2013/8/14 Daniel-Constantin Mierla :
> Provided that you haven't restarted (or if you did it, reproduce it again),
> can you send:
>
> - output of kamctl ps
> - the record from the database that matches the same ruid
> - if you restarted, send the error logs, too
It seems it's a problem with the V
Yeah that was what I was thinking but wasn't confident it was the most
efficient or 'right' way of handling it. Sounds like it is my best option. I do
have registrants that can't accept the behavior so I will probably add a column
to the subscriber table or add a usr_preference to distinguish th
You can do it however you like, but a common way this is handled is to map the
DID to a registrar AOR using alias_db, then to revert the user part of the
request URI to the DID (having stored it in a variable prior to the alias_db
lookup).
However, not all registrants can accept an invite to a
Wondering how others handle DID routing?
I have seen some former posts that talk about using alaias_db, however the
request URI gets changed to the users SIP URI and the receiving device
(asterisk) cant distinguish the DID as a result.
Any thoughts are appreciated.
-dan
Definitely not adding it back anywhere. Is it possible that loose_route()
or t_relay() would add it back? These are the relevant blocks of code this
call is traversing.
xlog("L_ERR","We have a Proxy Route request, performing loose routing to
end point [$(hdr(Route))]");
remove_hf("P-Proxy-Route"
Hi
A client of ours needs to insert information into a table at some point into
the call. I am in betwwen these 2 options:
1- Using http_query function from the UTILS module to call a web service that
will insert the information
passed in the URL in to the table
2- or using the sql_query funct
Wondering how others handle DID routing?
I have seen some former posts that talk about using alaias_db, however the
request URI gets changed to the users SIP URI and the receiving device
(asterisk) cant distinguish the DID as a result.
Any thoughts are appreciated.
-dan
___
Hi Geoffrey,
it looks like you add the Route header somewhere else, since the
Route-Header has a different position before and after Kamailio. Maybe
a lookup (with Path?) or similar? If you want to remove a header,
which you added previously in the routing logic, then you will have to
apply the ch
Hello,
Just wanted to shoot a quick email to the user list and see if anyone
had any 'gotchas' to keep in mind when migrating from 3.2 to 3.3.5.
I know table structure has changed and will need to be updated, but are
there otherwise any syntax and or major configuration differences?
Thank you
Hello,
Kamailio SIP Server v4.0.3 stable release is out.
This is a maintenance release of the latest stable branch, 4.0, that
includes fixes since release of v4.0.0. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.0.0. Deplo
Design :
5 Buildings and Data Center
Extensions :
Building 1: 1000 to 1010
Building 2: 2000 to 2010
Building 3: 3000 to 3010
Building 4: 4000 to 4010
Building 5: 5000 to 5010
In Data Center I going to keep the both Asterisk and Kamailio
Outside Calls are going through Asterisk Server. In Aster
Hi,
I have Two Servers.
Asterisk PBX Server : 192.168.20.196
Kamailio Server : 192.168.20.208
No. of users created in Kamailio : 10
Using command : kamctl add 1000 12345
Here i am using Kamailio as SIP Server and Asterisk as media.
In Asterisk Server i am going to use Digium Telephonic Card. So
Hi again,
Please explain the whole situation and your desired setup. Once the whole
picture is clear then anyone will be able to guide you.
As I can imagine you might need to change the DBURL
mysql://XXX:XXX@DBSERVER/DBNAME
string in your kamailio.cfg
ensure that the Kamailio server access the DB
Dear Sammy,
Here if i use defualt kamailio.cfg, Its working fine. I have created user
and registered with Zoiper Soft phone.
The Issue i found when i change that kamailio.cfg . I copied and pasted in
the kamailio.cfg. Some changes i made like IP and Databaes in that cfg file.
Regards,
Nishar Ha
Hi Sammy,
Thanks for support. What are the changes do i have to make id DB here.
One more thing i have to ask you is how the Asterisk communicate with
Kamailio SIP users.
Do i have to create SIP trunk in Asterisk ?
Regards,
Nishar Hamsa.
On Thu, Aug 15, 2013 at 3:49 PM, SamyGo wrote:
> Hi Ni
Hi Nishar,
Once you've separate Asterisk and Kamailio installation you'll need to
modify the variables and DB parameters in kamailio.cfg. Simply copying the
configurations file may give you errors. Please see the log files "syslog"
or "messages" according to your OS and see why starting of kamaili
Hi,
I am Trying to make a Seperate SIP Server (Kamailio) and Asterisk PBX
Server.
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
But in Above link the issue is actually both are in same server. But I need
to make seperate server SIP Authentication and Asterisk.
I tri
I wish RTPproxy developers reply to this for further insightful details.
Read more about the draw backs of this on the same email as found out by
Andres and the next reply is somewhat acceptable but in case of early media
that is handled by RTPproxy when final 200 OK comes and RTPproxy handles it
Hello,
I will do the release of Kamailio v4.0.3 later today, any backports to
branch 4.0 should be done before 14:00GMT.
Cheers,
Daniel
On 8/12/13 9:07 AM, Daniel-Constantin Mierla wrote:
Hello,
I am planning releasing v4.0.3 on Wednesday, Aug 14. If anyone has
backports for branch 4.0, t
Hi Daniel, Sammy, thanks for reply
But as Andres pointed out
http://lists.sip-router.org/pipermail/sr-users/2008-March/062795.html
After the pre-filling state, rtpproxy can re-fill media address. I think it
means that rtpproxy does not care about IP in U and L (Update, Lookup)
commands, it only ca
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