Please show a piece of opensips.cfg where you calling record_route() and SIP
debug of such call
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Jesse Cloutier
Sent: Wednesday, July 20, 2011 7:08 PM
To:
I think there were more error strings in log during compile process. Show all
error strings.
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Mark Holloway
Sent: Wednesday, July 06, 2011 11:23 AM
To: OpenSIPS users mailling
’
b2b_logic.c:624: error: too many arguments to function ‘process_bridge_action’
make[1]: *** [b2b_logic.o] Error 1
make[1]: Leaving directory `/usr/src/opensips-1.6.0-tls/modules/b2b_logic'
make: *** [modules] Error 2
On Jul 6, 2011, at 12:25 AM, Denis Putyato wrote:
I think there were more error
.
Regards,
Anca
On 06/14/2011 06:57 AM, Denis Putyato wrote:
Hello
I found that this problem appears when I use append_hf() to add some header in
local route of the proxy1 before sending INVITE to proxy2. Without this adding
problem disappears.
From: users-boun...@lists.opensips.org
Hello
I found that this problem appears when I use append_hf() to add some header in
local route of the proxy1 before sending INVITE to proxy2. Without this adding
problem disappears.
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis
Hello!
I have a such problem opensips1.6.4-2
There are two proxies of version 1.6.4.-2 which has been installed on the same
server.
One proxy (proxy1) using B2B “top hiding” and located in /usr/local/sbc and
using one signaling port
Another proxy (proxy2) is just SIP proxy and
Hello everybody!
Please give me some information about localcach module
Is it use shmem of Opensips for storing data?
Thank you
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Iancu [mailto:bog...@opensips.org]
Sent: Tuesday, May 10, 2011 12:43 PM
To: Denis Putyato
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] dialog and acc
Hi Denis,
I can tell it works for sure, as I'm using this kind of dirty trick to cope
with some buggy clients.
Best
Bogdan
I understand, thank you
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Tuesday, May 10, 2011 12:56 PM
To: Denis Putyato
Cc: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] dialog and acc
Hi Denis,
Yes, before the loose_route(), and it is critical
Hello!
Thank you Bogdan, I will try you decision
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Friday, May 06, 2011 9:25 PM
To: OpenSIPS users mailling list
Cc: Denis Putyato
Subject: Re: [OpenSIPS-Users] dialog and acc
Hi Denis,
From a proxy point of view, a 200OK
Hello
Try to use dialplan module
http://www.opensips.org/html/docs/modules/devel/dialplan.html#id249075
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Jan D.
Sent: Wednesday, May 04, 2011 1:00 PM
To:
Hello!
I noticed that cdr_flag in acc modules marks dialog for accounting as answered
even there was no ACK on 200 OK.
As a result, I have acc record which has a big duration and status 200 OK.
Thank you for any help.
___
Users
Hello
Try to use failed_transaction_flag
http://www.opensips.org/html/docs/modules/devel/acc.html#id292642
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Wesley Volcov
Sent: Thursday, April 28, 2011 4:13 PM
To: OpenSIPS users mailling list
I am using and failed_transaction_flag and db_missed_flag
for marking calls and I do not see any problem with sip_code while using
next_routing()
From: Wesley Volcov [mailto:wesleyvol...@gmail.com]
Sent: Thursday, April 28, 2011 4:33 PM
To: Denis Putyato
Cc: OpenSIPS users mailling list
Hello!
Sorry if my questions already appeared in mail list but
1) If I don`t use timeout in cache_store func. then record in cache will
live “forever” ?
2) If I try to cache_store attribute which already has record in cache
then this attribute will be rewritten?
Thank you
rather than the opensips
anyway, yes on both of them :)
On 19 April 2011 15:12, Denis Putyato denis7...@mail.ru wrote:
Hello!
Sorry if my questions already appeared in mail list but
1) If I don`t use timeout in cache_store func. then record in cache will
live “forever” ?
2
: Denis Putyato
Subject: Re: [OpenSIPS-Users] dialog and CANCEL
Hi Denis,
On 03/21/2011 01:55 PM, Denis Putyato wrote:
Hello
There is such scheme of call
One gateway – 1.1.1.1
Opensips – 2.2.2.2
Another gateway – 3.3.3.3
Calls from 1.1.1.1 to 3.3.3.3 through 2.2.2.2
I use CDR_flag
Hello
In SIP trace
1.1.1.1– callee
2.2.2.2 – Opensips
3.3.3.3 – callee
I have Opensips 1.6.4-2.
….
modparam(dialog, hash_size, 4096)
modparam(dialog, log_profile_hash_size, 12)
modparam(dialog, default_timeout, 1800)
modparam(dialog, timeout_avp, $avp(i:995))
Hello
There is such scheme of call
One gateway – 1.1.1.1
Opensips – 2.2.2.2
Another gateway – 3.3.3.3
Calls from 1.1.1.1 to 3.3.3.3 through 2.2.2.2
I use CDR_flag for accounting
A piece of script config:
…
modparam(dialog, default_timeout, 1800)
…
…
onreply_route[1] {
if
Hello
I attached an ngrep dump.
After this call I have accounting record with duration 1800 sec (I am using CDR
flag accounting).
(…
modparam(dialog, default_timeout, 1800)
…)
Calls from 1.1.1.1 to 3.3.3.3 through 2.2.2.2
2.2.2.2 – Opensips
1.1.1.1 and 3.3.3.3 – some gateways
Hello
I had the same problem on 1.6.4, you should use 1.6.4-2 version
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Iulian Macare
Sent: Friday, March 11, 2011 12:24 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] opensips 1.6.4 out
Of Bogdan-Andrei Iancu
Sent: Friday, March 04, 2011 6:58 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Script flag question
Hi Denis,
are you sure that your script flow does not hit any of those exit
statements before getting to the second xlog() ?
Regards,
Bogdan
Denis Putyato
Hello, Bogdan
When can I see information about the bug?
Thank you
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Friday, February 25, 2011 6:33 PM
To: OpenSIPS users mailling list
Subject: Re:
Thank you very much!
-Original Message-
From: sip.n...@gmail.com [mailto:sip.n...@gmail.com] On Behalf Of Ovidiu Sas
Sent: Thursday, March 03, 2011 4:14 PM
To: OpenSIPS users mailling list
Cc: Denis Putyato
Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB
app callback hooks not saved to DB
Hi Denis,
This is work on progress from Razvan - he will update you soon.
Regards,
Bogdan
Denis Putyato wrote:
Hello, Bogdan
When can I see information about the bug?
Thank you
-Original Message-
From: users-boun...@lists.opensips.org
list
Subject: Re: [OpenSIPS-Users] Dialog module app callback hooks not saved to DB
Hello Denis,
I have just added a patch in trunk that fixes this issue. Please update
your sources and let me know if the problem persists.
Regards,
Razvan
On 03/03/2011 03:21 PM, Denis Putyato wrote:
Thank you
module app callback hooks not saved to DB
Hello Denis,
The patch makes some changes to acc and dialog modules. Please make sure
you re-built them both.
Regards,
Razvan
On 03/03/2011 05:11 PM, Denis Putyato wrote:
Hello Razvan
During restart
Mar 3 18:08:38 opensips /usr/local/opensips1.6.4
to appear that
error. Please try to restart OpenSIPS with a fresh dialog ongoing and
let me know if the ERROR still appears.
Regards,
Razvan
On 03/03/2011 06:00 PM, Denis Putyato wrote:
Razvan
I re-built them both and receive the ERROR
-Original Message-
From: users-boun
.
Regards,
Bogdan
Denis Putyato wrote:
Hello Bogdan
- Server:: OpenSIPS (1.6.4-2-notls (x86_64/linux))
- fork = yes (default)
-
listen=xxx.xxx.xxx.xxx:5060
listen=xxx.xxx.xxx.xxx:5068
- i do not use tcp
-
loadmodule db_mysql.so
loadmodule sl.so
loadmodule tm.so
loadmodule signaling.so
...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, March 01, 2011 11:47 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] dialog statistic problem
Denis,
this xXx marker appears only ones, right ?
Regards,
Bogdan
Denis Putyato
once.
After a restart, please send me:
1) the two xXx logs (with the pid)
2) output of opensipsctl fifo ps
I want to identify what is the second process triggering the xXx
Thanks and regards,
Bogdan
Denis Putyato wrote:
No, during restart xXx appears twice
$cat /var/log/opensips | grep xXx
,
Bogdan
Denis Putyato wrote:
Bogdan, since Mar 1 07:55:36 I didn't restart Opensips so
Mar 1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips[11382]:
CRITICAL:dialog:child_init: xXx - active dialogs=2 , early=0
Mar 1 07:55:36 opensips /usr/local/opensips1.6.4-2/sbin/opensips
users mailling list
Subject: Re: [OpenSIPS-Users] dialog statistic problem
Hi Denis,
What revision number are you using ?
Also, do you have fork enabled?
Do you use multiple UDP interfaces?
Do you have TCP enabled ?
What are the modules you are using ?
Regards,
Bogdan
Denis Putyato wrote
Hello!
I found active dialog statistic problem
For example in dialog::active_dialogs I have 130 active calls, after restart
Opensips I already have dialog::active_dialogs nearly 260, i.e. the number of
active dialogs increase by 2.
Meanwhile, after restart, opensipsctl fifo dlg_list
Hello!
Opensips 1.6.4-2, MySQL installed on the same server as opensips.
Please can somebody explain why such message can appear in syslog? This happens
when I make “opensipsctl fifo dp_reload” after long period of time nothing to
do with opensips.
During processing calls opensips make
, use fix_nated_contact() for INVITE and 200 OK,
so that the received contact will be fixed with the layer3 IP, so the
dialog module will use the contact with a useful info.
Regards,
Bogdan
Denis Putyato wrote:
Hello!
I am using dialog module for control of call duration.
When timeout
Hello!
I am using dialog module for control of call duration.
When timeout of dialog expires I need Opensips send BYE not to caller and
callee contact (which is stored during creation of dialog) but to IP address
and port from which INVITE (caller) and 200 OK (callee) had been received.
Hello
Try to modify your code
if(!check_source_address(0)){
sl_send_reply(403, Forbidden);
exit;
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Alejandro Recarey
Sent: Monday, January
notification
will work, but you will have the same timeout in both situations.
Regards,
Razvan
On 01/12/2011 07:38 AM, Denis Putyato wrote:
Hello Razvan,
“OpenSIPS shouldn't even try to terminate the call because it isn't established
yet”
As I understand I just do not need to use –W key
the callee doesn't answer you can use
the tm module and set the fr_inv_timer parameter. You can get more details
from:
http://www.opensips.org/html/docs/modules/devel/tm.html#id250344
Regards,
Razvan
On 01/12/2011 02:38 PM, Denis Putyato wrote:
Razvan, I got rtpproxy from http://opensips.org
-January/thread.html
Regards,
Razvan
On 12/28/2010 04:52 PM, Denis Putyato wrote:
Hello Bogdan
RTP Proxy is working but timeout notification does not.
There is error /usr/local/opensips1.6.4/sbin/opensips[26496]:
DBG:nathelper:timeout_listener_process: unknown rtpproxy – ignoring
Hello!
I try patch rtpproxy gotten from git. And there is such error during patching
patch rtpproxy_timeout_notification.patch
patching file main.c
Hunk #1 succeeded at 70 (offset 2 lines).
Hunk #2 FAILED at 120.
Hunk #3 succeeded at 132 with fuzz 1 (offset 4 lines).
Hunk #4
://opensips.org/pub/rtpproxy/.
Regards,
Razvan
On 1/11/2011 2:19 PM, Denis Putyato wrote:
Hello!
I try patch rtpproxy gotten from git. And there is such error during patching
patch rtpproxy_timeout_notification.patch
patching file main.c
Hunk #1 succeeded at 70 (offset 2 lines).
Hunk #2 FAILED
Iancu
Sent: Tuesday, December 28, 2010 1:31 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications
Hi Denis,
Silly question, but have you applied to the official RTPproxy the
patches that comes with the nathelper module ?
Regards,
Bogdan
Denis Putyato
...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Tuesday, December 28, 2010 5:49 PM
To: OpenSIPS users mailling list; Razvan Crainea
Subject: Re: [OpenSIPS-Users] RTPProxy timeout notifications
Hi Denis,
Denis Putyato wrote:
Hello Bogdan!
1) There is no patch in official release
Hello!
During tests of new feature in rtpproxy I received such problem:
“Dec 27 11:42:42 opensips /usr/local/opensips1.6.4/sbin/opensips[26496]:
DBG:nathelper:timeout_listener_process: unknown rtpproxy – ignoring”
And log for rtpproxy
“Dec 28 07:43:37 opensips rtpproxy[28223]:
Hello!
Can anybody explain me how to use such command?
Do I need fill all parameters or I may fill just AOR and contact, for example?
Just when I try it I receive “400 Too few or too many arguments” or “400 Bad
parameter” and cannot understand what is wronge
Thank you
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
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
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:
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
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,
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
On 20/12/10 13:51, 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
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
Hello everybody!
There was a news in web site of Opensips about ACC module and CDR generation
http://lists.opensips.org/pipermail/news/2010-August/96.html
I use 1.6.3. Can I already install ACC for the Opensips with new feature? Or it
will be accessed only in next release of
.
The feature to directly generate CDRs in available in OpenSIPS trunk
(devel) and it will be available in opensips 1.6.4 stable (in mid December)
Regards,
Bogdan
Denis Putyato wrote:
Hello everybody!
There was a news in web site of Opensips about ACC module and CDR
generation
http
Hello Anton
And a reason of this? And what do you mean while ACC logging?
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Anton Zagorskiy
Sent: Monday, December 06, 2010 5:27 PM
To: 'OpenSIPS users mailling list'
Subject:
Yes, you are right there is no information about status 2 in doc. But as I
understand status 2 exists during time after create dialog and until final
reply.
Then status can be 3,4 (if final reply received) or 5(if there is no final
reply).
-Original Message-
From:
Hello!
Please tell me where in dr_rules table rule_attrs_avp (str) of DROUTING module
store?
http://www.opensips.org/html/docs/modules/1.6.x/drouting.html#id294102
Thank you.
___
Users mailing list
Users@lists.opensips.org
And what about
http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id250384
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bruce Borrett
Sent: Tuesday, November 16, 2010 1:40 PM
To: Users@lists.opensips.org
Subject: [OpenSIPS-Users] Timer
that you will need to stick to manual accounting.
Regards,
Ovidiu Sas
On Tue, Nov 16, 2010 at 1:02 AM, Denis Putyato denis7...@mail.ru wrote:
Thank you for reply
First variant is not quite flexible for me.
The second variant more interesting, but it doesn't work
A piece of code from opensips.cfg
/tm.html#id293687
Regards,
Ovidiu Sas
On Tue, Nov 16, 2010 at 12:30 AM, Denis Putyato denis7...@mail.ru wrote:
Hello!
Is there any chancy for accounting calls which were finished by sending
failure code using send_reply() func.?
Thank you
___
Users
Hello !
Please can somebody tell me how “ip_conntrack_udp_timeout_stream” works with
mediaproxy?
For example.
SIP client A à Opensips with mediaproxy à SIP client B
Call established. RTP send/receive for both clients. Then there is some problem
with ethernet for client A and he
and ip_conntrack_udp_timeout_stream
On 10/21/2010 12:30 PM, Denis Putyato wrote:
Hello !
Please can somebody tell me how “ip_conntrack_udp_timeout_stream” works
with mediaproxy?
For example.
SIP client A à Opensips with mediaproxy à SIP client B
Call established. RTP send/receive for both clients
] On Behalf Of Denis Putyato
Sent: Tuesday, July 27, 2010 8:11 PM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] FW: Error when setting OpenSips with Radius
Try to add “Sets” param. with User-Name attribute for acc_radius module in your
opensips.cfg
For example,
modparam(aaa_radius
= 6528822724
cisco-avpair = call-id=b3b259f5-f6fe-1810-86e0-001a803f2...@192.168.1.2
NAS-Port = 5060
NAS-IP-Address = 10.84.0.21
Thanks again.
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Denis Putyato
Sent: Wednesday, July 28, 2010 1
Try to add “Sets” param. with User-Name attribute for acc_radius module in your
opensips.cfg
For example,
modparam(aaa_radius,sets,set1 = (User-Name=$var(usr))”)
where $var(usr) is some PV of your callerid
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org]
Hello
Are you using regexp in repl_exp ?
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Антон Загорский
Sent: Tuesday, July 20, 2010 4:19 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Dialplan module
Hello.
Show the string in dialplan table wholly and what can you see in syslog while
process a call?
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Антон Загорский
Sent: Tuesday, July 20, 2010 4:39 PM
To: 'OpenSIPS users
to this email in case
you don't use the trunk version. Please let me know if there are any other
issues.
Regards,
Irina Stanescu
On Thu, Jul 15, 2010 at 3:29 PM, Denis Putyato denis7...@mail.ru wrote:
Bogdan, i made another call which makes opensips crash
...
Core was generated by `/usr/local
.
I added a fix on the trunk and i also attached the patch to this email in
case you don't use the trunk version. Please let me know if there are any
other issues.
Regards,
Irina Stanescu
On Thu, Jul 15, 2010 at 3:29 PM, Denis Putyato denis7...@mail.ru wrote:
Bogdan, i made another call
Subject: Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash
Hi Denis,
That's perfect - thank you.
Could you print in GDB the following values : vp , attr.
Regards,
Bogdan
Denis Putyato wrote:
Hello, Bogdan
Is this information you asked?
gdb /usr/local/opensips/sbin/opensips /core
GNU gdb
Hello everybody!
There is a problem with radius_send_auth(); function.
This function is called from request route and after opensips received
“Access-Accept” from radius server it is crashes with such error:
Jul 14 14:59:02 kam /usr/local/opensips/sbin/opensips[21556]:
14, 2010 7:37 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] ACC_RADIUS makes opensips crash
Hi Denis,
do you get a coredump file? if so, could you get a bracktrace from it
and post it here?
Regards,
Bogdan
Denis Putyato wrote:
Hello everybody!
There is a problem
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