Can someone please help me ind the right direction, i need to implement a
system to rewrite the destination URI from a list of possible destinations.
eg
call from 20304050
if(method == "INVITE") {
"check where 20304050 is"
then return and replace 20304050 with something like
+4520
Simply replace the $ru ($ru = NEWURI) and the call goes to the right
destination
If the remapping is fixed you and use ALIAS. for example.
Il 10/03/2014 08.11, Mike Claudi Pedersen ha scritto:
Can someone please help me ind the right direction, i need to
implement a system to rewrite the desti
this makes no sense to me, im not sure alias is what i wanna use. i just
want some method to redirect calls to another server, using a list of known
users and locations
2014-03-10 9:32 GMT+01:00 Stefano Pisani :
> Simply replace the $ru ($ru = NEWURI) and the call goes to the right
> destinatio
im trying to establish connection between 2 phones
user1: 43384001
user2: 43384002
i have user 2 added to usrloc with the cmd:
opensipsctl ul add 43384003 sip:+1043384003*2...@voip.local.com
which is an asterisk server that handles calls in a specific way.
but when i try to call i get "too many
im trying to establish connection between 2 phones
user1: 43384001
user2: 43384002
i have user 2 added to usrloc with the cmd:
opensipsctl ul add 43384003 sip:+1043384003*2...@voip.local.com
which is an asterisk server that handles calls in a specific way.
but when i try to call i get "too many
If you added an usrloc entry, then you will need to use some kind of
locaton lookup and then route accordingly.
- Laszlo
2014-03-10 11:03 GMT+01:00 Mike Claudi Pedersen :
> im trying to establish connection between 2 phones
> user1: 43384001
> user2: 43384002
>
> i have user 2 added to usrloc w
It seems an addressing issue.
Could you post your opensips.cfg?
Are you able to log the sip session using ngrep? Post it.
Il 10/03/2014 11.03, Mike Claudi Pedersen ha scritto:
im trying to establish connection between 2 phones
user1: 43384001
user2: 43384002
i have user 2 added to usrloc with t
could you please give an example?
2014-03-10 11:08 GMT+01:00 Laszlo :
> If you added an usrloc entry, then you will need to use some kind of
> locaton lookup and then route accordingly.
>
> - Laszlo
>
>
> 2014-03-10 11:03 GMT+01:00 Mike Claudi Pedersen >:
>
>> im trying to establish connection
Hello Khaled,
You could build a regexp dialplan rule such as "^[0-9]{7}$" in order to
strictly match, for example, a 7-digit number.
Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 03/07/2014 05:04 PM, M.Khaled W Chehab wrote:
Hi,
how to match a dialplan
the log are only spammed with subscriptions
2014-03-10 11:09 GMT+01:00 Stefano Pisani :
> It seems an addressing issue.
> Could you post your opensips.cfg?
> Are you able to log the sip session using ngrep? Post it.
>
> Il 10/03/2014 11.03, Mike Claudi Pedersen ha scritto:
>
> im trying to est
Hello Mike,
Instead of simply relaying initial INVITEs, you should also add some
registrar logic. You need at least something like this:
if (!lookup("location")) {
t_reply("404", "Not Found");
exit;
}
A successful registrar *lookup* will also rewrite the R-URI of your
request
if i just add a user and a usrloc to my usrloc database
eg.
opensipsctl add user1 1234
opensipsctl ul add user1 us...@voipserver.dk
how do i in the simplest manner rewrite the call?
so when i call user1 from user2, it rewrites user1 to us...@voipserver.dk
if i could just understand how this part
Hello,
For my first post to this mailing list, I just want to know what I need to
implement to use a LDAP remote directory with my OpenSIPS server ?
After some researches, I founded that I need to use LDAP module and Auth Module
right ?
Thanks for your help.
cyrill gremaud
___
nevermind, i found out that i forgot to add the user... only added the user
location. now it works as intended
2014-03-10 12:08 GMT+01:00 Mike Claudi Pedersen :
> if i just add a user and a usrloc to my usrloc database
> eg.
>
> opensipsctl add user1 1234
> opensipsctl ul add user1 us...@voipser
Hello, all!
We are proud to announce you that during this week, we'll be releasing
three new OpenSIPS minor versions: 1.8.4, 1.9.2 and 1.10.1. All of them
will include the latest fixes we've done during the last 6 months. If
you have any other outstanding issues, please report them asap on the
How to authenticate customers without username and password using tech
prefix on a AAA server and opensips?
Alcindo Schleder
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Users@lists.opensips.org
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Hi guys,
I updated my OpenSIPS server from version 1.6.4 to version 1.10. Nearly
everything is working. I only have some problems with the presence
stuff: IMHO the structure of XML presence object is not correct.
Working presence body (OpenSIPS 1.6.4):
--- SNIP ---
xmlns:dm="urn:ietf:params:
Hi,
On Mar 10, 2014, at 8:15 PM, Martin Stock wrote:
> Hi guys,
>
> I updated my OpenSIPS server from version 1.6.4 to version 1.10. Nearly
> everything is working. I only have some problems with the presence stuff:
> IMHO the structure of XML presence object is not correct.
>
> Working prese
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