p://www.opensips-solutions.com
On 05/21/2014 07:23 PM, Jorge Ortea wrote:
Hi all,
when I execute 'opensipsctl fifo dlg_list' shows this:
dialog:: hash=206:862937179
state:: 2
u
Hi all,
when I execute 'opensipsctl fifo dlg_list' shows this:
dialog:: hash=206:862937179
state:: 2
user_flags:: 0
timestart:: 0
timeout:: 0
callid:: 786344563@A.B.C.D
from_uri:: sip:x@MyDomain
to_uri:: sip:y@MyDomain
caller_tag:: 356538242
ca
, Jorge Ortea ha scritto:
Hi all,
I have a scenario with OpenSIPS 1.8 and Asterisks 1.4. Proxy
SIP has two ways to manage a call, the first is B2BUA and second
is be relay between UAC and Asterisk.
I have a problem, when
through asterisk to get transcoding.
Il 27/02/2014 16.51, Jorge Ortea ha scritto:
Hi all,
I have a scenario with OpenSIPS 1.8 and Asterisks 1.4. Proxy
SIP has two ways to manage a call, the first is B2BUA and second
is be
Hi all,
I have a scenario with OpenSIPS 1.8 and Asterisks 1.4. Proxy SIP has two ways
to manage a call, the first is B2BUA and second is be relay between UAC and
Asterisk.
I have a problem, when OpenSIPS works as B2BUA and both UAC can't negotiate
codec then this call failed. I would like re
Hi all,
I have a problem with dialog module, sometimes I see records in dialog table
that I think are wrong, those records have start_time > timeout and are
permanent. The others records are correct and of course start_time < timeout.
I too have noticed that the state of wrong records is 3 and th
;
á á á á }
You will need to somehow make this work for your setup but hopefully this shows
you what you are looking for.
On Mon, Dec 16, 2013 at 7:31 PM, Jeff Pyle wrote:
Jorge,
This is a function of Asterisk, not Opensips. áThis page may help you:
ááhttp://www.voztovoice.org/?q=node
Hi all,
Suppose a platform with OpenSIPS and several Asterisk behind. A new call in a
Asterisk that send to Opensips to route to uac1. The uac1 is ringing, it is
sending 180 Ringing, then from other uac wants CallPickup this call, this
feature is dialed but when the Invite reach to OpenSIPS,,, H
Any idea?
Thanks.Regards.
From: dar...@hotmail.com
To: users@lists.opensips.org
Date: Mon, 25 Nov 2013 15:59:46 +0100
Subject: [OpenSIPS-Users] Parallel Forking for MESSAGING
Hi all,
I use opensips-1.6.4-2 and when there are several
register from only one SIP account, INVITE works and all U
Hi all,
I use opensips-1.6.4-2 and when there are several
register from only one SIP account, INVITE works and all UAC registered
receive it. But on same scenario, MESSAGE is received only for one.
Is there any configuration needed?
Thanks.
Regards. __
: s...@ag-projects.com
> Date: Thu, 24 Jan 2013 10:18:27 +0100
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] mediaproxy behavior
>
>
> On Jan 23, 2013, at 12:58 PM, Jorge Ortea wrote:
>
> >
> > Hi all,
> >
> > Saul, this is right,
>
> On Jan 18, 2013, at 9:54 AM, Jorge Ortea wrote:
>
> > Hi all,
> >
> > Stefano, i ask me if could do it any other way.
> >
> > For example, what happen if I set PAT redirect with iptables from external
> > firewall to OpenSIPS private IP.
> &
, 16 Jan 2013 10:35:50 +0100
From: stefano.pis...@omnianet.it
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] mediaproxy behavior
Use rtpproxy :-)
s
Il 16/01/2013 09:53, Jorge Ortea ha
scritto:
Hi all
Hi all,
I use OpenSIPS + Mediaproxy and several asterisk behind. I have the next
problem: I would like SIP Proxy with a public IP and the asterisks in private
network, but it isn't possible because mediaproxy only can forward RTP on the
same network interface, this forces that each asterisk mus
Developer
http://www.opensips-solutions.com
On 12/06/2012 02:11 PM, Jorge Ortea wrote:
Hi all,
I have a voip platform with OpenSIPS and several asterisk
behind. I'm using Load-Balancer module, but I don't seem very
reliable,
Hi all,
I have a voip platform with OpenSIPS and several asterisk behind. I'm using
Load-Balancer module, but I don't seem very reliable, sometimes when take place
very much CPS, load_balance function returns null.
I noticed that Load-Balancer module has beta status, perhaps that is why this
m
Hi all,
I have a Voip platform with a Opensips 1.6.4 and several asterisks. To manage
transfers I search dialogs in mediagateways direct on mysql DB, but I am aware
to be able make it through dialog module and the profiles. This could have
important performance differences ?
Thanks.
Regards.
ease check SIP trace and confirm or
attach that here for our review.
For a workaround you can modify dialog table and allow callee_contract field to
be NULL. However i am not sure if this won't break anything else.
Thank you.
On Mon, Nov 12, 2012 at 1:52 PM, Jorge Ortea wrote:
Hel
;s telling you it can't add another dialog
to db. I am almost certain that's what's happening.
Regards,
Ali Pey
On Thu, Nov 8, 2012 at 4:22 AM, Jorge Ortea
wrote:
t's what's happening.
Regards,
Ali Pey
On Thu, Nov 8, 2012 at 4:22 AM, Jorge Ortea
wrote:
Hi all,
I am getting the
Hi all,
I am getting the following errors:
Nov 7 11:47:19 hgt-tero45 /usr/local/opensips/sbin/opensips[12582]:
ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db
Nov 7 11:47:19 hgt-tero45 /usr/local/opensips/sbin/opensips[12548]:
ERROR:dialog:update_dialog_dbinfo: could n
?
Very Thanks.
Regards.
> From: s...@ag-projects.com
> Date: Thu, 4 Oct 2012 10:43:46 +0200
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] media-relay problem
>
> Hi,
>
> On Oct 4, 2012, at 10:26 AM, Jorge Ortea wrote:
>
> > Hi all,
> >
Hi all,
I just installed Mediaproxy in a virtual machine and when I start media-relay,
it returns error:
(I had previously installed mediaproxy correctly)
[root@vir-voip01 lib]# /usr/bin/media-relay --no-fork
Starting MediaProxy Relay 2.4.4
Set resource limit for maximum open file descriptors to
/2012 11:58 AM, Jorge Ortea wrote:
Hi Bogdan,
I have 0.9.8b version of libssl.
- First core dump file:
(gdb) bt
#0 0x00aed746 in OPENSSL_cleanse () from /lib/libcrypto.so.6
#1 0x00d2f4b0 in ks
core file with:
#0 0x00d26c01 in SSL_CTX_get_timeout () from /lib/libssl.so.6
REgards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 08/30/2012 11:58 AM, Jorge Ortea wrote:
Hi
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 08/29/2012 09:48 PM, Jorge Ortea wrote:
Hello Bogdan,
I just see that there were 2 coredump files, this is the other
Hi Jorge,
I see a core file was generated - please extract the backtrace
from there with gdb and post it here.
Thanks and regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 08/29/2012 02:05 PM, Jorge O
nSIPS Founder and Developer
http://www.opensips-solutions.com
On 08/29/2012 02:05 PM, Jorge Ortea wrote:
Hi all,
I use OpenSIPS 1.6.4-2-tls from one year ago and it never did
anything like this. Simply auto-killed, I only have thi
Hi all,
I use OpenSIPS 1.6.4-2-tls from one year ago and it never did anything like
this. Simply auto-killed, I only have this log:
.
.
.
Aug 28 03:49:04 muc-vfk21 /usr/local/opensips/sbin/opensips[3801]:
INFO:core:tls_accept: client did not present a certificate
Aug 28 03:49:12 muc-vfk21 /usr
which
it's not using.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06/06/2012 11:38 AM, Jorge Ortea wrote:
Hi,
maybe I don't explained it well. I m
Hi,
maybe I don't explained it well. I meant add a column on modules tables at
database to use personal.
Thanks.
Regards.
From: dar...@hotmail.com
To: users@lists.opensips.org
Date: Tue, 5 Jun 2012 13:51:55 +0200
Subject: [OpenSIPS-Users] add fields
Hi,
Could have i any problem to add fi
Hi,
Could have i any problem to add fields on modules tables? to get a
'Description' field, for example ?
Thanks.
Regards.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mail
Hi,
I have a performance's question.
What is more efficient? dp_translate function of dialplan module or a read
query of MySQL ?
I need found a ip adress of list from this way or another.
Thanks.
Regards.
___
existing one), but it fails in timeout as you cannot
open a TCP conn somewhere behind a NAT.
Regards,
Bogdan
On 04/04/2012 06:06 PM, Jorge Ortea wrote:
Hi Bogdan,
Is correct, Z.Z.Z.Z:5062 is a public adress behind a
:
http://www.opensips.org/Resources/DocsCoreFcn18#toc78
Regards,
Bogdan
On 04/04/2012 05:16 PM, Jorge Ortea wrote:
Hi Bogdan,
Exactly, is ready, OpenSIPS try to reach to destination but now
the account 2105 haven
ed at the end - this failure resulted in the 477
reply.
Check the opensips logs to see error when processing the BYE.
Regards,
Bogdan
On 04/04/2012 11:42 AM, Jorge Ortea wrote:
Hi,
I have the follow VoIP pla
Hi,
I have the follow VoIP platform; OpenSIPS 1.6.4.2-tls + Mediaproxy 2.0 + a
pair of Asterisks 1.4 (behind SER)
It works fine but sometimes a sip message enters on a loop. Asterisk sends 5
sip messages at every turn
My logs in OpenSIPS:
Apr 4 10:14:17 alpha02 /usr/local/sbin/opensips[29
Hi,
I have OpenSIPS 1.6.4-tls with tls configuration in my opensips.cfg, it is
working correctly with my Cisco SIP phones.
/* uncomment the following lines to enable TLS support (default off) */
disable_tls = no
listen = tls:192.168.1.1:1234
tls_verify_server = 1
tls_verify_client = 0
t
Hi,
I have OpenSIPS 1.6.4-tls with tls configuration in my opensips.cfg, it is
working correctly with my Cisco SIP phones.
/* uncomment the following lines to enable TLS support (default off) */
disable_tls = no
listen = tls:192.168.1.1:1234
tls_verify_server = 1
tls_verify_client = 0
Hi,
I have OpenSIPS 1.6.4-tls with tls configuration in my opensips.cfg, it is
working correctly with my Cisco SIP phones.
/* uncomment the following lines to enable TLS support (default off) */
disable_tls = no
listen = tls:192.168.1.1:1234
tls_verify_server = 1
tls_verify_client = 0
tls_req
Hi,
when i restart opensips returns this:
Mar 6 15:20:02 ghss02 /usr/local/opensips/sbin/opensips[29437]:
NOTICE:signaling:mod_init: initializing module ...
Mar 6 15:20:02 ghss02 /usr/local/opensips/sbin/opensips[29437]:
INFO:sl:mod_init: Initializing StateLess engine
Mar 6 15:20:02 ghss0
Hi,
my question is how manage the keep-alives messages ??
I understand that the keep-alives only serve to keep open NAT.
Now, to avoid manage this messages in my routing logic, i am discarding this
messages, is this correct? or i should respond.
I too have detected that when my cisco telepho
answers first will get the call ) , or do
serial forking ( try each phone one at a time until you either get a
success or a failure on all phones ).
Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 02/08/2012 10:32 AM, Jorge Ortea wrote
Hi,
is possible to have several sip phones configurated with the same SIP account ?
If i want to have a SIP account configurated in two phones, for ex. my SIP
phone in the office and my smartphone, I need two accounts SIP perforce and
configure my PBX to manage multiDial to both accounts or ex
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