Hi, Jayesh!
Unfortunately you can't do this in OpenSIPS. The 'var' pseudo-variable
only accepts a constant name, like 'media'. There are some variables,
like AVP, that can also receive a variable name ( ex: $avp($avp(name))
), but note they can only receive a pseudo-variable, not a format ( li
Hi, Peter!
Using the 'append_hf' function, the header is inserted in the Request
message, not in the Reply. If you want to append a header to the Reply
message, you have to use the 'append_to_reply' function [1].
[1] http://www.opensips.org/html/docs/modules/1.8.x/sipmsgops#id249124
Regards,
Hi, Jayesh!
No, it won't affect performance in any way, as each opensipsctl tool is
a simple script that will communicate with different OpenSIPS instances.
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 05/29/2012 02:40 PM, Jayesh Nambiar wrote:
Hi Raz
Hello, Jayesh!
Opensipsctl uses the opensipsctlrc configuration file in order to detect
which FIFO file it should use. If the OSIPS_FIFO parameter is not set in
opensipsctlrc, it uses the default value of /tmp/opensips_fifo.
In order to run different instances of opensipsctl, you first have to
Hi, Arjun!
Are OpenSIPS and RTPProxy on the same machine? Also you should double
check your firewall isn't blocking the 7890 port on localhost.
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 05/24/2012 11:01 AM, Arjun Shankar K S wrote:
Hi All,
Greetin
“–n tcp:1.1.1.1:” where 1.1.1.1 – ip of opensips
serever.
Am I right?
From:
users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf
Of Razvan C
Hi, Denis!
The socket can be UNIX or TCP according to the documentation[1].
[1] http://www.opensips.org/html/docs/modules/devel/rtpproxy.html#id250454
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
Hi, Mariana!
Can you check if the variables are properly flushed into the DB after 200OK?
Also, if you only want to check the direction of a sequential request,
you could use directly the $DLG_dir [1] pseudovariable from the dialog
module.
[1] http://www.opensips.org/html/docs/modules/1.8.x/d
Hi, Jacek!
OpenSIPS 1.8 version is still in beta, but the stable release is coming
soon. You will receive an update regarding this during the next week.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 05/08/2012 11:42 AM, Jacek Konieczny wrote:
Hello,
Op
et
the acc variables, and my understanding is that I cannot access dlg
variables until I call loose_route (or match_dialog).
Is their a way to do both? Would it hurt to call loose_route() twice?
Regards,
Ryan
On Tue, May 1, 2012 at 1:14 AM, Razvan Crainea wrote:
Hi, Ryan!
Have you tried to set the
Hi, Ryan!
Have you tried to set the avp values before 'loose_route' call?
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/28/2012 06:28 PM, Ryan Bullock wrote:
Without any success I have been trying to get some values accounted
using the 'db_extra_bye'
Hi, Vladimir!
Please provide us a complete SIP trace of the call. You can send me on
private if you want.
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/20/2012 05:05 PM, Vladimir Vakulov wrote:
Good afternoon.
For several days I try to construct a
Hello, Will!
You are right, the ratelimit module is the one you should choose to
limit CPS. But unfortunately I think you misunderstood how it works. The
parameters of the rl_check functions are the followings:
key - the entity you want to limit the CPS. For example this can be an
IP, if you
Hello!
The log you provided with debug 5 isn't complete, it doesn't contain the
response received by OpenSIPS from RTPProxy. You should check RTPProxy's
logs in order to see why it sends a command answer with port 0.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions
Hi, Brett!
You are right, the order of the flatstore fields is exactly the one you
mentioned. More exactly, they are organized as follows:
method
from_tag
to_tag
callid
sip_code
sip_reason
time when the insert query is made
db_extra fields
db_extra_bye fileds, if you are using this feature
d
Hello, Moe!
I have just committed a fix on svn. Can you please update your sources
and try again? Let me know if the problem is not solved.
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/14/2012 12:57 PM, Moe Navid wrote:
Regarding the second one, I h
86_64/linux))
Content-Length: 0
It would seem that either $(ct.fields(uri) or $ru would grab the
Contact header string:
Contact: ;q=0.99
but xlog copies the received URI instead of the enum_pv_query results.
Is there something in addition to $ru that I need to look for to
capture the URI
Hi, Apenk!
Try to increase the debugging level to 6 and check if there is any
useful information there.
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/09/2012 12:13 PM, apenk wrote:
Hi,
I'm trying to make opensips work with nathelper and rtpproxy.
I
Hello, Andrew!
You cannot access the raw NAPTR response from the script, you can only
see the R-URI modified according to the ENUM regexp response. Therefore,
you can search for the rn param in the R-URI ($ru pseudovariable).
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
http://www.opensips
nsips-solutions.com
On 04/05/2012 11:06 AM, Marcello Lupo wrote:
Hi,
effectively i was using it after the loose_route().
I will try to do it after the loose_route().
Why it have to be done in this way? Just for information.
Thank you for the answer.
Regards
Marcello
On Apr 5, 2012, at 9:36 AM, Razvan Cr
Hi, Marcello!
The block used to handle the ACK timeout is executed before loose_route
or after? It should be before.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 04/04/2012 11:59 PM, Marcello Lupo wrote:
Hi,
I'm using opensips 1.6.4 with dialog support
Developer
http://www.opensips-solutions.com
On 03/29/2012 11:34 AM, Miha wrote:
On 3/29/2012 10:26 AM, Razvan Crainea wrote:
Hi, Miha!
If you take a closer look at your logs, you will find this error:
"No database module found"
This means that you haven't loaded any database m
Hi, Miha!
If you take a closer look at your logs, you will find this error:
"No database module found"
This means that you haven't loaded any database module (i.e. db_mysql)
in your configuration script.
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 03/
Sorry, small typo. The proper code should be:
if (is_method("INVITE") && !has_totag())
setflag(2);
Regards!
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 03/23/2012 06:20 PM, Razvan Crainea wrote:
Hello, Sebastien!
The problem is that
s thread because I am facing the same issue as John and I
am not sure I understood your answer.
What should be done in order to make the cdr_flag work with opensips
1.7 ?
Regards,
Sebastien
Le 20/09/2011 18:23, Razvan Crainea a écrit :
Hi John,
The problem is here:
if (method==
Hi, Alex!
Yes, the t_replicate function hasn't changed. Your problem is that you
are not passing properly the first parameter of the t_replicate()
function. It should be:
t_replicate("$du", "0x4");
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 02/29/20
Hi, Ryan!
The CacheDB feature was only added for the dialog profiles. This allows
you to do load balancing scenarios through multiple OpenSIPS instances
that share the same profiles.
Regarding the persistence of the dialog data, you will also have to use
a database backend. But if we are talki
Hello, all!
OpenSIPS has been enhanced with a new function, sipmsg_validate, that
verifies if the received SIP request or reply is compliant with the
RFC3261[1]. The function can be found in the new sipmsgops module[2],
along with all the SIP aware messages from the textops module[3]. In
cons
Hi, Steven!
The dialog module only stores the TO and FROM URIs, not the display
names. Also, the uac_replace_from function keeps track only of the URIs.
Therefore, the display names are ignored and they will not be
automatically restored on any sequential requests.
Regards,
--
Răzvan Craine
Hello, Anil!
You should replace the routeid column from your your dr_rules table with
an empty value. Otherwise that routeid will be called every time the
"do_routing" function is called, that's why the loop appears.
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
http://www.opensips-solution
Hello all!
The dialog profiles have been enhanced with support for the CacheDB
Key-Value Interface[1] in order to provide support for distributed
profiles. This new feature will allow script writers to share dialog
profiles between multiple instances of OpenSIPS that share the same
NoSQL back
Hi, Jan!
I am glad you are interested in developing a small feature in OpenSIPS,
but you should have opened this discussion on the devel list[1].
Your assumption is right, the rtpproxy_offer/answer functions don't
accept variables as arguments.
In order to change that, you will have to modify t
Hi, Taisto!
It is a fix that was not ported to 1.7 until now. I have just backported it.
Please update your svn sources and let me know if this doens't solve
your problem.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://opensips-solutions.org/
On 02/03/2012 05:24 PM, Taisto Qvist wrot
Hi, Matt!
So for different hashed strings like:
DBG:dispatcher:ds_hash_pvar: Hashing "061002"!
you always get the same output:
DBG:dispatcher:ds_select_dst: alg hash [1], id [1]
Also, are you assigning different weights for any of the Asterisk?
Regards,
--
Ra(zvan Crainea
OpenSIPS Develope
Hi, Matt!
Are you sure you are not using the same hashing value all the time?
If yes, can you increase your debugging level to 6 and provide more
information? I would be looking in the opensips log for lines that
contain the following strings: "ds_hash_pvar: Hashing " and
"ds_select_dst: alg h
Hello, Matt!
This part was not changed in OpenSIPS. The flags you are passing ("IE")
are simply forwarded to RTPProxy, and he is the one who decides what IP
to return to OpenSIPS. So I don't see why changing the rtpproxy.c
behaves differently.
You could try to capture the traffic sent by OpenS
Hi, Matt!
The fix was only reported by Ovidiu and I was the one that fixed it in
OpenSIPS's revision 8616 on trunk and 8639 on 1.7.1. This doesn't have
anything to do with Kamailio. Can you please check if you have these
changes? Because otherwise I don't think it will work.
Regards,
Ra(zvan
Hi, Matt!
What version of OpenSIPS are you using? There was a bug reported by
Ovidiu and fixed in revision 8639.
Both scenarios you presented can be implemented using the rtpproxy module.
Regards,
Ra(zvan
On 01/22/2012 07:15 PM, Matt Hamilton wrote:
Hi,
I can't seem to force a new IP to cha
Hi, Jayesh!
First of all, there are not so many changes between the 1.7.1 and trunk
for the accounting module. Only this new parameter was added as far as I
know, and it should be pretty stable. If you do want to use the module
from trunk, you can simply copy it in your OpenSIPS 1.7.1 folder,
Hello, Jayesh!
You can not achieve this scenario in OpenSIPS 1.7.1, but you can wit the
acc module from trunk version. There you have a new parameter,
db_extra_bye[1], that can evaluate the pseudo variables after the BYE
message is received.
An easier way to implement this, is to initialize
Hi, Plamen!
I have backported to 1.7.1 a fix from trunk for the uac module. Please
update your svn sources and try again. Let me know if you still have
problems.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
On 01/04/2012 04:17 PM, Razvan Crainea wrote:
Hello!
Are you sure you only call
Hello!
Are you sure you only call uac_replace_from/to only from the branch
route, not from main route? Also, what version of OpenSIPS are you using?
Regards!
--
Răzvan Crainea
OpenSIPS Developer
On 01/04/2012 01:34 PM, goup2010 wrote:
Every use of uac_replace_from from branch_route appen
Hi, Plamen!
I have just committed a fix for this. Please update your sources and try
again.
Regards,
Ra(zvan Crainea
On 01/03/2012 06:47 PM, goup2010 wrote:
I use svnrevision: 2:8632M. When try to compile revision 8639 get
follow error:
flatstore.c: In function 'flat_db_insert':
flatstore.c
;2","foobar")", I get a line telling
me resource "foobar" is not found in group 1???
So in a failure_route load_balance does care about the resource and
does NOT care about the group number, is this not strange?
Steven
*From:*users-boun...@lists.opensips.org
[mailto:
Hi, Faisal!
Only the load_balancer table is requiered. Your problem is that you
didn't specify the db_url parameter for the load_balancer module. See
http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id249159
for more details.
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
Hi, Steven!
This is not wrong, this is the desired behaviour in order to use the
failover features.
I see that you have only one destination in group 2. Will this be the
final scenario? If yes, then you shouldn't use load balance, but a
simple RURI rewriting.
However, if you want to use multip
Hello,
Starting from revision 8637, the exec_* functions are allowed to be
called from timer route. The commit was done on both trunk and 1.7.1.
--
Răzvan Crainea
OpenSIPS Developer
On 12/28/2011 03:45 PM, Bogdan-Andrei Iancu wrote:
Hi guys,
I just opened a ticket, to allow these functions
Hi, Nick!
In the traces you sent, it seems that the callee's RTP traffic doesn't
reach the media relay server. This is most likely because the RTPProxy
is behind NAT. In this case you have to double check two things:
* if the SDP is properly changed for the INVITEs and 200OKs
* if the media po
Hi, Darren!
There is indeed a bug there, but the patch I've attached should fix it.
Can you please apply it and test again? Let me know if this solves your
issue.
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
On 12/16/2011 06:47 AM, Darren DeLitizia wrote:
Is this valid naming for avp's
FROM hdr
What kind of debug would you need? I may be able to replace it on a
smaller scale.
Thanks!
-Brett
On Thu, Dec 15, 2011 at 4:25 AM, Razvan Crainea
mailto:razvancrai...@opensips.org>> wrote:
Hi, Brett!
Are you using uac_replace_from function with dialog support or the
Hello,
Jeff, I was wrong, this feature is not available in 1.6, only in 1.7 .
Anyway, according to Brett's traces it seems to be a problem with the
function and I would really appreciate if he could help me debugging this.
Regards,
Ra(zvan
On 12/15/2011 07:27 PM, Jeff Pyle wrote:
Brett,
Is
Hi, Brett!
Are you using uac_replace_from function with dialog support or the old
approach with route parameters? Would it be possible to increase
OpenSIPS debugging level and paste some extra information?
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
On 12/15/2011 12:41 AM, Brett Nemerof
Hello, Nick!
You should check in your OpenSIPS configuration file if you somehow call
the unforce_rtp_proxy() command twice. Or perhaps you actually receive
two BYE messages from the IP phone 2. Perhaps extra debugging for
RTPProxy will help you understand what really happens.
Regards,
--
R
k you very much for all the efforts. Really appreciate it. Need to get
hold of my DB guy for the stupid mistake !!
--- Jayesh
On Wed, Dec 7, 2011 at 8:13 PM, Razvan Crainea
wrote:
Hi, Jayesh!
Can you check in your mysql database if the vars column from the dialog
table is declared as TEXT or BL
Hi, Jayesh!
Can you check in your mysql database if the vars column from the dialog
table is declared as TEXT or BLOB and not CHAR? If not, please change
your column into BLOB:
ALTER TABLE dialog CHANGE vars vars BLOB;
Regards,
--
Răzvan Crainea
OpenSIPS Developer
On 12/07/2011 02:27 PM,
Hi, Jayesh!
I need the logs after opensips restarts. The result might also be pretty
large.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
On 12/07/2011 02:14 PM, Jayesh Nambiar wrote:
Hi Razwan,
I have applied the patch and made it working. Do you still only need
the logs after opensips s
bin here:
http://pastebin.com/gx0ZxFLb
Let me know if there is anything more to test.
--- Jayesh
On Wed, Dec 7, 2011 at 3:36 PM, Razvan Crainea
mailto:razvancrai...@opensips.org>>
wrote:
Sorr, I forgot to attach it. Here it is.
Regards,
-
ter opensips
shutdown in the paste-bin here:
http://pastebin.com/gx0ZxFLb
Let me know if there is anything more to test.
--- Jayesh
On Wed, Dec 7, 2011 at 3:36 PM, Razvan Crainea
mailto:razvancrai...@opensips.org>> wrote:
Sorr, I forgot to attach it. Here it is.
Regards,
--
Sorr, I forgot to attach it. Here it is.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
On 12/07/2011 11:59 AM, Jayesh Nambiar wrote:
Hi Razvan,
I don't see the patch attached !!
--- Jayesh
On Wed, Dec 7, 2011 at 2:57 PM, Razvan Crainea
mailto:razvancrai...@opensips.org>> wrot
DBG:db_mysql:db_mysql_do_prepared_query: set values for the statement run
I hope this will help you figure out something more relevant.
--- Jayesh
On Tue, Dec 6, 2011 at 7:56 PM, Razvan Crainea
mailto:razvancrai...@opensips.org>> wrote:
Hi, Jayesh!
Can you post your dlg_db_
Dec 6 17:24:13 dev /usr/local/sbin/opensips[1958]:
DBG:dialog:write_pair: Dumping var name: value: <1003>
Dec 6 17:24:13 dev /usr/local/sbin/opensips[1958]:
DBG:dialog:write_pair: Dumping var name: value: <1007>
Let me know if this helps.
Thanks,
--- Jayesh
On Tue, Dec 6, 201
route(default_relay);
exit;
}
}
But it is just that the cdr insert does not take place after the
restart !!
My debug level was at 3. Do you want more detailed logs to check??
Thanks,
--- Jayesh
On Mon, Dec 5, 2011 at 7:12 PM, Razvan Crainea
mailto:razvancrai...@opensips.org>
Hi Jayesh,
Can you please check if the dialogs are loaded back after a restart.
Also, do you see any errors in your log?
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
On 12/05/2011 02:31 PM, Bogdan-Andrei Iancu wrote:
Hi Jayesh,
Razvan is checking the code for you and running some tests.
ress[1] , " ) said : ", data
On 30 November 2011 13:14, Razvan Crainea
mailto:razvancrai...@opensips.org>>
wrote:
Hi, Pete!
In order to use OpenSIPS Event Interface, you need to create
an external application that listens for an event. This
Hi, Pete!
In order to use OpenSIPS Event Interface, you need to create an external
application that listens for an event. This application should create
the UNIX socket used for communication, and then subscribe to OpenSIPS
for a certain event, using the event_subscribe MI command. When the
e
Hi, Pete!
Yes, indeed the RabbitMQ currently uses the same value for the exchange
name and routing key. Also only the '/' vhost can be used for the
moment, as it is hardcoded in the code. But this module is still under
development and any new ideas are welcome.
Regards,
--
Ra(zvan Crainea
O
:rtpproxy:force_rtp_proxy_body: no available proxies
However, when I run netstat none of these ports seems to be used...
Best regards,
Sebastien
Le 08/11/2011 13:48, Razvan Crainea a écrit :
Hi Sebastien,
Taking a look into RTPProxy's code, I see that the error 10 is
returned when it can
Hi Sebastien,
Taking a look into RTPProxy's code, I see that the error 10 is returned
when it can't create a listener. This happens when RTPProxy can't create
or bind a socket, or doesn't have enough ports allocated. My guess is
that in your case it can't bind a socket on the interface specifi
roxy set 1 ?
- what does the node->rn_recheck_ticks parameter means ?
- do you think the bug is in my opensips.cfg or in the rtpproxy module ?
Thanks a lot for your help.
Best regards,
Sebastien
Le 08/11/2011 12:10, Razvan Crainea a écrit :
Hi Sebastien,
No, if rtpp_test returns 1, it
x27;m confused...
Best regards,
Sebastien
Le 04/11/2011 15:23, Razvan Crainea a écrit :
Hi Sebastien,
I will try to replicate this scenario and see if I am getting the
same behaviour. I will get back to you later.
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
On 11/04/2011 04:20 PM, Seba
forgot :
# - rtpproxy params -
modparam("rtpproxy", "rtpproxy_sock", "1 == udp:localhost:12221")
modparam("rtpproxy", "rtpproxy_sock", "2 == udp:localhost:12222")
Regards,
Sebastien
Le 04/11/2011 11:44, Razvan Crainea a éc
Hi Sebastien,
Are you sure that when you declare the RTPProxy sets you allocate them
the set identifiers (1 and 2)? Can you send us the rtpproxy_sock
parameters declaration?
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
On 11/04/2011 12:27 PM, Sebastien CRUAUX wrote:
Hi,
I am currently
Hi all,
The new version of "ratelimit" module on 1.8.0 (trunk) comes with a set
of major enhancements:
1) Dynamic and flexible pipes
The new version of ratelimit module allows you to create an unlimited
number of pipes and even more, in a dynamic way at runtime (versus the
old version wher
Hi Nick,
What version of OpenSIPS are you using?
Since 1.6.3, the force_rtp_proxy function does exist anymore. It has
been replaced with rtpproxy_offer/answer functions.
Regards,
--
Răzvan Crainea
OpenSIPS Developer
On 13.10.2011 11:30, Nick wrote:
Hello
When I start opensips. It display
Hi all,
OpenSIPS has been enhanced with a new module called **event_rabbitmq**.
This module allows the proxy to feed a RabbitMQ server with useful
runtime events and information about OpenSIPS.
More exactly, **event_rabbitmq** is designed as a transport module for
the new OpenSIPS Event Inte
Hi Akib,
In order to get the call duration, you will have to use the new cdr
accounting, found in both OpenSIPS 1.7 and trunk.
To do that, you will only have to set the 'cdr_flag' on all the initial
Invites and the duration will be automatically inserted into the
database. Therefore the extra
Hi Federico,
Make sure that you set the corresponding flag for the SIP Requests you
want to account.
Also, I saw you have two extra columns configured in the cfg. Have you
added those columns into the database too? Because otherwise the mysql
returns an error (can't find the 'from_uri' and 'to
Hi Alex,
According to your scenario, there is a late negotiation (SDP is
advertised in 200OK and ACK). I see there is 'rtpproxy_offer' called for
a 200OK, but I can't see any 'rtpproxy_answer' for an ACK message.
Perhaps that's why the SDP in the ACK remains the same and I don't think
it is a
Hi John,
The problem is here:
if (method=="INVITE" || method=="BYE") {
# Write CDR records to the database
setflag(2);
}
For any sequential request, the CDR engine ignores the flag (2 in your
case). But the standard accounting will still
Hi Alexander,
When I try to open the diagram you sent, I receive the following error:
Error 500
java.io.IOException: Operation not permitted
Message: Operation not permitted
RequestURI: /file/n6775967/sip_flow.jpg
Server: n2.nabble.com
Please provide a valid image so we can investigate your sc
Hi,
It seems there is a parsing error. Can you please try again with the
patch I attached?
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
On 05.09.2011 12:50, vivid333 wrote:
opensips.cfg
modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7890")
# I tried localhost, publi
ed another method of rewriting the phone number, the $tU does not
allow R/W so I need another function which will allow me to edit the $tU
field.
Mike
On 1/09/11 4:57 PM, Razvan Crainea wrote:
Hi Mike,
I think that the dialplan module [1] fits in your scenario.
You can use regular expressions
Hi,
If you receive this error only for the "alias_db" module, most likely
this module is not installed.
You should check if this module is included in the "exclude_modules"
list (in the OpenSIPS Makefile, line ~53).
Regards,
--
Ra(zvan Crainea
OpenSIPS Developer
On 31.08.2011 15:30, Brett
Hi Mike,
I think that the dialplan module [1] fits in your scenario.
You can use regular expressions to match each state code and append the
corresponding area code.
These rules can be added in your database using OpenSIPS CP.
[1] http://www.opensips.org/html/docs/modules/devel/dialplan.html
Hi Saul,
I've just committed a fix for those syntax errors. Please try again.
Regards,
Razvan Crainea
OpenSIPS Developer
On 24.08.2011 23:06, Saúl Ibarra Corretgé wrote:
Hi,
I never tried the OpenSIPS console before, but when I wanted to give it a try:
root@nwo:~# osipsconsole
s
What message is that error generated for? a Register or a 200 OK? Can
you send a trace taken on the proxy machine?
Regards,
Razvan Crainea
OpenSIPS Developer
On 24.08.2011 14:13, isshed wrote:
Yes I took a trace. The message is reaching on the proxy.
Also the proxy is running and listening
Hello Isshed,
Are you sure this is not a network problem ? I don't think the REGISTER
message can't event reach OpenSIPS proxy. Try to take a trace on the
proxy machine to see if everything is received.
Regards,
Razvan Crainea
OpenSIPS Developer
On 24.08.2011 14:04, isshed wrot
happy to include it in our next releases.
Regards,
Razvan Crainea
OpenSIPS Developer
On 23.08.2011 11:49, Saúl Ibarra Corretgé wrote:
Hi Razvan,
On Aug 23, 2011, at 10:33 AM, Razvan Crainea wrote:
Hi Saul,
On 22.08.2011 20:03, Saúl Ibarra Corretgé wrote:
Hi,
I've been testing the
__
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[3] http://sourceforge.net/projects/opensips/files/OpenSIPS/1.7.0/
Regards,
Razvan Crainea
OpenSIPS Developer
___
Use
Hi Ryan,
I've just committed a fix for this issue on trunk. Please update your
sources and try again. Let us know if your problem was solved.
Regards
Razvan Crainea
OpenSIPS Developer
On 11.08.2011 21:34, Ryan Revels wrote:
Bogdan,
This is a brand new install so no tables exist. He
eply is received for the initial INVITE
onreply_route[2] {
$dlg_val(cisco) = "true";
set_dlg_flag("3");
...
}
Please let us know if you find any problems.
Regards,
Razvan Crainea
OpenSIPS Developer
On 08.08.2011 23:58, Bobby Smith wrote:
Hi Razvan,
So in tryin
eply is received for the initial INVITE
onreply_route[2] {
$dlg_val(cisco) = "true";
set_dlg_flag("3");
...
}
Please let us know if you find any problems.
Regards,
Razvan Crainea
OpenSIPS Developer
___
Users ma
e the dialog is matched.
Please try again and let us know your results.
[1] http://www.opensips.org/html/docs/modules/devel/dialog.html#id293854
[2] http://www.opensips.org/html/docs/modules/devel/rr.html#loose-route-id
Regards,
Razvan Crainea
OpenSIPS Developer
On 05.08.2011 07:14, Bobby Smith
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 packet.
So, because it is pstn, i want to remove video capabi
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.
Aug 3 07:
Hi Dani,
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
Hi Chris,
I guess the "incorrect port 0" errors appear in the OpenSIPS logs. Do
you also receive errors in the RTPProxy logs?
OpenSIPS doesn't delete the dialogs because, perhaps, the RTPProxy
timeout notification is never received.
Regards,
Razvan Crainea
OpenSIPS
Hi Chris,
What svn revision are you using? I remember you reported this issue, but
I think this should be fixed in the latest svn.
Regards,
Razvan Crainea
OpenSIPS Developer
On 01.08.2011 10:51, Chris Martineau wrote:
Hi,
I have raised this issue before but now I have had it cause the
Hi Mickael,
Try to start a ngrep on the proxy. You will see exactly who sends the
second BYE.
Does this scenario happen for every call?
Regards,
Razvan Crainea
OpenSIPS Developer
On 29.07.2011 16:33, mick...@winlux.fr wrote:
Hi Razvan,
I crated:
/#The local route is executed
,
Razvan Crainea
OpenSIPS Developer
On 29.07.2011 15:20, mick...@winlux.fr wrote:
Thanks Razvan,
my version is 1.6.4-2-tls (svn revision: 2:8151)
but this function will work in ? knowing that my OpenSIPS generate
this second BYE.
Ex:
- UAC --> BYE --> Opensips --> BYE --> provider
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