Ok, I modified the supported codec on the SIP client and the call went through
successfully. However, I got one way audio. UA2(callee) with public IP can
hear, but UA1(caller) with private IP cannot.
The INVITE msg arrived on UA2 is like with the private IP in SDP. I think this
causes the
On Mon, Aug 31, 2009 at 8:10 AM, Leon Lileon...@aarnet.edu.au wrote:
Ok, I modified the supported codec on the SIP client and the call went
through successfully. However, I got one way audio. UA2(callee) with public
IP can hear, but UA1(caller) with private IP cannot.
The INVITE msg arrived
Hi Ashwini,
have you compiled the acc module with RADIUS support ? you need to go in
modules/acc/Makefile and enable the RADIUS support - then recompile and
reinstall.
Regards,
Bogdan
ASHWINI NAIDU wrote:
Hi,
I have upgraded the opensips from 1.5 to 1.5.2 . when i use
acc_rad_request
you should modify like this : modparam(usrloc, db_url,
postgres:)
2009-08-28
刘博
发件人： adolphus (via Nabble)
发送时间： 2009-08-26 11:04:29
收件人： Aris
抄送：
主题： [OpenSIPS-Users] 1.5.2 OpenSIPS and postgresql
Hi,
I've been wrestling with postgres and OpenSIPs and have been
Hello Bogdan,
Thank you for your help.
My problem of Dialog module is solved. The dialog were not inserted into
the DB. But now its inserted. I have set the flag before the INVITE and
after forwardign the call to proxy. Now its working fine.
One more question I would like to ask is that, for
I'm afraid you have a mixture of code there.
The makefile you post belong to trunk version but you mention using
1.5..so, what opensips version are you actually using?
Regards,
Bogdan
ASHWINI NAIDU wrote:
Hi BogDan,
This is what i have in Make file of acc module
*# $Id: Makefile 5958
Hi Khan,
You can start with 2 simple checks:
1) be sure your force_rtp_proxy() functions are triggred both for
request and reply - put some xlog to see if you get there in the script
2) check the messages with SDP (on the outgoing part) if they have the
rtpproxy indication in SDP
Regards,
Hi Jan,
the packaging for deb was not yet updated (the packages are in process
of being generated)hopefully they will be updated this week.
Regards,
Bogdan
Jan D. wrote:
Today I tried to update my Debian AMD64 (test) system to 1.5.3 without any
luck. Seems like a little oeps or I am
Hi Uwe,
something like, if you already registered 4 times, the fifth one to be
forwarded to Asterisk instead of local registering ?
Regards,
Bogdan
Uwe Kastens wrote:
Hi list,
I would like to implement the following. A user should be able to
register n times with opensips. If one or more
Hi Jayesh,
AFAIK, nat_traversal module does not use a DB, but a file named
keepalive_state that will be created in the working dir of opensips -
this file creates the conflict.
As the file name is configurable (module parameter) you can renamed it
for the second instance of opensips (to
Hi Sebastian,
As the error says, the drouting module accepts only AVPs as input
variable (and not script variables as in your script). try:
$avp(i:10) = 100;
do_routing($avp(i:10));
Regards,
Bogdan
Sebastian Sastre wrote:
Bogdan,
Thanks for your reply. I can't seem to pass the variable
Compile the aaa_radius module also and configure the aaa_url param in
the acc module (see
http://www.opensips.org/html/docs/modules/devel/acc.html#id271395)
Regards,
Bogdan
ASHWINI NAIDU wrote:
Hi BogDan,
Yes i am using the information in trunk/opensips_head. So how can
i enable
[OpenSIPS-Users] Got No suitable relay found from mediaproxy-2.3.6
Dear All,
I got 'No suitable relay found' from media-dispatcher when I'm calling
out from opensips to a Cisco 5300.
I'm using NAT'ed ip and Cisco 5300 is using a public ip.
I notice there is a post
Hi,
Even after doing that i am getting *missing loadmodule * error
for*acc_rad_request
*
On Mon, Aug 31, 2009 at 1:48 PM, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Compile the aaa_radius module also and configure the aaa_url param in
the acc module (see
Hi Bogdan,
If the Account make the 1st sucessfull registration it should be relayed
to an asterisk. The 2nd, 3rd ... registration could be relayed. If all
registrations have expired or a deleted the registration on the asterisk
should be removed as well.
Br
uwe
Bogdan-Andrei Iancu schrieb:
Hi Ashwini,
ASHWINI NAIDU wrote:
Hi Bogdan,
I guess there are some documentation errors for few modules. The
*acc_rad_request* exported function is replaced by* acc_aaa_request*
in *acc module*.
Indeed, you are right - the RAD token was replace with AAA - I will
update the 1.6
Hi,
I am looking for the ability to send media backward before a call is
established, allowing the remote side to send some DTMF before a call is
established. For that purpose, is it possible to delay 200 OK on
OpenSIPS for a pre-configured interval?
So far I have considered the b2bua module,
Hi Bogdan,
Sorry, trying again:
Users should register to opensips. If one successfull registration at
opensips for one account is present, opensips should register with
asterisk (plus would be, if the data could be different). Background is,
that I would like to use the parallel forking with
Debian packages are available for download:
http://opensips.org/pub/opensips/latest/packages/debian/
or for apt :
http://www.opensips.org/apt/
Packages were generated for unstable and etch.
Regards,
Bogdan
http://www.opensips.org/apt/
Bogdan-Andrei Iancu wrote:
Hi Andrew,
There are ways to delay the 200 OK, but it is an ugly approach - simply
use sleep() function in onreply route, but I strongly recommend not to
do it as it has a huge penalty in performance.
What I rather suggest is to use the new B2B support in opensips - you
just have to create
Hi Uwe,
So first registrions stays with OpenSIPsm while the next ones are
forwarded to Asterisk, right?
For the one registered with Asterisk, should opensips stay in the middle
(as mid-registrar) or Asterisk will directly register the client contact?
Regards,
Bogdan
Uwe Kastens wrote:
Hi
2009/8/31 Bogdan-Andrei Iancu bog...@voice-system.ro:
Debian packages are available for download:
http://opensips.org/pub/opensips/latest/packages/debian/
or for apt :
http://www.opensips.org/apt/
Packages were generated for unstable and etch.
Hi Bogdan, current Debian
Hi Bogdan,
So first registrions stays with OpenSIPsm while the next ones are
forwarded to Asterisk, right?
Nearly. IF the user is registered with opensips, opensips should send a
registration to asterisk (and keep at alive). If more registrations are
done with the same user account,
Iñaki Baz Castillo wrote:
2009/8/31 Bogdan-Andrei Iancu bog...@voice-system.ro:
Debian packages are available for download:
http://opensips.org/pub/opensips/latest/packages/debian/
or for apt :
http://www.opensips.org/apt/
Packages were generated for unstable and etch.
Hi Urmi,
urmi lakkad wrote:
Hello Bogdan,
Thank you for your help.
My problem of Dialog module is solved. The dialog were not inserted
into the DB. But now its inserted. I have set the flag before the
INVITE and after forwardign the call to proxy. Now its working fine.
I'm glad you
We are in the process of upgrading from OpenSER 1.2.3 to Opensips 1.5.
After migrating over our current script and tweaking a few settings (a few
module names were different etc) It looks like things are pretty much
resolved, the only errors we are receiving now when we attempt to start
opensips
Hi,
I am using uac_replace_from(display,uri) with the restore mode set to
auto.
Everything is fine with the SIP URI, but the display name is not set
correctly in the ACK and the BYE.
1. In the ACK, the display name is NOT changed as it was in the INVITE.
2. When the BYE is sent by the
Hi!
Looks like OSP module doesn't compile for me on 1.5.x (1.4.5 compiles without
problem - on the same machine). I tried the trunk with latest patches from Mr.
Di-Shi Sun, without success.
I have osptoolkit 3.4.2 installed. Please advise.
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align
I think the part is in line 14 till 19:
Via: SIP/2.0/UDP
115.43.124.42:23102:1;branch=z9hG4bK1505e2940e3e30927.2fc2d921ec4e7e4c7;rport
After 23102 comes :1
I think the :1 is the bogus part in this trace.
I also have other traces where i see ip:portbranche instead of
ip:port;branche. I can
Hello All,I first time tried to use the mediaproxy with media-relay on a
remote host and everything worked well as expected. But after a span of an
hour or so the connection to that relay was lost and I could see following
messages in the logs:
media-dispatcher[23102]: error: Error processing
hi all
please could anyone help me configure a redirect server using OpenSIPS?
i need to route registrations from server1.com, that is the opensips
redirect server to server2.com, that is any registrare server
thanks for any help
__ Information from ESET NOD32 Antivirus, version of
Hello community ,
Here's a SOAP ( as in Simple Object Access Protocol ) user provisioning
interface for openser/kamailio/opensips :
http://openzips.poly.ro/
Its called SOAPPI and its written in php.
You can use it for ya know... provisioning... you can build a website and use
SOAP to
Hello,
I am able to register with opensips or relay register to my asterisk
server. What I want is:
a) insert/delete the UL of opensips each time a
registration/deregistration takes place at the asterisk (Should work, if
I use on reply route and insert information with exec or sql).
b)
Hi,
Recompile opensips-1.5.2 with radius support. I guess that will solve the
problem.
In /opensips-x.x.x/modules/acc/Makefile
uncomment the line
*ENABLE_RADIUS_ACC=true*
Then recompile the opensips.
On Mon, Aug 31, 2009 at 7:19 PM, Justin Moore jmo...@sagisys.com wrote:
We are in
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