rate. (updating, inserting
into databse is show then call rate)
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entry=0x0) at net/net_udp.c:448
#18 0x0041a9d3 in main_loop () at main.c:722
#19 main (argc=, argv=) at main.c:1259
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in/2.2
7e2d6e4..c63e14d master -> origin/master
Updating f880642..66ae29f
Fast-forward
modules/sst/sst_handlers.c | 42
+-
modules/sst/sst_handlers.h |1 +
2 files changed, 34 insertions(+), 9 deletions(-)
root@sp01:/home/openips/opensips_1_11#
options = 0x81e78f8 f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:
ret = -1
seed = 1704724837
rfd = optimized out
__FUNCTION__ = main
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http
for this kind of services.
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:18, Dani Popa wrote:
Hi,
t_uac_dlg with socket 'tcp:x.x.x.x' should work ?
When i try to use t_uac_dlg with socket 'tcp:x.x.x.x' i see that the
SIP message is sent over udp.
Thanks
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Hi,
t_uac_dlg with socket 'tcp:x.x.x.x' should work ?
When i try to use t_uac_dlg with socket 'tcp:x.x.x.x' i see that the SIP
message is sent over udp.
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), each watcher_username should receive a NOTIFY,
this is how i should understand this table ?
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Hi all,
There is any way to check if Opensips instance have dialog in any state
defined by Replaces Header of new incoming call ?
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There is any way to handle replay for sip keepalive OPTIONS packet when
using nathelper module ?
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Chougule
Cell: 08097989101
Skype-ID: aamir_ryu
--- Sent from my BlackBerry ---
-Original Message-
From: Dani Popa dani.p...@gmail.com
Sender: users-boun...@lists.opensips.org
Date: Fri, 13 Sep 2013 13:12:51
To: OpenSIPS users mailling listusers@lists.opensips.org
Reply
@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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- SYSSVOIP
www.syssvoip.com.br
55 3537 2030
2013/7/17 Dani Popa dani.p...@gmail.com
set opensips peer to insecure=port,invite
On Wed, Jul 17, 2013 at 1:12 PM, Willian Mazzardo - SYSSVOIP
will...@syssvoip.com.br wrote:
Hi Stephens... how do I do this?
Willian Mazzardo
Depto TI - SYSSVOIP
)
Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br
55 3537 2030
2013/7/17 Dani Popa dani.p...@gmail.com
what contex hit invite from opensips ?
On Wed, Jul 17, 2013 at 1:24 PM, Willian Mazzardo - SYSSVOIP
will...@syssvoip.com.br wrote:
Hi Dani ... thanks ... i have for now
places and opensips does not show it to you
unless you have debug on.
Regards,
Qasim
On Thu, Jun 20, 2013 at 11:25 PM, Dani Popa dani.p...@gmail.com wrote:
any ideea ?
On Tue, Jun 18, 2013 at 7:10 PM, Dani Popa dani.p...@gmail.com wrote:
Hi all,
I use acc with radius and when i set
suggest you to use the manual accounting in this case.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 06/18/2013 07:10 PM, Dani Popa wrote:
Hi all,
I use acc with radius and when i set accountig flag in local_route i
dont receive any
any ideea ?
On Tue, Jun 18, 2013 at 7:10 PM, Dani Popa dani.p...@gmail.com wrote:
Hi all,
I use acc with radius and when i set accountig flag in local_route i
dont receive any accountig request on radius server. As I see local_route
was hit twice on dialog timeout and i dont understand
,
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, LatePackets, LostPackets. I know,
some of you will recomand mediaproxy and it's not good for me, because i
chosed to use rtpproxy because, i can insert and record media in curent
stream. So the question is: there is any way to have such information at
the end of call?
Thanks,
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)
(A)trying -opensips -trying(B)
(A)ringing -opensips -ringing(B)
(A)progress -opensips
(A)200ok -opensips -200OK(B)
(A) ACK -opensips -ACK(B)
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)
(A)trying -opensips -trying(B)
(A)ringing -opensips -ringing(B)
(A)progress -opensips
(A)200ok -opensips -200OK(B)
(A) ACK -opensips -ACK(B)
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, LatePackets, LostPackets. I know,
some of you will recomand mediaproxy and it's not good for me, because i
chosed to use rtpproxy because, i can insert and record media in curent
stream. So the question is: there is any way to have such information at
the end of call?
Thanks,
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://lists.opensips.org/cgi-**bin/mailman/listinfo/usershttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
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, 2013 at 4:50 PM, Saúl Ibarra Corretgé
s...@ag-projects.comwrote:
On Feb 18, 2013, at 2:26 PM, Dani Popa wrote:
Hi,
I think it's more helpful if you can give us calltrace in case of using
msrp, sipproxy and of course 2 sip clients. Msrprelay it's act as a
mediaproxy or the sip client
/listinfo/users
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Unavailable);
};
}
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Five SIP clients with the same username.
Dani
On Fri, Feb 15, 2013 at 12:19 PM, Dani Popa dani.p...@gmail.com wrote:
Hi,
Regarding msilo module and example from the documentation, one simple
question:
if i have 5 clients already registered and non of them know IM(message sip
method
route is triggered when the transaction
fails).
So the final answer - one time.
Regards
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 02/15/2013 12:22 PM, Dani Popa wrote:
Five SIP clients with the same username.
Dani
On Fri, Feb 15, 2013 at 12
Hi,
I wondering if it posiible to add sdp on 180 ringing in order to play some
ringing tone. The ideea si that i want to play from rtpproxy with
rtpproxy_stream2uac/rtpproxy_stream2uas some music as ringback tone to
calling party if it's online.
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I just want to play media on replay route in case of 18[013] reply, so i'm sure
the user was alerted if i got one of them, i'm pretty sure is not the case from
the link below and also inserted media is not a fake ringback.
Thanks anyway!
Dani Popa
On Feb 13, 2013, at 0:56, Daniel Goepp d
media,
not just append an SDP to a 180.
Good luck though:)
-dg
On Tue, Feb 12, 2013 at 3:28 PM, Dani Popa dani.p...@gmail.com wrote:
I just want to play media on replay route in case of 18[013] reply, so
i'm sure the user was alerted if i got one of them, i'm pretty sure
As far as I know, opensips send 487, when receiving 200ok, when forking
On Dec 5, 2012 8:19 AM, M.Khaled W Chehab kche...@icucall.com wrote:
Dears ,
How to send a 487 request terminated and drop the call directly if
the UA send a cancel ,since now I am sending 200 canceling to UA and
://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Hi,
I know it's a weird question, but still, it is possible to add a delay
(let's say 5 seconds) for the first invite(somehow to increase post dial
delay with 5 seconds).
Thanks,
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be true network latency simulation.
On Tue, Nov 22, 2011 at 6:26 PM, Dani Popa dani.p...@gmail.com
mailto:dani.p...@gmail.com wrote:
Hi,
I know it's a weird question, but still, it is possible to add a
delay (let's say 5 seconds) for the first invite(somehow to
increase post
to insert
into a DB the new timeout_avp value.
[1] http://www.opensips.org/html/**docs/modules/devel/avpops.**
html#id250328http://www.opensips.org/html/docs/modules/devel/avpops.html#id250328
Regards,
Vlad Paiu
OpenSIPS Developer
On 10/31/2011 07:02 PM, Dani Popa wrote:
hi
hi,
it is possible somehow to change/update the dialog timeout_avp(value of
it) on the fly. Meaning, after the dialog is established, to change it
somehow from fifo ? I want to use It for simultaneous prepaid calls.
Thanks,
Dani
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hi,
load module presence_mwi
and
if(is_method(SUBSCRIBE)) {
if (!has_totag()) {
if (avp_check($hdr(Event), eq/message-summary/i)) {
rewritehostport(asterisk.host);
record_route();
if (!t_relay()) {
t_reply(500,
Hi,
You have a lot of invite there, and it's hard to follow a single call
trace. Can you post a single call trace? Do you make nat detection and
fix nat , do you use mediaproxy or nathelper to pass media behaind nat ?
Dani
On 10/27/11 09:53, Nick wrote:
Hello
It's my network
idoubs on
Him
if you look for asterisk tools, i think you should ask on asterisk
mailing list, not opensips.
Dani
On 10/24/11 11:54, Faisal Rehman wrote:
Hi
I am in search of an opensource/paid tool for the monitoring and
analysis of ASR ACD from Master.csv (of Asterisk), before that Sammy
to
extend such presence rules to session requests and implement an OpenSIPS module
to query them but it would be s a stretch of imagination.
Adrian
On Sep 23, 2011, at 3:05 PM, Dani Popa wrote:
Hi all,
Does opensips have implemented something like authorize_messages to authorize
IM by xcap
Hi all,
Does opensips have implemented something like authorize_messages to
authorize IM by xcap ?
Thanks,
Dani Popa
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not delete the information that
opensips has published with pua_dialoginfo because each record is
identified by a ETAG and when updating/inserting a match against this
ETAG is done. Please look closer in presentity table.
Regards,
Anca
On Thu, Sep 15, 2011 at 4:09 PM, Dani Popa dani.p...@gmail.com
Hi,
I'm using pua_dialoginfo to publish dialog info. My problem is that if
in the middle of call, my softphone will send PUBLISH, it will overwrite
the publish from dialog info, and i don't want this. Can you give me a
hint how should i avoid this overwriting, if it possible ?
Thanks,
Dani
methods the registering UA supports.
Regards,
Vlad Paiu
OpenSIPS Developer
On 09/13/2011 10:42 PM, Dani Popa wrote:
Hi all,
What does it mean methods=0x1F6F from register contact when i see it
with opensipsctl ul show, and how can i decode it ?
Contact::
sip:@x.x.x.x:xxx;transport=UDP;ob;q
Paiu
OpenSIPS Developer
On 09/13/2011 10:42 PM, Dani Popa wrote:
Hi all,
What does it mean methods=0x1F6F from register contact when i see it
with opensipsctl ul show, and how can i decode it ?
Contact::
sip:@x.x.x.x:xxx;transport=UDP;ob;q=;expires=525;flags=0x0;cflags=0x0;socket
Hi,
My opensips used for presence stoped with Segmentation fault.
root@test:/home# gdb opensips_1_6/opensips core
GNU gdb (GDB) 7.3-debian
Copyright (C) 2011 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later
http://gnu.org/licenses/gpl.html
This is free software: you
, could you please do
p row_vals[0]
and paste here the output ?
Regards,
Vlad Paiu
OpenSIPS Developer
On 09/14/2011 02:53 PM, Dani Popa wrote:
Hi,
My opensips used for presence stoped with Segmentation fault.
root@test:/home# gdb opensips_1_6/opensips core
GNU gdb (GDB) 7.3-debian
Copyright
On 09/14/2011 03:53 PM, Dani Popa wrote:
Hi,
(gdb) frame 0
#0 0xb6c2c3ee in get_rules_doc (user=0xbfb603ac, domain=0xbfb603bc,
type=2, rules_doc=0xb6c6a2a4) at xcap_auth.c:527
527 body.len = strlen(body.s);
(gdb) p row_vals[0]
value has been optimized out
(gdb)
On 09/14/11 15:43
Thanks,
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solution to add NAT knowledge to dlg_tophiding.c?
This seems like a lot of code to duplicate.
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Hi,
I found, i think, a good document about integrating xcap with presence.
Maybe some of you need this:
http://download.oracle.com/docs/cd/E17667_01/doc.50/e17669/cpt_concepts.htm
Dani
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are you sure that is not handled as retrasmision ? Do you see the times
that invite hit call_control ?
dani
On 09/07/11 14:00, Mino Haluz wrote:
Hi,
I'm using kamailio+callcontrol2.0.14 , and when kamailio receives
identical 3 INVITES, the callcontrol function never returns -3 (return
, set:
memlog=6
memdump=1
in order to get only the memory dump without all runtime logs from mem
debugger.
Regards,
Bogdan
On 08/19/2011 01:13 PM, Dani Popa wrote:
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS
Hi,
Thanks for your response. Right now PKG_MEM_POOL_SIZE is 8*1024*1024
and i have 33 users online using presence(it's right that any expire
timers regarding publish and notify are 60 seconds instead 3600 as it is
in documentation) . What value should i use for let's say, 100k users
using
compiled on 05:48:37 Aug 19 2011 with gcc 4.5.2
Thanks,
Dani Popa
On 08/19/11 17:31, Bogdan-Andrei Iancu wrote:
Hi Dani,
You can not have comments in multi-line assignments
So, instead of
DEFS+= $(extra_defs) \
.
-DCHANGEABLE_DEBUG_LEVEL \
#-DF_MALLOC
Thanks,
Dani
On 08/19/11 18:01, Bogdan-Andrei Iancu wrote:
Hi Dani,
In your case opensips will act as UAC (not server), so you need to
define your custom user_agent_header:
http://www.opensips.org/Resources/DocsCoreFcn17#toc96
Regards,
Bogdan
On 08/16/2011 03:12 PM, Dani Popa wrote
Hi,
I think you could use dialog profile, but not sure.
Dani
On 08/19/11 23:17, Robert Thomas wrote:
Hi,
I have a load balancer module to distribute calls among my
Gateways. I can use the lb_list command to see the active calls per gw, but I
would like something similar to graph my customer
Hi,
Where should i find memory dump ? I have something in logs about memory.
I'll attach an file. Please let me know if this is what you need.
I also increased PKG_MEM_POOL_SIZE = 8 *1024 * 1024, and shared mem to
256, and also updated opensips 1.6.4 to latest svn revision, i think.
Hi again,
i also saw that i compiled opensips with libxmlrpc-c3-dev and
libxmlrpc-c3 and i was warned somewhere that i'll compile it on my own
risk. Now i removed libxmlrpc-c3-dev and libxmlrpc-c3 and i compiled
with libxmlrpc-c++4-dev without warnings.
Let's see what we will get!
Thanks,
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS+= $(extra_defs) \
.
.
.
.
-DCHANGEABLE_DEBUG_LEVEL \
#-DF_MALLOC \
-DDBG_QM_MALLOC \
#-DDBG_F_MALLOC \
opensips will not be compiled with
to get only the memory dump without all runtime logs from mem
debugger.
Regards,
Bogdan
On 08/19/2011 01:13 PM, Dani Popa wrote:
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS+= $(extra_defs
to get only the memory dump without all runtime logs from mem
debugger.
Regards,
Bogdan
On 08/19/2011 01:13 PM, Dani Popa wrote:
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS+= $(extra_defs
the memory dump without all runtime logs from mem
debugger.
Regards,
Bogdan
On 08/19/2011 01:13 PM, Dani Popa wrote:
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS+= $(extra_defs) \
.
.
.
.
-DCHANGEABLE_DEBUG_LEVEL
support, set:
memlog=6
memdump=1
in order to get only the memory dump without all runtime logs from mem
debugger.
Regards,
Bogdan
On 08/19/2011 01:13 PM, Dani Popa wrote:
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS
that it RFC compliant.
Regards,
Bogdan
On 08/05/2011 07:30 PM, Dani Popa wrote:
Hi,
Ok, but also, registrar module support non case sensitive sip
username.
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Hi,
When using m_store($ru) the SIP messages sent back to sender have
default server_header and not the one i rewrite it.
Dani
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Hi,
Ok, but also, registrar module support non case sensitive sip username.
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On 8/5/11 11:40 AM, Vlad Paiu wrote:
Hello,
What you're asking for is against the RFC 3261 URI comparison rules,
which states that comparison of the userinfo part of the URI should be
done case
Hi all,
How can i remove all sip video body headers regardin video. Should i
remove any line from body after m=video, or how. Please give me a
hint, if you have.
Thanks,
Dani
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,
Why would you do that? If you don't want to allow video, you can
simply replace the video port in the m= line with 0.
Regards,
Razvan Crainea
OpenSIPS Developer
On 04.08.2011 16:58, Dani Popa wrote:
Hi all,
How can i remove all sip video body headers regardin video. Should i
remove any
, Razvan Crainea wrote:
Hi Dani,
It seems you are out of memory. What version of OpenSIPS are you using?
Regards,
Razvan Crainea
OpenSIPS Developer
On 04.08.2011 16:07, Dani Popa wrote:
Hi,
How can i solve this kind of problems ? Opensips doesn't crash, but
it not respond to any sip requests
Ok,
thanks for quick response.
Dani
On 08/04/11 18:26, Vlad Paiu wrote:
Hello,
Is it possible that you upgrade to 1.7 ? It is possible that this
issue was fixed in the latest OpenSIPS version.
If not, go to Makefile.defs, uncomment the line with
-DDBG_QM_MALLOC \
and comment the line
with 0.
[1] http://www.opensips.org/html/docs/modules/devel/textops.html#id293910
Regards,
Razvan Crainea
OpenSIPS Developer
On 04.08.2011 18:03, Dani Popa wrote:
Hi,
In fact, i have some problems with one of my pstn gw's that send 400
Incorrect content length, i think, because of too long sip
Hi,
it is somehow that username from sip uri to be non case sensitive when
we talk about presence and xcap storage? I mean, if userA add userB, in
his contact list, i need userA to be able to add userB even he add
him(type) as USERB.
Dani
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first
aaa_radius_auth and specific sql procedure in sql server.
the second
asterisk/freeswitch load balncing
Dani
On 07/12/11 17:06, duane.lar...@gmail.com wrote:
For your first question would this work?
http://www.ag-projects.com/projects-products-96/535-call-control
For your second
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Hi,
As far as i know it's hard to insert media from other sources in proxy
mode for situation like call hold or in call media insert. If you find a
solution, please let me know.
Dani
On 06/29/11 10:06, Barsan Liviu wrote:
Hi,
Yes, exactly. And obviously for this we want just one way
.Probably you need to explicitly
test with the UACs you want use.
Regards,
Bogdan
On 06/20/2011 07:08 PM, Dani Popa wrote:
Hi all,
It is viable solution to use 30(1|2|5) redirect for REGISTER sip
messages ?
Thanks,
Dani
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Hi all,
It is viable solution to use 30(1|2|5) redirect for REGISTER sip messages ?
Thanks,
Dani
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Hi all,
I looked on the internet for MOH with opensips as sip proxy(not b2b) and
other media servers (sems,asterisk,etc). The answers on internet was
that is not possible because SIP implementation and because
sems,asterisk are full implemented sip servers(invite from opensips to
media
Hi,
I thought so, but I needed confirmation.
Thanks Adrian,
Dani
On 06/16/11 15:46, Adrian Georgescu wrote:
You cannot do this reliably unless you insert a B2BUA in the call flow.
Adrian
On Jun 16, 2011, at 2:11 PM, Dani Popa wrote:
Hi all,
I looked on the internet for MOH
when start openxcap, it try to take schema from www.w3.org/2001/xml.xsd
and www.w3.org doesn't responde.
I changed schemaLocation in
/usr/local/pymodules/python2.6/xcap/appusage/xml-schemas/xcap-directory.xsd
and pointed to local file.
Dani
On 06/06/11 03:01, duane.lar...@gmail.com wrote:
Hi Liviu,
What kernel do you have on running media-relay machine ?
Thanks,
Dani
On 05/26/11 11:14, Barsan Liviu wrote:
Hi,
With the python-gnutls update to 1.2.1 the mediaproxy works fine.
A suggestion: would be welcome a minimal install guide for
Ubuntu/Debian, for example I spent several
root@test:/opensips_1_6# opensips -V
version: opensips 1.6.4-2-tls (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
Hi,
do you have news about this mediaproxy issues ?
Thanks,
Dani
On 05/03/11 11:52, Dani Popa wrote:
Ok,
Thanks,
Dani
On Tue, May 3, 2011 at 10:00 AM, Saúl Ibarra Corretgé
s...@ag-projects.com mailto:s...@ag-projects.com wrote:
On 05/02/2011 10:58 PM, Dani Popa wrote:
Hi
Ok,
Thanks,
Dani
On Tue, May 3, 2011 at 10:00 AM, Saúl Ibarra Corretgé
s...@ag-projects.comwrote:
On 05/02/2011 10:58 PM, Dani Popa wrote:
Hi,
Do you have any news with this issues ?
Unfortunately not. I didn't have time to go and fix this yet, sorry.
--
Saúl Ibarra Corretgé
AG
Hi,
Do you have any news with this issues ?
Thanks,
Dani
On Thu, Apr 21, 2011 at 3:31 PM, Dani Popa dani.p...@gmail.com wrote:
OK,
Thanks,
Dani
On 04/21/11 15:14, Saúl Ibarra Corretgé wrote:
I'm not talking abut binding ports for streams, i'm talking about stream
packets and bytes
hit.. but still not the same.
Can you please paste the output of 'bt'**
http://opensips.svn.sourceforge.net/viewvc/opensips/branches/1.6/modules/tm/uac.c?revision=7747view=markup
in gdb?
Regards,
--
Anca Vamanu
OpenSIPS Developer
On 04/20/2011 03:11 PM, Dani Popa wrote:
Hi,
I have
Hi,
yes, i was able to install it and run it, but i have some issues. I dont
have stream statistics: caller_bytes,callee_bytes,caller_packets and
callee_packets. Also, if i'm not sure if media timeout is working,
because i tried to simulate a hang call (in the middle of call, i
restart my
, Saúl Ibarra Corretgé wrote:
On 04/21/2011 12:44 PM, Dani Popa wrote:
Hi,
yes, i was able to install it and run it, but i have some issues. I dont
have stream statistics: caller_bytes,callee_bytes,caller_packets and
callee_packets. Also, if i'm not sure if media timeout is working,
because i
On 04/21/11 14:13, Saúl Ibarra Corretgé wrote:
On 04/21/2011 01:06 PM, Dani Popa wrote:
sure,
Apr 21 06:06:41 test media-relay[4903]:
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50012
Apr 21 06:06:41 test media-relay[4903]:
mediaproxy.mediacontrol.StreamListenerProtocol
OK,
Thanks,
Dani
On 04/21/11 15:14, Saúl Ibarra Corretgé wrote:
I'm not talking abut binding ports for streams, i'm talking about stream
packets and bytes info on telnet localhost 25060.
I meant the statisticas that get printed in syslog after the call is
closed.
[{from_tag: 4fc7812b,
Hi,
I have a problem using b2b_init_request with top hiding. When i
receive 200 ok for invite, opensips crash with
ERROR:nat_traversal:__dialog_confirmed: FAKED reply - exit.
In core dump this is where opensips crash:
#0 get_source_uri (dlg=0xb2b4bc84, type=8, _params=0xb70b3c20) at
wondering if is a network card driver issue or kernel
issue(if so, i'm dont know how to make troubleshooting, where should i
see the callee_bytes and caller_bytes in kernel stats).
Dani
On 04/18/11 10:43, Saúl Ibarra Corretgé wrote:
On 04/15/2011 02:42 PM, Dani Popa wrote:
Hi,
Mediaproxy
Hi,
Mediaproxy radius request does not populate Kbin and Kbout. Also i tried
to see sessions on port 25061 and also there callee_bytes and
caller_bytes are 0.
opensips:~# telnet localhost 25061
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
sessions
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