib/opensips/modules/
mpath=/usr/local/lib64/opensips/modules/
# Uncomment this if you want to use SQL database
loadmodule db_mysql.so
loadmodule sl.so
loadmodule maxfwd.so
loadmodule textops.so
loadmodule avpops.so
loadmodule tm.so
loadmodule rr.so
loadmodule dialog.so
loadmodule signaling.so
loadmodule options.so
loadmodule localcache.so
loadmodule usrloc.so
loadmodule presence.so
loadmodule presence_xml.so
loadmodule presence_dialoginfo.so
loadmodule pua.so
loadmodule pua_dialoginfo.so
#loadmodule pua_bla.so
loadmodule pua_usrloc.so
loadmodule registrar.so
loadmodule mi_fifo.so
#loadmodule xlog.so
# Uncomment this if you want digest authentication
# db_mysql.so must be loaded !
loadmodule auth.so
loadmodule auth_db.so
# !! Nathelper
loadmodule nathelper.so
# - setting module-specific parameters ---
# -- mi_fifo params --
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
modparam(usrloc, db_mode, 2)
modparam(usrloc|dialog|dispatcher|presence|presence_xml|pua|avpops,
db_url, mysql://opensips:xx...@localhost/opensips)
modparam(avpops,avp_table,usr_preferences)
#modparam(dispatcher, force_dst, 1)
# Only use username
#modparam(dispatcher, flags, 1)
# Store passwords for 1 hour in cache
modparam(auth,username_spec,$avp(i:54))
modparam(auth,password_spec,$avp(i:55))
modparam(auth,calculate_ha1,1)
modparam(auth_db, db_url,
mysql://opensipsro:x...@localhost/opensips)
modparam(auth_db, calculate_ha1, yes)
modparam(auth_db, password_column, password)
modparam(auth_db, load_credentials, $avp(i:55)=password)
modparam(rr, enable_full_lr, 1)
modparam(dialog, dlg_flag, 4)
modparam(dialog, profiles_with_value, caller)
modparam(usrloc,nat_bflag,6)
modparam(nathelper,sipping_bflag,8)
#modparam(nathelper, natping_interval, 30)
modparam(nathelper, ping_nated_only, 1) # Ping only clients behind NAT
#modparam(nathelper, natping_interval, 30)
modparam(nathelper, sipping_from, sip:pin...@208.xxx.xxx.6)
modparam(nathelper, rtpproxy_sock, unix:/var/run/rtpproxy/rtpproxy.sock)
modparam(nathelper, received_avp, $avp(i:42))
modparam(registrar, received_avp, $avp(i:42))
modparam(presence, server_address, sip:s...@208.xxx.xxx.6:5060)
modparam(presence, expires_offset, 10)
modparam(presence, mix_dialog_presence, 1)
#modparam(presence, fallback2db, 1)
modparam(presence_xml, force_active, 1)
modparam(presence_dialoginfo, force_single_dialog, 1)
modparam(pua_dialoginfo, presence_server, sip:s...@208.xxx.xxx.6:5060)
modparam(pua_dialoginfo, include_callid, 1)
modparam(pua_dialoginfo, include_tags, 1)
modparam(pua_dialoginfo, caller_confirmed, 1)
modparam(pua_usrloc, default_domain, 208.xxx.xxx.6)
modparam(pua_usrloc, presence_server, sip:s...@208.xxx.xxx.6:5060)
#modparam(stun,primary_ip,208.xxx.xxx.6)
#modparam(stun,alternate_ip,208.90.184.10)
#modparam(stun,primary_port,5060)
#modparam(stun,alternate_port,3479)
# - request routing logic ---
# main routing logic
route{
if (!is_method(NOTIFY))
xlog(L_INFO, New request - Request/failure/branch routes:
M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n);
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
};
if
Re: [OpenSIPS-Users] Opensips 1.6.3 stable version
Hi Antonio, Usually I prefer to use SVN checkout (versus a tarball) as it is simpler to update in the future. BTW, check the 1.6.4 stable release - http://www.opensips.org/Main/Ver164 Regards, Bogdan Antonio Anderson Souza wrote: Dear List, I'm using Opensips 1.6.3 version from the branch 1.6 [1] in my development environment, due a patch that it's not available in the stable download [2], but I'll need to make some installations in production environments, and I'd like to hear form you guys which is the better choice, use the stable version[1] and apply only the patch that I need, or use the version in the branch [2]? [1] - http://opensips.org/pub/opensips/1.6.3/src/opensips-1.6.3-notls_src.tar.gz [2] - http://opensips.svn.sourceforge.net/viewvc/opensips/branches/1.6 Best Regards, Antonio Anderson Souza Blog http://www.antonioams.com - Twitter http://twitter.com/antonioams - LinkedIn http://br.linkedin.com/in/antonioams - Facebook http://www.facebook.com/antonioams ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] back to back request and reply routes
Hi Pete, you cannot combine dialog and b2bua at the same passing through opensipsit does not work like that. Regards, Bogdan Pete Kelly wrote: Hi In the b2b_entities module it is possible to define a reply and a request route. I know it is not possible to modify SIP within these routes, however is anything else possible apart from logging and inspection of the SIP? Specifically I am thinking about using the dialog module within these routes. Pete ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Not enough free memory
Hi Ronald, You haven't properly enabled the mem debugger in your Makefile.defs please post here the section where you did the changes just to double check it. Regards, Bogdan Ronald Cepres wrote: Hi Ovidiu, I tried enabling memory manager debug but doing so causes it to somewhat crash (no error logs but opensips stops running) with just a single call. Here is the link to download the logs i fetched with memory manager debug enabled if this would help: http://www.2shared.com/file/npidL9pz/opensips.html Thanks! Regards, Ronald On Fri, Dec 17, 2010 at 1:56 AM, Ovidiu Sas o...@voipembedded.com mailto:o...@voipembedded.com wrote: Here's how to investigate/debug memory issues: http://www.opensips.org/Resources/DocsTsMem Regards, Ovidiu Sas On Thu, Dec 16, 2010 at 12:44 PM, Ronald Cepres rbcep...@gmail.com mailto:rbcep...@gmail.com wrote: Hi to all, If I run OpenSIPS (1.6.3 ) for a long time while calls are coming in, it suddenly stops with the following sample logs: Dec 16 12:26:43 [12114] ERROR:core:new_avp: no more shm mem Dec 16 12:26:43 [12114] ERROR:core:add_avp: Failed to create new avp structure Dec 16 12:26:43 [12114] ERROR:avpops:db_query_avp: unable to add avp Dec 16 12:26:43 [12103] ERROR:core:new_avp: no more shm mem Dec 16 12:26:43 [12103] ERROR:core:add_avp: Failed to create new avp structure Dec 16 12:26:43 [12103] ERROR:avpops:db_query_avp: unable to add avp Dec 16 12:26:43 [12106] WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Dec 16 12:26:43 [12106] WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Dec 16 12:26:43 [12106] ERROR:dialog:dlg_add_leg_info: no more shm mem Dec 16 12:26:43 [12106] ERROR:dialog:init_leg_info: dlg_add_leg_info failed Dec 16 12:26:43 [12106] ERROR:dialog:push_reply_in_dialog: could not add further info to the dialog Dec 16 12:26:43 [12105] WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Dec 16 12:26:43 [12105] ERROR:core:new_avp: no more shm mem Dec 16 12:26:43 [12105] ERROR:core:add_avp: Failed to create new avp structure Dec 16 12:26:43 [12102] WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Dec 16 12:26:43 [12105] ERROR:avpops:db_query_avp: unable to add avp Dec 16 12:26:43 [12102] ERROR:dialog:build_new_dlg: no more shm mem (277) Dec 16 12:26:43 [12105] WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Dec 16 12:26:43 [12105] ERROR:tm:new_t: out of mem Dec 16 12:26:43 [12105] ERROR:tm:t_newtran: new_t failed Dec 16 12:26:43 [12117] WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation Dec 16 12:26:43 [12102] ERROR:dialog:dlg_create_dialog: failed to create new dialog Dec 16 12:26:43 [12102] ERROR:dialog:set_dlg_profile: dialog was not yet created - script error Dec 16 12:26:43 [12102] ERROR:dialog:w_set_dlg_profile: failed to set profile Is there a way that memory leak would happen from a mis-configured routing script? If so, how do I trace the part of the routing script that causes the leak? Thanks! ___ 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 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 Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] receive port during call process
Hi Denis, The ACK is routed based on the RR headers that were collected by INVITE and returned in 200 OK. For such a scenario to work, opensips is doing double routing (adds for itself 2 RR headers, one with the inbound interface, one with the outbound interface). Could you post the SIP capture of such a call to check if correct from SIP point of view? Regards, Bogdan Denis Putyato wrote: Hello! I have a such problem. Opensips using 2 ports One – 5068 for client which must register on Opensips Second – 5060 for all other clients. 1) Client А registering on Opensips (socket: udp:2.2.2.2:5068). Client A is behind NAT. 2) Client А receives incoming call (via lookup() function). 3) Call has such way Cisco (source port 1 sends INVITE to port 5060 of Opensips) – Opensips (receives INVITE from Cisco to port 5060 and sends the INVITE to client from source port 5068 to some client`s port) – Client A. Everything work fine until client А answers. Then Client A sends to Opensips (port 5068) 200 OK, Opensips retransmit it to Cisco (from port 5060 to 1), Cisco sends to Opensips ACK (from port 1 to 5060) AND Opensips retransmit this ACK to client`s port FROM PORT 5060, BUT NOT 5068. This ACK didn`t reach client A because of NAT. Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Shared DB tables among several opensips instances
Hi James, if DB sharing is possible, depends on what module are you using. Some modules allow their tables to be shared by other OpenSIPS instances, other not. Like usrloc (registration storage of opensips) can share its table only if in db_mode 3 (DB_ONLY). Regards, Bogdan James Lamanna wrote: Hi, Is it possible to share the same DB tables among several running OpenSIPs instances? What I'm trying to do is use OpenSIPs as a registration front-end to Asterisk. The idea is to have a cluster of registration servers, and then a cluster of Asterisk servers. Can an Asterisk server pass a call to any of the OpenSIPs servers in the cluster, and have opensips deliver the call even if the phone isn't directly registered to that particular server? Or does the call always have to be passed to the opensips instance that the phone is directly registered to, even if there is a shared DB? Thanks. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASE] OpenSIPS 1.6.4 major release is out
Debian packages are available now for: - 4.0 etch - 5.0 lenny - sid/ unstable You can download directly the deb files from : http://opensips.org/pub/opensips/latest/packages/debian/ or you can use the opensips.org debian repo: deb http://www.opensips.org/apt/ lenny main deb http://www.opensips.org/apt/ sid main deb http://www.opensips.org/apt/ etch main Best regards, Bogdan Bogdan-Andrei Iancu wrote: Hello all, OpenSIPS 1.6.4 is a major release and it is the third release following the new release policy . 1.6.4 release brings both new features / enhancement and a lot of fixes. The most important additions : - CDR support in ACC module - Media timeout and call termination with nathelper and dialog module - new dialog Presence Call Info - B2BUA API The listing with all additions and fixes is available under http://www.opensips.org/Main/Ver164. Migration documentation (from 1.6.3 to 1.6.4) can be found under http://www.opensips.org/Resources/DocsMigration163to164 OpenSIPS 1.6.4 is now ready for download on project web site and SF download system. The full ChangeLog is available under http://opensips.org/pub/opensips/1.6.4/src/ChangeLog To get the OpenSIPS 1.6.4 version, see the Downloads page - http://www.opensips.org/Resources/Downloads Enjoy, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips just stops responding
Hello,
It is quite old post, but i have just encoutered quite similiar problem.
I have the latest revision installed $Revision: 4448 in my server.
Opensips is starting itself properly:
# ps -ef | grep opensips
root 20982 6115 0 02:01 pts/100:00:00 gdb /usr/local/sbin/opensips
root 21326 1 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21328 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21329 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21330 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21331 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21332 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21333 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21334 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21335 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21336 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21337 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21338 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21339 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21340 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21341 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21342 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21343 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21392 6258 0 02:18 pts/200:00:00 grep opensips
and there is a opensips.pid file generated.
The problem is that opensips is not responding to any request, there
are no debug information at all (default opensips.conf file)
I even created simple script route { log... } and the effect is the same.
Here is how opensips is starting (debug 5):
Dec 22 02:25:50 mac opensips: WARNING:core:fix_socket_list: could not
rev. resolve 62.29.162.76
Dec 22 02:25:50 mac opensips: WARNING:core:fix_socket_list: could not
rev. resolve 62.29.162.76
Dec 22 02:25:50 mac opensips: INFO:core:init_tcp: using epoll_lt as
the TCP io watch method (auto detected)
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: NOTICE:core:main:
version: opensips 1.6.4-notls (i386/linux)
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:core:main:
using 32 Mb shared memory
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:core:main:
using 1 Mb private memory per process
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
NOTICE:signaling:mod_init: initializing module ...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:sl:mod_init:
Initializing StateLess engine
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:tm:mod_init:
TM - initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:rr:mod_init:
rr - initializing
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:maxfwd:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:usrloc:ul_init_locks: locks array size 512
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:registrar:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:textops:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:acc:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255
kb
more info from gdb /usr/local/sbin/opensips
(gdb)
No stack.
and the output from opensipsctl fifo ps
Process:: ID=0 PID=21326 Type=attendant
Process:: ID=1 PID=21328 Type=MI FIFO
Process:: ID=2 PID=21329 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=3 PID=21330 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=4 PID=21331 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=5 PID=21332 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=6 PID=21333 Type=SIP receiver udp:62.29.162.76:5060
Process:: ID=7 PID=21334 Type=SIP receiver udp:62.29.162.76:5060
Process:: ID=8 PID=21335 Type=SIP receiver udp:62.29.162.76:5060
Process:: ID=9 PID=21336 Type=SIP receiver udp:62.29.162.76:5060
Process:: ID=10 PID=21337 Type=time_keeper
Process:: ID=11 PID=21338 Type=timer
Process:: ID=12 PID=21339 Type=TCP receiver
Process:: ID=13 PID=21340 Type=TCP receiver
Process:: ID=14 PID=21341 Type=TCP receiver
Process:: ID=15
[OpenSIPS-Users] Presentity record not deleted in database
I am hoping someone can help me with a Presence issue. I have two Bria Clients with XCAP and presence set up. Each client is able to see the others presence. When I Disable an account on one of the Bria clients the other client see's that the user is no longer available (Which is how it should be). The issue I am seeing is that instead of logging the user out I am exiting/shutting down the Bria client. When this happens the Bria client shuts down, but the other bria client still thinks that the user is availabe. I can tell that the problem is because the Presentity record in the MySQL Presentity table is not removed. I have attached two different NGREP siptraces. One siptrace is a good trace when I log out a user from the Bria client. The other siptrace is the bad trace when I shut down the bria client and the user still appears to be available. I am pretty sure that Bria is not working correctly when it is being shut down and that is why OpenSIPS is not deleting the Presentity record. One of the differences I see when looking at the Bad Logout.txt and the Good Logout.txt is that the Expires: time on the bad trace shows 3600 seconds, but the good trace shows 0 seconds for the Expires:. Any ideas? Bad logout . # U 2010/12/21 19:49:08.309107 75.XXX.XXX.158:1696 - 173.XXX.XXX.88:5060 PUBLISH sip:9xx22x1...@irock.com SIP/2.0. Via: SIP/2.0/UDP 192.168.33.28:1696;branch=z9hG4bK-d8754z-255fea98f6b405cd-1---d8754z-;rport. Max-Forwards: 70. Contact: sip:9xx22x1...@75.xxx.xxx.158:1696;transport=udp. To: Moo sip:9xx22x1...@irock.com. From: Moo sip:9xx22x1...@irock.com;tag=d98ab372. Call-ID: MTY5ZjM1ZmU2YjI1YWYzM2E3ZDQyY2YwNTc4MDY0Y2I.. CSeq: 3 PUBLISH. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/pidf+xml. SIP-If-Match: a.1292905955.11145.88.1. User-Agent: Bria 3 release 3.1.2 stamp 58754. Event: presence. Content-Length: 421. . ?xml version='1.0' encoding='UTF-8'?presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:lt='urn:ietf:params:xml:ns:location-type' entity='sip:9xx22x1...@irock.com'tuple id='t84ea9bf3'statusbasicopen/basic/status/tupledm:person id='p960d0c33'/dm:person/presence # U 2010/12/21 19:49:08.337579 75.XXX.XXX.158:1696 - 173.XXX.XXX.88:5060 SUBSCRIBE sip:s...@173.xxx.xxx.88:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.33.28:1696;branch=z9hG4bK-d8754z-060489f24ae80cca-1---d8754z-;rport. Max-Forwards: 70. Contact: sip:9xx22x1...@75.xxx.xxx.158:1696;transport=udp. To: sip:9xx27x2...@irock.com;tag=155c340f586c28d0300cf5a6ccf90d99-5ddb. From: Moo sip:9xx22x1...@irock.com;tag=3e7dfbd6. Call-ID: OTZiY2IxZDU1YWNlNjRkYTAxZDViMTIwN2U0ZTNlZjE.. CSeq: 2 SUBSCRIBE. Subject: . Expires: 0. Accept: multipart/related, application/rlmi+xml, application/pidf+xml. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: Bria 3 release 3.1.2 stamp 58754. Event: presence. Content-Length: 0. . # U 2010/12/21 19:49:08.342830 75.XXX.XXX.158:1696 - 173.XXX.XXX.88:5060 REGISTER sip:irock.com SIP/2.0. Via: SIP/2.0/UDP 192.168.33.28:1696;branch=z9hG4bK-d8754z-af182e22565fde78-1---d8754z-;rport. Max-Forwards: 70. Contact: sip:9xx22x1...@75.xxx.xxx.158:1696;transport=udp;rinstance=535052ae244d4bb9;expires=0. To: Moo sip:9xx22x1...@irock.com. From: Moo sip:9xx22x1...@irock.com;tag=fe637cac. Call-ID: NjE5MWRkMDAzNzljZDU2NmJkOGNkZTAwZjZlMWYyNDU.. CSeq: 5 REGISTER. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: Bria 3 release 3.1.2 stamp 58754. Authorization: Digest username=9XX22X1XX2,realm=irock.com,nonce=4d11589de1ab46c898b7237bbfa541a9ff840893,uri=sip:irock.com,response=8fee318953c5b53263b19668fb5f4519,cnonce=64d2e5890eb463f40d4b666bc31d8873,nc=0004,qop=auth,algorithm=MD5. Content-Length: 0. . # U 2010/12/21 19:49:08.369303 173.XXX.XXX.88:5060 - 75.XXX.XXX.158:1696 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.33.28:1696;branch=z9hG4bK-d8754z-af182e22565fde78-1---d8754z-;rport=1696;received=75.XXX.XXX.158. To: Moo sip:9xx22x1...@irock.com;tag=c97b4d1cb1f3d0da549e06a8d482ef63.69f4. From: Moo sip:9xx22x1...@irock.com;tag=fe637cac. Call-ID: NjE5MWRkMDAzNzljZDU2NmJkOGNkZTAwZjZlMWYyNDU.. CSeq: 5 REGISTER. WWW-Authenticate: Digest realm=irock.com, nonce=4d115932fda8f48635d3e9b4804fa51764240559, qop=auth, stale=true. Server: OpenSIPS (1.6.4-notls (x86_64/linux)). Content-Length: 0. . # U 2010/12/21 19:49:08.401596 173.XXX.XXX.88:5060 - 75.XXX.XXX.158:1696 SIP/2.0 202 OK. Via: SIP/2.0/UDP 192.168.33.28:1696;branch=z9hG4bK-d8754z-060489f24ae80cca-1---d8754z-;rport=1696;received=75.XXX.XXX.158. To: sip:9xx27x2...@irock.com;tag=155c340f586c28d0300cf5a6ccf90d99-5ddb. From: Moo sip:9xx22x1...@irock.com;tag=3e7dfbd6. Call-ID: OTZiY2IxZDU1YWNlNjRkYTAxZDViMTIwN2U0ZTNlZjE.. CSeq: 2
Re: [OpenSIPS-Users] destroy dialog on transaction timeout and load balancer
Howdy Bogdan, We've currently been using option 2 to reasonable success (a shell script / cron), so we'll probably stick with that in the future. I think playing around with the new rtpproxy session timeout settings might also help keep the problem in check. Thanks, and thanks for the recent release! Bobby Smith On Tue, Dec 21, 2010 at 9:02 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Bobby, Form dialog state machine point of view, an dialog without 200OK ACK is ok, it will not timeout it. So, I see 2 future solutions: 1) we change the code so that (optional) timeout and dialog termination will be done for missing ACK also 2) make a small shell script that takes the list of dialogs (opensipsctl fifo dlg_list), greps for the dialogs in CONFIRMED BUT NOT ACKED state and if lifetime longer than X sec, it terminates the dialog via opensipsctl fifo dlg_end_dlg Regards, Bogdan Bobby Smith wrote: Is there an easy way/example from a scripting perspective to check if a dialog's status is 3 (not received an ACK yet), after a certain period of time, time out ONLY dialogs in this state? I've identified a situation where, if the UAC goes unresponsive, and never sends an ACK, we will eventually stop retransmitting the 200 OK to this but the dialog will remain established in memory. Because of this, the load balancer resources list becomes highly inaccurate over time. I'm hesitant to set the dialog default timeout flag, because there are certain situations where a long call (6 hours +) could occur, and we need resource accuracy to be 1 hour ish. Thanks, Bobby Smith ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ 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] receive port during call process
Hello, Bogdan! modparam(rr, enable_double_rr, 1) helps me. Before this parameter had value 0. Thank you. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, December 21, 2010 5:32 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] receive port during call process Hi Denis, The ACK is routed based on the RR headers that were collected by INVITE and returned in 200 OK. For such a scenario to work, opensips is doing double routing (adds for itself 2 RR headers, one with the inbound interface, one with the outbound interface). Could you post the SIP capture of such a call to check if correct from SIP point of view? Regards, Bogdan Denis Putyato wrote: Hello! I have a such problem. Opensips using 2 ports One – 5068 for client which must register on Opensips Second – 5060 for all other clients. 1) Client А registering on Opensips (socket: udp:2.2.2.2:5068). Client A is behind NAT. 2) Client А receives incoming call (via lookup() function). 3) Call has such way Cisco (source port 1 sends INVITE to port 5060 of Opensips) – Opensips (receives INVITE from Cisco to port 5060 and sends the INVITE to client from source port 5068 to some client`s port) – Client A. Everything work fine until client А answers. Then Client A sends to Opensips (port 5068) 200 OK, Opensips retransmit it to Cisco (from port 5060 to 1), Cisco sends to Opensips ACK (from port 1 to 5060) AND Opensips retransmit this ACK to client`s port FROM PORT 5060, BUT NOT 5068. This ACK didn`t reach client A because of NAT. Thank you for any help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA www.voice-system.ro ___ 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
