Hi ALL,
I have a problem to configure LCR module to work properly (kamailio rel. 1.5)
The task in my case is to handle:
1) All calls coming with CLD number prefix 482223344 from my MGW
(77.77.77.77) towards kamilio and direct them to sip world
(88.88.88.88)
2) Calls with CLD starting with 48 coming from SIP world towards
kamailio and direct them to my MGWs. Additionaly i need to route it on
the basis of CLI number.
So MGWs <---> Kamailio <---> SIPProxies (88.88.88.88)
In db:
+----+-----------+----------------+--------+----------+
| id | prefix | from_uri | grp_id | priority |
+----+-----------+----------------+--------+----------+
| 1 | 48 | 48 | 1 | 1 |
| 2 | 48 | 49 | 2 | 1 |
| 3 | 482223344 | 77.77.77.77 | 3 | 1 |
+----+-----------+----------------+--------+----------+
gw,
+----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+
| id | gw_name | grp_id | ip_addr | hostname | port |
uri_scheme | transport | strip | tag | weight | ping | flags |
+----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+
| 1 | CLI48 | 1 | 77.77.77.70 | NULL | 5060 | NULL |
NULL | NULL | NULL | 150 | 0 | 0 |
| 2 | CLI48backup | 1 | 77.77.77.71 | NULL | 5060 |
NULL | NULL | NULL | NULL | 150 | 0 | 0 |
| 3 | CLI49 | 2 | 77.77.77.75 | NULL | 5060 | NULL |
NULL | NULL | NULL | 15 | 0 | 0 |
| 4 | CLI49backup | 2 | 77.77.77.76 | NULL | 5060 |
NULL | NULL | NULL | NULL | 15 | 0 | 0 |
| 5 | SIPWORLD | 3 | 88.88.88.88 | NULL | 5060 |
NULL | NULL | NULL | NULL | 250 | 0 | 0 |
+----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+
Unfortunately:
- call is originated on 77.77.77.77 with CLD 48222334455 number and
kamailio forward this call to grp_id=3 SIPWORLD which is okay. The
problem is that if the call fail, kamailio will try to use 77.77.77.70
and 77.77.77.71 from grp_id=1 which is wrong. I have no idea how to
provide a kind of huntstop in grp_id=3.
I've been trying with
+----+-----------+----------------+--------+----------+
| id | prefix | from_uri | grp_id | priority |
+----+-----------+----------------+--------+----------+
| 1 | 48 | 88.88.88.88 | 1 | 1 |
| 2 | 48 | 88.88.88.88 | 2 | 1 |
| 3 | 482223344 | 77.77.77.77 | 3 | 1 |
+----+-----------+----------------+--------+----------+
but in this case i am unable to provide kamailio with CLI number routing.
Here is a part of my config file...
if (!load_gws()) {
sl_send_reply("503", "Unable to load gateways");
exit;
}
if (!next_gw()) {
sl_send_reply("503", "Unable to find a
gateway");
exit;
}
route(1);
exit;
and failover for route(1)...
failure_route[11] {
# In case of failure, send it to an alternative route:
if (t_check_status("408|404|5[0-9][0-9]")) {
if (!next_gw()) {
t_reply("503", "Service not available, no more
gateways");
exit;
}
else {
t_on_failure("11");
t_relay();
}
exit;
}
}
Anybody could help my to get out of that?
Thx,
Maciej.
Hey,
I am using a Cisco WIP310 wifi phone. Seeing as wifi is very battery
demanding, the phone goes into a standby mode. When it's in the standby
mode, it takes a few seconds to come back on.
So I send an INVITE to the phone, statefully using TM, I send out a
CANCEL before the phone returns the "180 ringing" message.
Somehow the device is answering the CANCEL before the INVITE, so the
result is that it responds to the CANCEL with "481 Call Leg/Transaction
Does Not Exist.", after that it responds to the INVITE with a "180
Ringing", the phone than rings indefinitely because the CANCEL is not
sent out again as the transaction is completed from the 481 SIP message.
I had a look at the CANCEL, there is no Route: header or tag on the To,
so it looks like it is part of a new dialog ( I believe that's what
rfc3261 says.
As far as I can tell, my Kamailio is working properly. It is
retransmitting the messages at the proper intervals, and it is passing
along the messages as it receives them.
The trouble is that the device answers the initial CANCEL before it
answers any of the retransmitted INVITEs.
Is there something that I can do in Kamailio to resolve this issue ? Is
there an option that I can set that will cause Kamailio to relay the
CANCEL only to devices that have already returned a 100 Trying or 180
Trying ?
What information do you need to know about my config? What parts of the
SIP trace do you need ?
Thanks,
David
Hello all;
I need some Clarification about timestamp header in RFC 3261; i need to use
it to estimate the RTT value;
how it is work in this scenario?:
Invite
UA --------------------> Proxy1
| | Invite
| | ------------------------------
--------->Proxy2
| 100 trying
| |
| <---------------------------|
| invite
|
| |
| |
|
--------------------------------------->UA2
| | <-------100 trying-------------------
| |
|
| | <--------180
Ringing-------------- |
| | <---------------180 Ringing-------
| |
|
| |
|
|<---180 Ringing----------|
| |
UA send invite to proxy1 with timestamp=10 ; then the Proxy1 send the invite
message to proxy2 and reply with provisional response (100 trying) and with
timestamp=10 0.5;
When UA receive 100trying; UA will adapt its retransmition timer T1 to the
calculated value from timestamp;
then UA adapt T2 Timer value; is This true?
Here Can Proxy1 send invite message with timestamp header?;
for example timestamp=8 and when proxy2 recive invite request, it send it
to UA and he send 100 trying to proxy1 with
timestamp=8 0.4;
Please Help me;
Thanks;
Mustafa AL Rifaee
Hi all,
I'm in doubt of the means of setflag()'s parameter. I have looked for
it everywhere including internet and the ser source code, but no any
docs explained it in detail. can anybody tell me about it ?
--
With best regards
by sunone
Hello,
I have problems with rewriting $rU.
Kamailio 3.0.1 on debian squeeze x86_64
In config file I have:
$avp(s:dnumber) = $(ru{re.subst,/^sip:\*99\*(.*)@.*/\1/});
$rU=$avp(s:dnumber);
And log file shows following warnings:
Jul 5 17:22:30 voice /usr/sbin/kamailio[23641]: WARNING: <core>
[rvalue.c:987]: automatic string to int conversion for
"sip:1234567890@voice.serv.net" failed
Jul 5 17:22:30 voice /usr/sbin/kamailio[23641]: WARNING: <core>
[rvalue.c:1839]: rval expression conversion to int failed (630,22-630,23)
Could you please show any samples how to change $rU with regexp matched
part of $rU ?
BR,
Dmitri
Hi all,
I'm using SER, SEMS and RTPPROXY, they all work fine. When caller or callee
want to stop the dialog, one of them may send BYE to proxy, but the sip
proxy
doesn't receive it. so dows the other's BYE. In this case, how the proxy
verdict the
dialog is over and log it to mysql ?
--
With best regards
by zhoutj
Hi everyone,
I have a problem about Agents status.
Now, i want to develop ACD system but i don't know how to collect agent
status.
Please help me. Thanks in advance!
--
Best Regards,
Mr House Cricket
