Hi,
I am facing an Problem in Dialplan Module.
Scenario:
========
I am Dialling 00091xxxxxxxxxx number, I need to remove only the prefix 000,
then the call placement will be 91xxxxxxxxxx.
My dialplan rule:
==============
MariaDB [kamailio]> select * from dialplan;
+----+------+----+----------+-----------+-----------+-----------+----------+-------+
| id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp
| attrs |
+----+------+----+----------+-----------+-----------+-----------+----------+-------+
| 1 | 1 | 1 | 1 | ^000$ | 0 | ^000$ |
| |
+----+------+----+----------+-----------+-----------+-----------+----------+-------+
But the prefix 000 is not replaced. Please Guide me to Resolve this issue.
Do I need to change somet rules??
Additional info:
=============
But, With additional to the above dialplan data rule I used the below rule.
+----+------+----+----------+-----------+-----------+-----------+----------+-------+
| id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp
| attrs |
+----+------+----+----------+-----------+-----------+-----------+----------+-------+
| 1 | 1 | 1 | 1 | ^000$ | 0 | ^000$ |
| | 111
+----+------+----+----------+-----------+-----------+-----------+----------+-------+
The above rule replacing only the 000, If we Dial 000xxxx its omitting,
While Dialing 000 its replacing that with 111.
Thanks & Regards,
Logeshwaran
Hi all,
We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp
proxy. Kamailio have public IP, asterisk - no. All calls between clients now
going like that:
UserA ---sip--> Kamailio --> Asterisk --> UserB
\-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB
All clients of course from Internet and behind Nat. Main problem is amount
of traffic going through Kamailio and Asterisk. We need to pay for every
additional GB behind limit in tariff plan to hosting provider.
So we decided to try route all rtp traffic between users directly.
UserA ---sip--> Kamailio --> Asterisk --> UserB
\-rtp--> --> --> UserB
Is it possible at all? Maybe someone already did it .
--
BR, Alex
Hello,
I just pushed a bit of clean up to acc module, therefore be aware of
following changes:
1) the define conditions on SQL_ACC were removed -- this was enabled for
more than 10 years and only made the code look complex and hard to
follow up its logic.
2) the code related to DIAMETER accounting was relocated to acc_diameter
(new) module. It was a consistent size of code that was not enabled for
sooo... long. It is now a dedicated module, similar to acc_radius. The
diameter accounting code, even a new module now, is in the same stage,
compiling but not tested, in pair with auth_diameter module, it may
work, but very likely not.
In summary, what's important for those using the acc module -- it offers
the same functionality as it was enabled by default in the past 10 years
or more, only the unused code was relocated. It offers the functionality
of writing accounting records to syslog and sql databases.
The acc module is now slimmer, only with the code that it needs,
therefore easier to maintain and enhance for the future.
Hopefully, there was no side effect with this update -- anyhow, if you
find any issue, just open a bug report on github project.
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com
Hi all,
We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp proxy. Kamailio have public IP, asterisk - no. All calls between clients now going like that:
UserA ---sip--> Kamailio --> Asterisk --> UserB
\-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB
All clients of course from Internet and behind Nat. Main problem is amount of traffic going through Kamailio and Asterisk. We need to pay for every additional GB behind limit in tariff plan to hosting provider.
So we decided to try route all rtp traffic between users directly.
UserA ---sip--> Kamailio --> Asterisk --> UserB
\-rtp--> --> --> UserB
Is it possible at all? Maybe someone already did it ...
--
BR, Alex
Hello,
trying to make the git repository a bit slimmer and avoid unuseful
indexing of unused modules, I am considering to move misc/obsolete to
its own github repository (planned to be named kamailio-obsolete). The
initial idea was to keep there modules that were duplicated or no longer
maintained, just in case someone will come later and wants to revive it.
However, iirc, it is more than 4 years and nobody touched them, so very
likely it requires a bit of work to revive any of them, so a fresh
re-import will be the same from an external repository.
If anyone still wants to keep them in the main kamailio repository,
respond on the mailing list and present the benefits. If not, I will do
the operation in the next several days.
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com
Hello All,
I have the following SDP coming in from a carrier and would like to remove
the following line. Cant figure how to go about it.
line to remove : *a=cdsc: 1 image udptl t38*
sdp is as below
Content-Type: application/sdp.
Content-Length: 293.
.
v=0.
o=hiQ9200 5859020170708143552 1175715965 IN IP4 X.Y.Z.A.
s=Phone Call via hiQ9200 SIPCA.
c=IN IP4 X.Y.Z.A.
t=0 0.
m=audio 16648 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sqn: 0.
*a=cdsc: 1 image udptl t38.*
a=sendrecv.
a=ptime:20.
This reply comes from a upstream carrier in session progress and 200ok
causing one-way audio for me.
Thanking You,
Sunil More
Ph : 9503338275
Hi,
Is there any billing application or modules to use with kamailio , for
example if i want to check the credits of the user on every passing minute
, and when he's out of credits i stop the call.
Thanks & Kind Regards,
Logeshwaran G