Thanks.
Could somebody share a simple config file using dispatcher module.
Thanks
Gonzo
--- On Thu, 1/22/09, Iñaki Baz Castillo wrote:
From: Iñaki Baz Castillo
Subject: Re: [OpenSIPS-Users] Asteriak load balance
To: gonzov...@yahoo.com
Date: Thursday, January 22, 2009, 10:24 PM
El Jueves
El Jueves, 22 de Enero de 2009, Gonzalo Gonzalez escribió:
> What is the best module to use for load balance with 5 asterisk servers?
>
> user >> Opensips --->> Asterisk --->> PSTN
Try dispacher module.
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Dispatcher.
Gonzalo Gonzalez wrote:
> What is the best module to use for load balance with 5 asterisk servers?
>
> user >> Opensips --->> Asterisk --->> PSTN
>
>
>
>
>
> _
What is the best module to use for load balance with 5 asterisk servers?
user >> Opensips --->> Asterisk --->> PSTN
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Thanks a lot.
all is ok, and very good.
i will rewrite my script.
Iñaki Baz Castillo wrote:
>
> El Jueves, 22 de Enero de 2009, pjgonzalez escribió:
>> The user has 3 form to receive his inbount calls base on the following
>> preference:
>>
>> When the user is online:
>>
>> he can select on
Hi Joan,
See answers inline,
On Jan 22, 2009, at 6:42 PM, Joan wrote:
Good afternoon,
I am trying to install cdrtool to an ubuntu 8.04, there are some
problems I found so far:
.- I'd prefer to build myself the .deb, so everything is on the right
place whithout having to tweak everything by
El Jueves, 22 de Enero de 2009, pjgonzalez escribió:
> The user has 3 form to receive his inbount calls base on the following
> preference:
>
> When the user is online:
>
> he can select one of the following state:
>
> 1-Receive his calls on his softphone.
> 2-Forward his calls to another number on
The user has 3 form to receive his inbount calls base on the following
preference:
When the user is online:
he can select one of the following state:
1-Receive his calls on his softphone.
2-Forward his calls to another number on PSTN.
when user is offline:
sent his calls to voicemail.
Iñak
I am trying to translate the config file on the link from openser to opensips
1.4, but I am having an issue registering users, it looks like there is a loop,
but I don't know what am I doing wrong. I can make a call from a user account
to PSTN thru asterisk; I mean:
user >> opensips --->> a
El Jueves, 22 de Enero de 2009, pjgonzalez escribió:
> Ok,
>
> Here the code.
And which is the desired behaviour?
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Ok,
Here the code.
#
# #
#Authentication and Authorization against a radius server. #
# #
El Jueves, 22 de Enero de 2009, pjgonzalez escribió:
> I do ti but right now the call only ring on the Softphone, User B.
Sorry, but things are not so simple. You should provide a more detailed
description of the desired behaviour, current behaviour and code.
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Iñaki Baz Castillo
I do ti but right now the call only ring on the Softphone, User B.
thanks.
Iñaki Baz Castillo wrote:
>
> 2009/1/22 pjgonzalez :
>>
>> hi every body, im having problems when in my inbounts calls when i try to
>> forward the calls: for example.
>>
>> USER A(PSTN USER) calls User B (Behind Nat)
Good afternoon,
I am trying to install cdrtool to an ubuntu 8.04, there are some
problems I found so far:
.- I'd prefer to build myself the .deb, so everything is on the right
place whithout having to tweak everything by hand, but I cannot find
the Makefile, makedeb, whatever, any doc explaining
Not only the cleverest, I think it's the best!
I maintain the configuration for 12 SIP Proxies (using OpenSER) and I don't
need 12 different configuration files. A number of macros transform one m4
file into one of 12 different configuration files with the stroke of a short
command line.
I'd
Is there not a way to store in an AVP the gateway that was used in the call?
I have this need as well..-Brett
On Thu, Jan 22, 2009 at 9:44 AM, ibrahim tunali wrote:
> Hi Bogdan,
>
> I have already set attr avp like you sent. The problem occured when
> fill the attr field. If I leave blank attr f
Hi Bogdan,
I tried now and opensips started without the problem. But i did not
get the value after calling "do_routing()".
I see NULL value on output of xlog;
"PSTN Termination RURI:9XX USER:XXX DOMAIN:sip.x.com
CARRIER:"
if(!do_routing()){
xlog("L_WARN","No
Hi Ibrahim,
I found a bug related to what you described - I made a fix on SVN -
please update and test again.
Thanks and regards,
Bogdan
ibrahim tunali wrote:
> Hi Bogdan,
>
> I have already set attr avp like you sent. The problem occured when
> fill the attr field. If I leave blank attr field
Hi Bogdan,
I have already set attr avp like you sent. The problem occured when
fill the attr field. If I leave blank attr field in DB it starts but
when i set something like "gw1" it crashed.
My db row;
+--+--+---+---++---+-+
| gwid | type
Hi Ibrahim,
have you set the attr avp? Something like:
modparam("drouting", "attrs_avp", '$avp(s:dr_attrs)')
Then after do_routing() or use_next_gw(), do :
xlog("-gw attr is $avp(s:dr_attrs)\n");
the value of the attr is whatever you want - the module does not interpret it -
it is ju
Hi Matti,
no, there is no other solution - using M4 is the cleverest solution ;)
Regards,
Bogdan
Matti Zemack wrote:
>
> Hi again,
>
> First of all, Thanks for all help I received previously today.
> Fantastic and very quick response.
>
> So here is my second newbie-question: Is there any way t
Hello,
I'm playing with the new module drouting on svn trunk and i need to
get "which gateway is used on last request". I might be able to get it
with "attrs_avp" and "attrs" field on dr_gateways table, i guess. I
try some values to attrs but opensips crashed.
Could you give an example to use att
Hi again,
First of all, Thanks for all help I received previously today. Fantastic and
very quick response.
So here is my second newbie-question: Is there any way to include files at
startup? It would be nice to split a 2000+ lines config file into easier to
handle parts.
(After this morni
2009/1/22 pjgonzalez :
>
> hi every body, im having problems when in my inbounts calls when i try to
> forward the calls: for example.
>
> USER A(PSTN USER) calls User B (Behind Nat) then the calls is forwarded to
> User C (ON PSTN).
>
> the problem is that the calls is ringing on user B and User
Removes all of them. See:
http://www.opensips.org/html/docs/modules/1.4.x/textops.html#id271734
Regards,
Bogdan
Robert Borz wrote:
> Hi,
>
> so remove_hf("Proxy-Authorization") removes any Proxy-Authorization headers
> or just the first?
>
> What will happen if there are more than one proy auth.
hi every body, im having problems when in my inbounts calls when i try to
forward the calls: for example.
USER A(PSTN USER) calls User B (Behind Nat) then the calls is forwarded to
User C (ON PSTN).
the problem is that the calls is ringing on user B and User C so when i take
the call its drop
Hi,
so remove_hf("Proxy-Authorization") removes any Proxy-Authorization headers or
just the first?
What will happen if there are more than one proy auth. Header fields, will all
of them be removed? This would be a solution... but I don't want to close our
Sip proxy and only let outgoing and in
Hi Khan,
yes, in some points you are right - some things need to be changed in
OCP in order to make the integration with Opensips simpler. This will be
a priority for the next release of OCP.
Khan Friend wrote:
> I am in process of installation of OCP, I read the installation
> instructions.
Hi Robert,
I did read your email (even sent a reply ;))
consume_credentials() function removes only credentials that were
checked for authentication, so, in order to make it work, you have to
previously do authentication. The function works in this way because a
requests may contain credential
Hi Matteo,
Please update from SVN - I did so more logging. Also be sure you use
also the latest version of TM (update the whole SVN tree to the latest
trunk version).
Regards,
Bogdan
mmarzu...@interfree.it wrote:
> Hi.
> I deleted append_branch by the script, but the failure route does not wor
Hi,
Not sure if you figured this out, but this can only happen when
xcap-diff publishing is enabled.
So, unless you require this functionality disable it with
[OpenSER]
enable_publish_xcapdiff = no
> I am having the same issue with Kamailio. Did you ever figure out what this
> was happening?
>
Hi.
I deleted append_branch by the script, but the failure route does not work.
In the syslog there is:
Jan 22 11:52:45 opensips-lab /usr/local/sbin/opensips[3645]: Method is an INVITE
Jan 22 11:52:45 opensips-lab /usr/local/sbin/opensips[3645]: Call to PSTN
Jan 22 11:52:45 opensips-lab /usr/loca
Hi Bogdan,
thanks a lot for your response!
So... you're exactly saying where I run into. ;)
Did you read my last mails with the subject "consume_credentials doesn't work
with auth_radius module"?
Maybe you can help me just by telling me what's the way people do in production.
So... concerning
Hi Matti,
Matti Zemack wrote:
>
> Hi all,
>
> I’m new to Opensips and love the speed of the code. Even in the VmWare
> dev server I’m running on my (ancient) desktop machine!
>
> As a first attempt at my new job I thought I could try and rewrite the
> config-scripts used at my company so that on
2009/1/22 Matti Zemack :
> As a first attempt at my new job I thought I could try and rewrite the
> config-scripts used at my company so that one script easily could be used in
> both dev and all the different Live environments.
>
> Basically I could change a variable early in the opensips.cfg, an
Hi all,
I'm new to Opensips and love the speed of the code. Even in the VmWare
dev server I'm running on my (ancient) desktop machine!
As a first attempt at my new job I thought I could try and rewrite the
config-scripts used at my company so that one script easily could be
used in both
2009/1/21 Dan Pascu :
> On Tuesday 20 January 2009, Joan wrote:
>> I am having problems when building opensips 1.5 rev 5182 on a ubuntu
>> 8.04 On the 2009-01-15 build there was no problem on the same system,
>> something might have been broken afterwards.
>> I attach the build log
>
> You did a sv
2009/1/22 Pezhman Lali :
> Dear,
> I find nathelper and nat_traversal modules very similar.
> I think the only different between them is using rtpproxy.
> is any more different?
nathelper, until now, solves the NAT signalling problem. this means:
detects NAT based on request/response headers and
Dear,I find nathelper and nat_traversal modules very similar.
I think the only different between them is using rtpproxy.
is any more different?
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