Re: [OpenSIPS-Users] Registrar not saving received from Path header
Hello Bogdan,
I added the patch and here is what I found: OpenSips[4378]: [ID 269964
local1.debug] DBG:registrar:pack_ci: xXx - flags are 0. I have also
included the log file.
Thanks
Nathaniel Keeling
On 5/16/13 3:47 AM, Bogdan-Andrei Iancu wrote:
Hello Nathaniel,
Attached is an extended patch - remove the old one and apply this one.
Again look for any xXx logs .
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/14/2013 02:47 PM, Nathaniel L Keeling III wrote:
Hello Bogdan,
here is the output from opensips's og file of the save() with the
patch and the code snippet from the opensips.cfg. I did not see any
ant logs with xXx. Also,I have usrloc's db_mode set to 3.
xlog(SAVING THE SUBSCRIBER INTO THE LOCATION TABLE
);
if (!save(location,p1))
{
xlog(L_ERR, ERR:callerid:$ci|end|System error trying to
save Register's request location);
sl_reply_error();
}
xlog(L_NOTICE, NOTICE:callerid:$ci|end|The subscriber has
successfully registered with Akan Voice);
exit;
May 16 23:35:53 OpenSips[4378]: [ID 292666 local1.debug] DBG:core:parse_msg:
SIP Request:
May 16 23:35:53 OpenSips[4378]: [ID 776402 local1.debug] DBG:core:parse_msg:
method: REGISTER
May 16 23:35:53 OpenSips[4378]: [ID 700387 local1.debug] DBG:core:parse_msg:
uri: sip:akanvoice.com
May 16 23:35:53 OpenSips[4378]: [ID 641661 local1.debug] DBG:core:parse_msg:
version: SIP/2.0
May 16 23:35:53 OpenSips[4378]: [ID 497291 local1.debug]
DBG:core:parse_headers: flags=2
May 16 23:35:53 OpenSips[4378]: [ID 202288 local1.debug]
DBG:core:parse_via_param: found param type 232, branch = z9hG4bK69354;
state=16
May 16 23:35:53 OpenSips[4378]: [ID 218084 local1.debug] DBG:core:parse_via:
end of header reached, state=5
May 16 23:35:53 OpenSips[4378]: [ID 636936 local1.debug]
DBG:core:parse_headers: via found, flags=2
May 16 23:35:53 OpenSips[4378]: [ID 481110 local1.debug]
DBG:core:parse_headers: this is the first via
May 16 23:35:53 OpenSips[4378]: [ID 994387 local1.debug]
DBG:core:parse_headers: header field type 1, name=Via, body=SIP/2.0/UDP
209.252.110.38:5060;branch=z9hG4bK69354
May 16 23:35:53 OpenSips[4378]: [ID 499462 local1.debug] DBG:core:parse_msg:
first via: SIP/2.0/UDP 209.252.110.38:5060(5060)
May 16 23:35:53 OpenSips[4378]: [ID 573780 local1.debug] DBG:core:parse_msg:
;branch=z9hG4bK69354
May 16 23:35:53 OpenSips[4378]: [ID 937246 local1.debug] DBG:core:parse_msg:
May 16 23:35:53 OpenSips[4378]: [ID 979351 local1.debug] DBG:core:parse_msg:
exiting
May 16 23:35:53 OpenSips[4378]: [ID 911547 local1.debug]
DBG:core:receive_msg: After parse_msg...
May 16 23:35:53 OpenSips[4378]: [ID 451678 local1.debug]
DBG:core:receive_msg: preparing to run routing scripts...
May 16 23:35:53 OpenSips[4378]: [ID 497291 local1.debug]
DBG:core:parse_headers: flags=40
May 16 23:35:53 OpenSips[4378]: [ID 202288 local1.debug]
DBG:core:parse_via_param: found param type 234, received = 208.54.44.253;
state=6
May 16 23:35:53 OpenSips[4378]: [ID 202288 local1.debug]
DBG:core:parse_via_param: found param type 235, rport = 44494; state=6
May 16 23:35:53 OpenSips[4378]: [ID 202288 local1.debug]
DBG:core:parse_via_param: found param type 232, branch = z9hG4bK69354;
state=16
May 16 23:35:53 OpenSips[4378]: [ID 218084 local1.debug] DBG:core:parse_via:
end of header reached, state=5
May 16 23:35:53 OpenSips[4378]: [ID 636936 local1.debug]
DBG:core:parse_headers: via found, flags=40
May 16 23:35:53 OpenSips[4378]: [ID 903840 local1.debug]
DBG:core:parse_headers: parse_headers: this is the second via
May 16 23:35:53 OpenSips[4378]: [ID 994387 local1.debug]
DBG:core:parse_headers: header field type 1, name=Via, body=SIP/2.0/UDP
100.229.65.174:43669;received=208.54.44.253;rport=44494;branch=z9hG4bK69354
May 16 23:35:53 OpenSips[4378]: [ID 994387 local1.debug]
DBG:core:parse_headers: header field type 8, name=Max-Forwards, body=30
May 16 23:35:53 OpenSips[4378]: [ID 218084 local1.debug] DBG:core:parse_to:
end of header reached, state=10
May 16 23:35:53 OpenSips[4378]: [ID 841317 local1.debug] DBG:core:parse_to:
display={}, ruri={sip:nkeel...@akanvoice.com}
May 16 23:35:53 OpenSips[4378]: [ID 993225 local1.debug]
DBG:core:get_hdr_field: To [30]; uri=[sip:nkeel...@akanvoice.com]
May 16 23:35:53 OpenSips[4378]: [ID 159376 local1.debug]
DBG:core:get_hdr_field: to body [sip:nkeel...@akanvoice.com^M
May 16 23:35:53 ]
May 16 23:35:53 OpenSips[4378]: [ID 994387 local1.debug]
DBG:core:parse_headers: header field type 3, name=To,
body=sip:nkeel...@akanvoice.com
May 16 23:35:53 OpenSips[4378]: [ID 994387 local1.debug]
DBG:core:parse_headers: header field type 4, name=From,
body=sip:nkeel...@akanvoice.com;tag=z9hG4bK97036383
May 16 23:35:53 OpenSips[4378]: [ID 994387 local1.debug]
DBG:core:parse_headers: header field type 6,
Re: [OpenSIPS-Users] Where to place acc_aaa_request ?
Via Radius using acc module, as you suggest before !
Michele
On 16/05/2013 16:13, Bogdan-Andrei Iancu wrote:
Well, do you want to do accouting via RADIUS (aaa) or via DB (in acc
table) ???
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/16/2013 01:18 PM, Michele Pinassi wrote:
Thanks Bodgan for your kindly reply but now accounting don't work:
nothing will be added to acc table !
Here's the full routing logic. Maybe there's something wrong:
modparam(aaa_radius, radius_config,
/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, early_media, 0)
modparam(acc, report_cancels, 0)
modparam(acc, detect_direction, 0)
modparam(acc, log_level, 1)
modparam(acc, aaa_url, radius:/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_flag, 1)
modparam(acc, aaa_extra, via=$hdr(Via[*]); \
Digest-User-Name=$Au; \
Calling-Station-Id=$from; \
Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ru; \
Sip-RPid=$avp(s:rpid); \
Source-IP=$avp(s:source_ip); \
Source-Port=$avp(s:source_port); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
Canonical-URI=$avp(s:can_uri); \
Divert-Reason=$avp(s:divert_reason); \
User-Agent=$hdr(user-agent); \
Contact=$hdr(contact); \
Event=$hdr(event) ;\
ENUM-TLD=$avp(s:enum_tld))
### Routing Logic
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
if (msg:len = 2048 ) {
sl_send_reply(513, Message too big);
exit;
};
if (check_address(4,$si,$sp,$proto)) {
#xlog(L_INFO,IP $si Allowed);
} else {
xlog(L_INFO,IP $si Forbidden);
sl_send_reply(403, Forbidden);
}
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
setflag(1);
} else if (is_method(INVITE)) {
record_route();
}
route(1);
} else {
/* uncomment the following lines if you want to enable
presence */
if (is_method(SUBSCRIBE) $rd == voip.unisi.it) {
route(2);
exit;
}
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
t_relay();
exit;
} else {
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
if (is_method(INVITE)) {
setflag(1);
}
t_check_trans();
if (!(method==REGISTER) is_from_local())
{
if(!check_source_address(0)){
if (!proxy_authorize(, subscriber)) {
proxy_challenge(, 0);
exit;
}
if (!db_check_from()) {
sl_send_reply(403,Forbidden auth ID);
exit;
}
consume_credentials();
# caller authenticated
}
}
# preloaded route checking
if (loose_route()) {
xlog(L_ERR, Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}
# record routing
if (!is_method(REGISTER|MESSAGE))
record_route();
if (!uri==myself) {
append_hf(P-hint: outbound\r\n);
route(1);
}
if( is_method(PUBLISH|SUBSCRIBE)) {
route(2);
}
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);
Re: [OpenSIPS-Users] OpenSIPS failover
Hi Jacek Konieczny, it would be great if we can control every service from one place. can you give me some or idea about how do i configure Heartbeat with pacemaker to have service-level control ? Please provide sample configuration or link if possible. Thanks Regards Juned -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-failover-tp4997077p7586372.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
Re: [OpenSIPS-Users] Odd opensips REGISTER/INVITE behavior for many simultaneous users
James, Old registrations should not be the problem - your problem (as I understand it) is missing registration, not too many :)... BTW, for the failed calls, do you get a 404 not found from scrip or a 408 timeout ? If you consider it, I can send you a script with an extension of usrloc to log when a new AOR is added or when a whole AOR is removed, so you can use it to doublecheck if your registrations are continuous in time . Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/16/2013 09:36 PM, James Tranovich wrote: Hello Bogdan, Thanks for your reply! We are using opensips 1.8.0-notls (x86_64/linux). We do not think this issue is load related but perhaps older registrations have not yet expired. We will try setting the min_expires parameter to a low number to test this hypothesis; any other approaches we could try? Thanks once again! James On Thu, May 16, 2013 at 1:53 AM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hello James, No, there is no such known bug or issue. What I suspect is that there are short intervals (milisecs) where an AOR is not registered - this may happen because : - the test tool is not performing properly under high load and fails to do re-register before old registration expires. - OpenSIPS is overloaded (too few processes ?) and it is not able to process traffic in realtime (check the LOAD related statistics). What OpenSIPs versions are you using ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/15/2013 01:56 AM, James Tranovich wrote: Hello all -- First, we love opensips :) Lately, we have been running into a strange issue which seems to be related to handling a ton of REGISTER messages. Basically, we have a test script that tries to simulate about 50 to 100 simultaneous calls; they all register en masse and then randomly start placing calls to another test number (after a random time interval). Every once in a while, though, an INVITE won't go through because opensips apparently can't find that phone number. Oddly enough, if we do an opensipsctl online immediately before/after, that command shows that, in fact, the recipient's number is present and presumably already registered. SIP logs/ngrep tracing confirm this. I was wondering if this is a known bug. This behavior only happens when registering a certain number of calls at once; if we test with a low number of calls (10, say), this behavior does not happen. It may be that we are spamming opensips with too many REGISTER messages (authentication is required, so two REGISTER messages are sent, the first w/o auth, the second with auth). But I don't see why opensips should have problems with this. Any thoughts on this? Is this a known issue already? (Searching for this issue didn't yield much). Thanks! James ___ 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] VIA relay error using mhomed=1
Hello Qasim, So you have multiple interfaces in OpenSIPS - are all of them the same protocol ? Please try to post a SIP capture of the full call, to see how the RR part is done. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/16/2013 01:07 PM, qasimak...@gmail.com wrote: On further investigation i see that i only face this issue when both caller and callee are on the same network. If both are on separate network it works fine. Regards, Qasim On Thu, May 16, 2013 at 3:05 PM, qasimak...@gmail.com mailto:qasimak...@gmail.com qasimak...@gmail.com mailto:qasimak...@gmail.com wrote: yes. Regards, Qasim On Thu, May 16, 2013 at 2:50 PM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: And do you have UDP 202.152.203.195 port 6000 as listener defined in OpenSIPS ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/16/2013 12:32 PM, qasimak...@gmail.com mailto:qasimak...@gmail.com wrote: Hi Bodgan, Yes i see the following route header in my packet. Route: sip:622190004002@202.152.203.195:6000;lr;ftag=3b710c25;did=e55.a77ff685 And yes i am routing it through loose_route. Regards, Qasim On Wed, May 15, 2013 at 10:40 PM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hello Qasim, The ACK should be routed via loose_route() based on the Route headers from it. Could you check if the Route hdrs (from the ACK) are correctly reflecting your opensips interfaces ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/14/2013 07:55 AM, qasimak...@gmail.com mailto:qasimak...@gmail.com wrote: Hi, I am using OpenSIPs in Public-Private bridging mode and have enabled mhomed=1. But the problem is that when we have a call in which both parties are on Public interface the INVITE gets relayed properly but and ACK of that invite gives the following error. ERROR:core:get_out_socket: no socket found ERROR:core:forward_request: cannot forward to af 2, proto 1 no correspondinglistening socket Regards, Qasim ___ 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] Error 404 not here
Hello, Script look ok - what I suspect is that one of the end points messes up the Record-Routes in the messages, so the sequential requests (ACK, BYE) cannot be properly routed. To confirm this I need to see a SIP capture (ngrep) for the whole call, showing full messages (what you had in the first email was just an opensips log). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/16/2013 02:39 PM, sermj 2012 wrote: Dear Bodgan, Thank you for your fast response. Ya i am using default configuration file (opensips.cfg). below find the attachment for the same. And i am getting this '404 not here' response for the SIP request 'BYE' ,which you can see in previous message attachment file. Where 192.168.2.40 is my Opensips server , 192. 168.2.97 and 192.168.2.98 are Wi-Fi enabled VoIP clients. In this calling process,while after received the call, in callee its showing like 'waiting for ACK'. But the SMS is working fine with the same configurations. awaiting your reply, I hope you could help me. On Thu, May 16, 2013 at 2:00 PM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hello, For what SIP requests do you get the 404 not here reply ? Also, are you using the default opensips cfg file ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/16/2013 10:33 AM, sermj 2012 wrote: Hi All, My VoIP clients are registerd with opensips. When i tried to call other end client, i can hear only one way audio in clients (In caller),somewhere is going wrong and i am not getting audio (Voice) in both sides. By using tshark tracer i can see there is an '404 not here' status code. PS: I am working under standalone network infrastructure, and VoIP phones are Wi-Fi enabled. I just reinstalled my operating system (Ubuntu 10.04) and Opensips for several times,but unable to resolve this issue. what may be the reason for this '404 Not here' issue? and how can i solve it? Please help me, i have just stuck with this issue from last 15 days. Find the attachment, which is sip trace file. I hope you could help me... ___ 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] use of is_from_gw directive from drouting module.
Hello Miguel,
The "3" from the is_from_gw() matches the "type" column" from the
dr_gateways table - it has nothing to do with the dr_group table.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/16/2013 11:18 PM, Miguel J. López Valverde wrote:
Dear Bogdan-Andrei
I've tried to asign a new group for this providers, (group 3, I
can't asign it to group 0 because this group is used by another
type of users). I've filled dr_groups registers with this new
information, (dr_gateways was filled before, on 1.8 release):
select * from dr_gateways;
++-+--++---++---++---+
| id | gwid | type | address | strip |
pri_prefix | attrs | probe_mode | description |
++-+--++---++---++---+
| 1 | Provider_0 | 2 | XXX.XXX.XXX.XXX| 0 |
NULL | NULL | 0 | Provider0_incoming|
select * from dr_groups;
++---+-+-+---+
| id | username | domain | groupid |
description |
++---+-+-+---+
| 24 | Provider_0 | XXX.XXX.XXX.XXX| 3 |
Provider0_incoming |
And the new configuration:
} else if (is_from_gw()) {
# request comes from gw
setflag(21);
} else if (is_from_gw("3","n")) {
# request comes from gw with strange udp-ports
setflag(21);
}
With this configuration, the is_from_gw() directive is working
right for providers without dr_groups db information and using
5060 port, but the is_from_gw("3","n") aren't working, it don't
match with providers of group 3 and working with ports diferents
of 5060.
¿Are this configuration wrong?.
Thankyou very much for your help.
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-
-Mensaje original-
De: Bogdan-Andrei Iancu bog...@opensips.org
Para: OpenSIPS users mailling list users@lists.opensips.org
Cc: "\"Miguel J.\" López Valverde" mjlo...@smartic.es
Asunto: Re: [OpenSIPS-Users] use of is_from_gw directive
from drouting module.
Fecha: Wed, 15 May 2013 20:59:47 +0300
Hello Miguel,
Starting with 1.9, DR module does SIP wise resolving of the
destination (in order to find all the IPs behind the a FQDN, via
NATPR, SRV and A lookups). A side effect is that according to
SIP, no port means 5060.
In your case, the "n" flag should do the trick - but I
understand that when using it, your problem is what group to use
(by the way, "" group is translated to group 0 ) . Are your GWs
in various groups?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/15/2013 04:16 AM, Miguel J. López Valverde wrote:
Dear UserList Opensips:
I used drouting with OpenSips 1.8.0 release, the gateways
list was in the dr_gateways table and no ports where
configurated in it. For incoming
Re: [OpenSIPS-Users] Where to place acc_aaa_request ?
That means you do it (from OpenSIPS perspective) via RADIUS, and the
configuration seems ok for that ; Could you confirm that OpenSIPS is
sending RADIUS packages to the RADIUS server ? The RARDIUS server is the
one responsible for writing in whatever file or DB the received data.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/17/2013 10:34 AM, Michele Pinassi wrote:
Via Radius using acc module, as you suggest before !
Michele
On 16/05/2013 16:13, Bogdan-Andrei Iancu wrote:
Well, do you want to do accouting via RADIUS (aaa) or via DB (in acc
table) ???
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/16/2013 01:18 PM, Michele Pinassi wrote:
Thanks Bodgan for your kindly reply but now accounting don't work:
nothing will be added to acc table !
Here's the full routing logic. Maybe there's something wrong:
modparam(aaa_radius, radius_config,
/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, early_media, 0)
modparam(acc, report_cancels, 0)
modparam(acc, detect_direction, 0)
modparam(acc, log_level, 1)
modparam(acc, aaa_url, radius:/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_flag, 1)
modparam(acc, aaa_extra, via=$hdr(Via[*]); \
Digest-User-Name=$Au; \
Calling-Station-Id=$from; \
Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ru; \
Sip-RPid=$avp(s:rpid); \
Source-IP=$avp(s:source_ip); \
Source-Port=$avp(s:source_port); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
Canonical-URI=$avp(s:can_uri); \
Divert-Reason=$avp(s:divert_reason); \
User-Agent=$hdr(user-agent); \
Contact=$hdr(contact); \
Event=$hdr(event) ;\
ENUM-TLD=$avp(s:enum_tld))
### Routing Logic
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
if (msg:len = 2048 ) {
sl_send_reply(513, Message too big);
exit;
};
if (check_address(4,$si,$sp,$proto)) {
# xlog(L_INFO,IP $si Allowed);
} else {
xlog(L_INFO,IP $si Forbidden);
sl_send_reply(403, Forbidden);
}
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
setflag(1);
} else if (is_method(INVITE)) {
record_route();
}
route(1);
} else {
/* uncomment the following lines if you want to enable
presence */
if (is_method(SUBSCRIBE) $rd == voip.unisi.it) {
route(2);
exit;
}
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
t_relay();
exit;
} else {
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
if (is_method(INVITE)) {
setflag(1);
}
t_check_trans();
if (!(method==REGISTER) is_from_local())
{
if(!check_source_address(0)){
if (!proxy_authorize(, subscriber)) {
proxy_challenge(, 0);
exit;
}
if (!db_check_from()) {
sl_send_reply(403,Forbidden auth ID);
exit;
}
consume_credentials();
# caller authenticated
}
}
# preloaded route checking
if (loose_route()) {
xlog(L_ERR, Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}
# record routing
if (!is_method(REGISTER|MESSAGE))
record_route();
if (!uri==myself) {
append_hf(P-hint: outbound\r\n);
route(1);
}
if( is_method(PUBLISH|SUBSCRIBE)) {
route(2);
}
if (is_method(REGISTER)) {
#
Re: [OpenSIPS-Users] Registrar not saving received from Path header
Hello Nathaniel,
That is odd.it's like you do not set the p1 flag
I tested and I with p1 flag I get:
May 17 14:05:03 [7944] DBG:registrar:pack_ci: xXx - flags are 10
Are you sure your script gets to the right save() ??
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/17/2013 09:37 AM, Nathaniel L Keeling III wrote:
Hello Bogdan,
I added the patch and here is what I found: OpenSips[4378]: [ID
269964 local1.debug] DBG:registrar:pack_ci: xXx - flags are 0. I have
also included the log file.
Thanks
Nathaniel Keeling
On 5/16/13 3:47 AM, Bogdan-Andrei Iancu wrote:
Hello Nathaniel,
Attached is an extended patch - remove the old one and apply this
one. Again look for any xXx logs .
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/14/2013 02:47 PM, Nathaniel L Keeling III wrote:
Hello Bogdan,
here is the output from opensips's og file of the save() with the
patch and the code snippet from the opensips.cfg. I did not see any
ant logs with xXx. Also,I have usrloc's db_mode set to 3.
xlog(SAVING THE SUBSCRIBER INTO THE LOCATION TABLE
);
if (!save(location,p1))
{
xlog(L_ERR, ERR:callerid:$ci|end|System error trying to
save Register's request location);
sl_reply_error();
}
xlog(L_NOTICE, NOTICE:callerid:$ci|end|The subscriber has
successfully registered with Akan Voice);
exit;
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Re: [OpenSIPS-Users] OpenSIPS failover
Heartbeat is old and probably deprecated by now. You need to use pacemaker/corosync Same project just reworked. Or cman for RHEL. http://www.linux-ha.org/doc/users-guide/users-guide.html Good luck On 5/17/13, juned jkhan6...@gmail.com wrote: Hi Jacek Konieczny, it would be great if we can control every service from one place. can you give me some or idea about how do i configure Heartbeat with pacemaker to have service-level control ? Please provide sample configuration or link if possible. Thanks Regards Juned -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-failover-tp4997077p7586372.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
Re: [OpenSIPS-Users] VIA relay error using mhomed=1
Hi Bodgan, I have sent you SIP capture in private as the server was on public IP. Regards, Qasim On Fri, May 17, 2013 at 3:50 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: ** Hello Qasim, So you have multiple interfaces in OpenSIPS - are all of them the same protocol ? Please try to post a SIP capture of the full call, to see how the RR part is done. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 05/16/2013 01:07 PM, qasimak...@gmail.com wrote: On further investigation i see that i only face this issue when both caller and callee are on the same network. If both are on separate network it works fine. Regards, Qasim On Thu, May 16, 2013 at 3:05 PM, qasimak...@gmail.com qasimak...@gmail.com wrote: yes. Regards, Qasim On Thu, May 16, 2013 at 2:50 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: And do you have UDP 202.152.203.195 port 6000 as listener defined in OpenSIPS ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 05/16/2013 12:32 PM, qasimak...@gmail.com wrote: Hi Bodgan, Yes i see the following route header in my packet. Route: sip:622190004002@202.152.203.195:6000;lr;ftag=3b710c25;did=e55.a77ff685 And yes i am routing it through loose_route. Regards, Qasim On Wed, May 15, 2013 at 10:40 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hello Qasim, The ACK should be routed via loose_route() based on the Route headers from it. Could you check if the Route hdrs (from the ACK) are correctly reflecting your opensips interfaces ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 05/14/2013 07:55 AM, qasimak...@gmail.com wrote: Hi, I am using OpenSIPs in Public-Private bridging mode and have enabled mhomed=1. But the problem is that when we have a call in which both parties are on Public interface the INVITE gets relayed properly but and ACK of that invite gives the following error. ERROR:core:get_out_socket: no socket found ERROR:core:forward_request: cannot forward to af 2, proto 1 no correspondinglistening socket Regards, Qasim ___ 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
Re: [OpenSIPS-Users] Where to place acc_aaa_request ?
Yes, the radius server receive the packets.
I saw them in the text log of freeradius server. Here's an entry example:
Fri May 17 13:54:11 2013
Acct-Status-Type = Stop
Service-Type = Sip-Session
Sip-Response-Code = 200
Sip-Method = Bye
Event-Timestamp = May 17 2013 13:54:11 CEST
Sip-From-Tag = da61bce06d
Sip-To-Tag = 6a3c0aa1e36e0c87i0
Acct-Session-Id = 9f4987d21afa6fc9
Digest-Attributes = 0x0a143530303540766f69702e756e6973692e6974
Calling-Station-Id = sip:5...@voip.unisi.it
Called-Station-Id = sip:2233@172.20.1.4
Sip-Translated-Request-URI = sip:2233@172.20.1.4:5060
User-Agent = Cisco/SPA502G-7.4.8a
NAS-Port = 5060
Acct-Delay-Time = 0
NAS-IP-Address = 127.0.0.1
Acct-Unique-Session-Id = de5f87e909fa9a63
Timestamp = 1368791651
But in the 'radacc' mysql table i have all calls (missed too).
Michele
On 17/05/2013 13:00, Bogdan-Andrei Iancu wrote:
That means you do it (from OpenSIPS perspective) via RADIUS, and the
configuration seems ok for that ; Could you confirm that OpenSIPS is
sending RADIUS packages to the RADIUS server ? The RARDIUS server is the
one responsible for writing in whatever file or DB the received data.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/17/2013 10:34 AM, Michele Pinassi wrote:
Via Radius using acc module, as you suggest before !
Michele
On 16/05/2013 16:13, Bogdan-Andrei Iancu wrote:
Well, do you want to do accouting via RADIUS (aaa) or via DB (in acc
table) ???
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/16/2013 01:18 PM, Michele Pinassi wrote:
Thanks Bodgan for your kindly reply but now accounting don't work:
nothing will be added to acc table !
Here's the full routing logic. Maybe there's something wrong:
modparam(aaa_radius, radius_config,
/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, early_media, 0)
modparam(acc, report_cancels, 0)
modparam(acc, detect_direction, 0)
modparam(acc, log_level, 1)
modparam(acc, aaa_url, radius:/etc/radiusclient-ng/radiusclient.conf)
modparam(acc, aaa_flag, 1)
modparam(acc, aaa_extra, via=$hdr(Via[*]); \
Digest-User-Name=$Au; \
Calling-Station-Id=$from; \
Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ru; \
Sip-RPid=$avp(s:rpid); \
Source-IP=$avp(s:source_ip); \
Source-Port=$avp(s:source_port); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
Canonical-URI=$avp(s:can_uri); \
Divert-Reason=$avp(s:divert_reason); \
User-Agent=$hdr(user-agent); \
Contact=$hdr(contact); \
Event=$hdr(event) ;\
ENUM-TLD=$avp(s:enum_tld))
### Routing Logic
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
if (msg:len = 2048 ) {
sl_send_reply(513, Message too big);
exit;
};
if (check_address(4,$si,$sp,$proto)) {
# xlog(L_INFO,IP $si Allowed);
} else {
xlog(L_INFO,IP $si Forbidden);
sl_send_reply(403, Forbidden);
}
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
setflag(1);
} else if (is_method(INVITE)) {
record_route();
}
route(1);
} else {
/* uncomment the following lines if you want to enable
presence */
if (is_method(SUBSCRIBE) $rd == voip.unisi.it) {
route(2);
exit;
}
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
t_relay();
exit;
} else {
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
if (is_method(INVITE)) {
setflag(1);
}
t_check_trans();
if (!(method==REGISTER) is_from_local())
{
if(!check_source_address(0)){
if
Re: [OpenSIPS-Users] Error 404 not here
Well, the traces are not complete - I do not see the starting INVITE. Anyhow, looking that BYE, I see it has no Route hdr, thing that confirms my suspicion on a bogus UA you are using. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/17/2013 04:22 PM, sermj 2012 wrote: Dear Bodgan, Thank you very much for your reply, Please find below the attachments, which are SIP captures (using ngrep). i hope this will give you a more complete picture. I believe your suspection, Please suggest me to solve this issue. I hope you could help me. On Fri, May 17, 2013 at 4:25 PM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hello, Script look ok - what I suspect is that one of the end points messes up the Record-Routes in the messages, so the sequential requests (ACK, BYE) cannot be properly routed. To confirm this I need to see a SIP capture (ngrep) for the whole call, showing full messages (what you had in the first email was just an opensips log). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/16/2013 02:39 PM, sermj 2012 wrote: Dear Bodgan, Thank you for your fast response. Ya i am using default configuration file (opensips.cfg). below find the attachment for the same. And i am getting this '404 not here' response for the SIP request 'BYE' ,which you can see in previous message attachment file. Where 192.168.2.40 is my Opensips server , 192. 168.2.97 and 192.168.2.98 are Wi-Fi enabled VoIP clients. In this calling process,while after received the call, in callee its showing like 'waiting for ACK'. But the SMS is working fine with the same configurations. awaiting your reply, I hope you could help me. On Thu, May 16, 2013 at 2:00 PM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hello, For what SIP requests do you get the 404 not here reply ? Also, are you using the default opensips cfg file ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/16/2013 10:33 AM, sermj 2012 wrote: Hi All, My VoIP clients are registerd with opensips. When i tried to call other end client, i can hear only one way audio in clients (In caller),somewhere is going wrong and i am not getting audio (Voice) in both sides. By using tshark tracer i can see there is an '404 not here' status code. PS: I am working under standalone network infrastructure, and VoIP phones are Wi-Fi enabled. I just reinstalled my operating system (Ubuntu 10.04) and Opensips for several times,but unable to resolve this issue. what may be the reason for this '404 Not here' issue? and how can i solve it? Please help me, i have just stuck with this issue from last 15 days. Find the attachment, which is sip trace file. I hope you could help me... ___ 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] Where to place acc_aaa_request ?
I just breezed through this email. and even I am not sure what seems to be the problem. I am going to assume that you want to log accounting using Radius and acc modules as discussed, and want to store only established calls in radacct. If you do not want to account for missed calls then comment out: modparam(acc, aaa_missed_flag, 2) modparam(acc, log_missed_flag, 2) # when routing via usrloc, log the missed calls also setflag(2); Kind Regards, Nick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Odd opensips REGISTER/INVITE behavior for many simultaneous users
Not sure if this would apply but for heavy registers you may want to employ mem-cache http://www.opensips.org/Documentation/Tutorials-MemoryCaching. We are about to integrate that in our system, and I thought of this inquiry. Hope this helps, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Odd opensips REGISTER/INVITE behavior for many simultaneous users
Just saw that localcache has been removed for 1.8. Please disregard the last message. Kind Regards, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Odd opensips REGISTER/INVITE behavior for many simultaneous users
The implementation has been moved to cacehdb http://www.opensips.org/html/docs/modules/devel/cachedb_memcached.html; post 1.7. Sorry for the Noise, Nick. On 5/17/13, Nick Khamis sym...@gmail.com wrote: Just saw that localcache has been removed for 1.8. Please disregard the last message. Kind Regards, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Slight problem routing 100s and 183s
Hello Nick, Last time I check Canada was still in the North Hemisphere, so summer should come :)We are just ending the spring here in Europe. So, the new box is the OUT one ? I'm asking as your trace shows the .5 IP which belong to the IN server From your description I do not really understand what is the actual problem - maybe you can off list send me a pcap of the call with some more details. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/16/2013 09:02 PM, Nick Khamis wrote: Hello Everyone, Hope all is well. Here in Canada our 2 weeks of summer is almost over and now it's almost autumn ;). Those of you who are in Montreal know what I am talking about Long story short, we love OpenSIPS so much that we decided to add another box between our media servers and our service provider, yielding a: NAT Box OpenSIPSIn -- Asterisk1...N OpenSIPSOut OpenSIPSIn: 192.168.2.5 Asterisk: 192.168.2.10 OpenSIPSOut: 192.168.2.20 Everything was working fine in our natted environment until we added OpenSIPSOut. Looking at the trace, I see a problem with via and rr. The trace from OpenSIPSIn: U 2013/05/16 13:12:53.978573 192.168.2.5:5060 - 192.168.2.10:5060 INVITE sip:15148392...@server.example.com:5060;user=phone SIP/2.0. Record-Route: sip:192.168.2.5;lr;did=66a.32d38963. Via: SIP/2.0/UDP 192.168.2.5:5060;branch=z9hG4bKd9a4.dfbf0a33.0. Via: SIP/2.0/UDP 192.168.2.11:5060;rport=5060;received=192.168.2.11;branch=z9hG4bKc3495b99FCBA96F0. Then the Giving a Try coming in from our services provider to OpenSIPSIn do not get responded to: U 2013/05/16 13:12:54.177744 10.5.2.13:5060 - 192.168.2.5:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.2.20:5060;received=79.12.11.7;branch=z9hG4bK5b6b.146a4f8.0. Via: SIP/2.0/UDP 192.168.2.10:5060;received=192.168.2.10;branch=z9hG4bK1225fb65;rport=5060. Givng a try Givng a try Givng a try . And so is the case for Session Progress coming in from our service provider: U 2013/05/16 13:13:07.655052 10.5.2.13:5060 - 192.168.2.5:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK5b6b.146a4f8.0. Via: SIP/2.0/UDP 192.168.2.10:5060;received=192.168.2.10;branch=z9hG4bK1225fb65;rport=5060. Record-Route: sip:192.168.2.20;lr;did=4e7.35cb3c86. Session Progress Session Progress Session Progress . To make things more interesting, asterisk creates a new callid when receiving the initial request from OpenSIPSIn: Call-ID: 16a8997f-217a7945-ca8ec106@192.168.2.11. vs Call-ID: 1f5b92da3d973b2b7a6fb2752e8df585@192.168.2.10:5060. In the past, we handled BYEs getting 404'ed by opensips because of the change in callid by explicitly forcing the dialog matching using match/validate/ and fix_route_dialog() (Thanks Vlad! ;) Would we force dialog matching for the 100 and 180's the same way we did for BYEs? If so where would be the safest place to do this. Thanks in Advance, Nick. ___ 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] memory consumed by t_relay
Hi, I do not see any Memory status (shm): in your logs - that is the part for dumping the shared memory (which you are suspecting of leaking). There are only logs for the pkg (private, per process) memory. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/17/2013 05:54 AM, microx wrote: Hi Bogdan-Andrei, Please refer to the log OpenSIP_outbound_memory.log http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7586368/OpenSIP_outbound_memory.log Many thanks for your help. Best regards, Chen-Che -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/memory-consumed-by-t-relay-tp7586016p7586368.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
[OpenSIPS-Users] OpenSIPS/STUN
Hi, I want to use OpenSIPS as the proxy/load balancer for a bunch of FreeSWITCH media servers. Some of the UACs will be in private networks behind NATs. My question is this: Can OpenSIPS be setup to change the IP/Port info in SDP to the translated one before sending the SIP request on to the FreeSWITCH server. Or do I have to use a separate STUN server and have my UACs get the translated addresses and include them in SDP info? Thank you *Oleg* ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Slight problem routing 100s and 183s
Bogdan, I see how busy you are with OpenSIPS so I will make it count. Yes OpenSIP-Out is the new box that we have put in place to: Bellow is a quick network diagram. The issue we are experiencing is that the 100s, 183s and 200s that come back from the carrier do not get processed or even responded to by OpenSIPS-In. The complete sip trace for OpenSIPS-In can be found at http://pastebin.com/iGeWsc40;. I did not include anything for OUT since it is performing as expected. Some things to notice are the changed CallID. This is done by asterisk (192.168.2.10): Initial: Call-ID: 4737d441-5fb15ea7-7142c0d8@192.168.2.11. Modified: Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060. And the vanishing of RR: Record-Route: sip:192.168.2.5;lr;did=b82. 180aabc6. This is also due to asterisk's recreation of the initial INVITE. When it comes to network appliances, this is the last piece of the pie. From now on it's mainly business logic, which should be less of a learning curve for us!!! I decided to post my problem online with example values, so it would hopefully help someone in the future. Kind Regards, Nick. [image: network.jpg]https://mail.google.com/mail/ca/?ui=2ik=e9f48992abview=attth=13eb42dafefa444eattid=0.1disp=inlinerealattid=f_hgttk2a11safe=1zw network.jpg___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
