Hi Bogdan,
Thanks for taking care of this bug. Now it works fine after few tests. Will
let you know if other issue show up after deep tests.
I assume that this update applies also for Load Balancing module when
setting destination set (dst_uri as IPv6 SIP URI), since the syntax is
similar.
Thank
Ok. For a second I thought my question was going to be one of those "it's
right here in the documentation", but I was positive that I couldn't find
anything about it.
Appreciate it.
Iñaki Baz Castillo wrote:
>
> El Martes, 15 de Septiembre de 2009, osiris123d escribió:
>> How do you use the
El Martes, 15 de Septiembre de 2009, osiris123d escribió:
> How do you use the email_address field in the Subscriber table? Looks like
> there is not a "opensipsctl fifo" command for adding or editing this field
> so that means the only way is to get into the database and add it manually
> which w
How do you use the email_address field in the Subscriber table? Looks like
there is not a "opensipsctl fifo" command for adding or editing this field
so that means the only way is to get into the database and add it manually
which would require you to restart openSIPS. I also can't seem to find
El Martes, 15 de Septiembre de 2009, michel freiha escribió:
> Dear All,
>
> I'm developing a Softphone from scratch and I'm using PJSIP sip
> stackI'm using OpenSIPS as proxy server with rtpproxy...
> My question is:
>
> Can rtpproxy handle the NAT traversal issue without intervention of a
Dear All,
I'm developing a Softphone from scratch and I'm using PJSIP sip stackI'm
using OpenSIPS as proxy server with rtpproxy...
My question is:
Can rtpproxy handle the NAT traversal issue without intervention of a
standalone STUN/TURN/ICE solution or I should have my own STUN/TURN/ICE
inst
Hi All,
I need some help with implementing multi-domain authorization for register
messages. I went through the documentation available and made changes to the
config file as is directed in the config file. I also added the new domain to
the domain table. Now when I try to register any user it
Following up... I did have a bad IP address in the server_address module
parameter. Fixing it did not help the situation.
With the top-hiding scenario called before auth, I notice a bunch of error
messages in the log at debug=3 including:
ERROR:tm:_reply_light: failed to generate 407 reply when
Hi Bogdan,
I tried it first after the auth for initial INVITEs. It seems to re-auth
(another 407 is sent to the UAC), but then forwards the INVITE upstream to
the UAS. In this case the call UAS sends 100, 180, then 200. Opensips
appears to forward only the 200 and not the 180. The UAC ignores
Hi Bogdan,
1.6 revision 6127 seems to handle the CANCELs just fine on my system.
- Jeff
On 9/15/09 9:08 AM, "Bogdan-Andrei Iancu" wrote:
> Hi Jeff,
>
> I just run a fast CANCEL test (made a call and cancelled it) and it
> looks ok (using the default opensips script).
>
> If the update did
Hi Juan,
There is somebody working on that, hopefully will be ready before the
svn freeze (on Thursaday).
Regards,
Bogdan
Juan Jose Lopez Juarez wrote:
> Hi.
>
> I'm trying to authenticate using dynamic bind to the ldap.
>
> I've seen that the feature it is been requested on:
>
> http://sourcef
Hi All,
in the NAT_TRAVERSAL module is present a |nat_keepalive()| function to
enable the keepalive mechanism Vs. an UA.
The question is... After that i call the nat_keepalive() function, how I
can stop the keepalive mechanism??
I see that when an UA send a De-Registration SIP Message, the
Hi Bogdan et all,
Any luck with this?
Thanks,
Sheran
On Tue, Sep 15, 2009 at 6:01 PM, Sheran Corera wrote:
> Hi,
>
> I am trying to use my opensips application to work as a router whereby
> extensions would be registered in my opensips box and the REGISTER
> request would be forwarded to my ISP
Hi All,
in the NAT_TRAVERSAL module is present a |nat_keepalive()| function to
enable the keepalive mechanism Vs. an UA.
The question is... After that i call the nat_keepalive() function, how I
can stop the keepalive mechanism??
I see that when an UA send a De-Registration SIP Message, the
Hi.
I'm trying to authenticate using dynamic bind to the ldap.
I've seen that the feature it is been requested on:
http://sourceforge.net/tracker/?func=detail&atid=1086413&aid=2822174&group_id=232389
But it doesn't seem to have any progress.
Any idea if this functionality is going to be implem
In that example, Asterisk has nothing to do with AUTH.
1) opensips get the REGISTER and sends back a challange
2) opensips get the REGISTER with credentials
3) opensips does the auth and forwards the REGISTER to *
Regards,
Bogdan
Uwe Kastens wrote:
> Hello Bogdan,
>
> Thank you for the example. I
Hello Bogdan,
Thank you for the example. In that case the asterisk have to accept the
registration without starting a auth itself?
BR
Uwe
Bogdan-Andrei Iancu schrieb:
> Hi Uwe,
>
> If you look at the default opensips script, you have a section (by
> default commented out) where the REGISTER
Hi Bogdan,
I had tried both. I'm compiling the latest 1.6; when that completes I'll
try again. Which way is recommended?
I have a functioning configuration otherwise. If I simply want topology
hiding between the UAC and UAS, it is as simple as invoking the b2bua at
some point and the rest will
Hi Jeff,
How are the auth and b2bua chained ? you first do auth and after that
invoke the b2bua ? or?
Regards,
Bogdan
Jeff Pyle wrote:
> Hello,
>
> What impact does authentication have on the b2bua modules? When I put
> b2b_init_request("top hiding") into my otherwise functioning script, very
Hi Uwe,
If you look at the default opensips script, you have a section (by
default commented out) where the REGISTER requests are authenticated and
if passing the auth doing save("location").
What you have to do is, after the REGISTER auth, instead of pushing the
REGISTER to the local registra
Thanks for the extra info.
Bogdan-Andrei Iancu wrote:
>
> Hi Duane,
>
> In opensips 1.5.x, the usage of append_branch is slightly different than
> before. Actually its usage from failure_route was aligned to the usage
> from request route.
>
> See: http://www.opensips.org/Resources/Docs
Hi Jonathan,
Jonathan González wrote:
> Hi there,
>
> I have been configuring OpenSIPS for a while the last 2 days and I
> have been able to configure a simple server that register clients
> against a Database. The problem I am facing is that when the call is
> established there's no Audio. I c
Hi Julien,
Julien Chavanton wrote:
> Thank you, I will look further after using group, I am also facig some
> requirements to append prefix, when a call comes in from a defined
> Gateway.
> Here is a recap, of what I understand after reading testing Dynamic
> Routing :
>
> When a call is comm
Hi Duane,
In opensips 1.5.x, the usage of append_branch is slightly different than
before. Actually its usage from failure_route was aligned to the usage
from request route.
See: http://www.opensips.org/Resources/DocsMigration14to15
http://www.opensips.org/Resources/DocsMigration14to15#toc4
Hi Alain,
I did a small fix on the SVN in regards to dealing with IPv6 IP
addresses in the dispatcher module. According to my tests, it should
work now.
So, please update from SVN and give it a try.
Thanks for report and regards,
Bogdan
alain bernard wrote:
> Folks,
>
> We are using dispatche
Hello João,
João Antunes wrote:
> Hi!
>
> I would like to know if it's possible to use the LDAP module along with
> the AUTH module to use LDAP for authenticating SIP users. Of course that
> an attribute with the MD5 hash is needed in the LDAP, but i already have
> that.
>
yes, you can have ei
2009/9/15 Bogdan-Andrei Iancu :
>> So in case time selection is null in botrules, priority mechanism
>> doesn't matter, right?
>> If so, I suggest to explain it in the documentation, as I couldn't
>> understand it after several reads :)
>>
>
> Let me tell you a joke:
> Q: "What do engineers and do
Hi Jeff,
I just run a fast CANCEL test (made a call and cancelled it) and it
looks ok (using the default opensips script).
If the update did not fix your issue, just let me know.
Regards,
Bogdan
Jeff Pyle wrote:
> Hi Bogdan,
>
>
> On 9/15/09 2:39 AM, "Bogdan-Andrei Iancu" wrote:
>
>
>> so
Iñaki Baz Castillo wrote:
> 2009/9/15 Bogdan-Andrei Iancu :
>
>> Hi Iñaki,
>>
>> priority applies only to rules that overlap - this can happens only when
>> time selection is used for the rules:
>>
>> Ex: RULE1: for prefix 1234, all the time, use GW1, prio =1
>> RULE2: for prefix 1234, d
2009/9/15 Bogdan-Andrei Iancu :
> The module does not support native weight, but you can simulate it (with
> order 1) :
> RULE1 -> GW: gw1,gw1,gw2,gw2,gw2
>
> this will give you a 40% 60% distribution for the first selected GW from
> the set.
ok, it's enough for my needs :)
--
Iñaki Baz Casti
2009/9/15 Bogdan-Andrei Iancu :
> Hi Iñaki,
>
> priority applies only to rules that overlap - this can happens only when
> time selection is used for the rules:
>
> Ex: RULE1: for prefix 1234, all the time, use GW1, prio =1
> RULE2: for prefix 1234, during weekend, use GW1, prio = 4
>
> Her
Hi Bogdan,
On 9/15/09 2:39 AM, "Bogdan-Andrei Iancu" wrote:
> so you actually do not get any CANCEL in your script? do you see it at
> network level getting to your server? also tried to place some xlog in
> the very beginning of the script?
Right. I had something like this at the very begin
Hi,
I am trying to use my opensips application to work as a router whereby
extensions would be registered in my opensips box and the REGISTER
request would be forwarded to my ISP. If the ISP is down it would
connect to a secondary ISP. So essentially what I am trying to do is
route the REGISTER me
Iñaki Baz Castillo wrote:
> 2009/9/14 Iñaki Baz Castillo :
>
>> 2009/9/14 Iñaki Baz Castillo :
>>
>>> In my example call the rule 1 is choosen (since it has highest prioriry).
>>> Gateways 1 and 2 fail (reply 503 code) and there is no more failover, this
>>> is:
>>> servers 3 and 4 are not
Hi Iñaki,
Iñaki Baz Castillo wrote:
> 2009/9/14 Iñaki Baz Castillo :
>
>> In my example call the rule 1 is choosen (since it has highest prioriry).
>> Gateways 1 and 2 fail (reply 503 code) and there is no more failover, this
>> is:
>> servers 3 and 4 are not tryed, is it the expected behaviou
Hi Iñaki,
priority applies only to rules that overlap - this can happens only when
time selection is used for the rules:
Ex: RULE1: for prefix 1234, all the time, use GW1, prio =1
RULE2: for prefix 1234, during weekend, use GW1, prio = 4
Here the rules will overlap during the weekend -
2009/9/14 Iñaki Baz Castillo :
> 2009/9/14 Iñaki Baz Castillo :
>> In my example call the rule 1 is choosen (since it has highest prioriry).
>> Gateways 1 and 2 fail (reply 503 code) and there is no more failover, this
>> is:
>> servers 3 and 4 are not tryed, is it the expected behaviour?
>
> Afte
Italo Dacosta wrote:
>> Ok - that's the first problem. Number of children == number of
>> processes. You cannot handle more than 4 messages at one time right
>> now.
>>
>
> I forgot to mention that I tested with several number of children
> already. Surprisingly, increasing the number of proce
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