Hi there,
After reading about Kamailio, i like this project but i am having trouble
finding the wikis or step by step guide to successfully integrate asterisk
with kamailio for simple use case where phones register to the asterisk pbx
via kamailio and make/receive calls to ITSP, there is no proper
documentation on 2 products integration, really appreciated your
guidance and if somebody could provide me the step by step guide.
Looking forward to the guides.
I have 2 function first
If(is_user_in(“Request-URI”, ”ld”)){
.....
}
So i dont know what to do , second
If(get_user_group(“Request-URI”, ”$avp(i:10)”){
...
}
I put users in 2 groups but they could call each others but i dont want that .
Any help please ,
Thank you.
Hi Duarte,
I'm not sure if your situation is more complicated than mine. I have two
dispatcher nodes in the HA setup, with real IPs say IPa, IPb. They share a
floating/virtual IP, let's call it VIP.
Both nodes listen to the VIP (not its real IP), and use the VIP as the
dispatcher's socket. In this case the master node which has the VIP can
always send OPTIONS pings fine.
My problem was with the backup node, which doesn't have the VIP so fails to
send OPTIONS pings. Joel Serrano gave a good suggestion: both nodes start
up with polling disabled, and only when a node turns master it enables the
polling. This has solved my problem.
Cheers,
Yufei
Hello everybody,
I'm trying to route incoming calls to kamailio with specific subscriber
but I can't make it working.
The complete scenario is:
SoftSwith -> kamailio 5.3 -> zoiper
What i'm trying to do is to receive a call into the zoiper when I dial
specific DID which is already coming into the kamailio. It goes until
the routing from kamilio to the subscriber account but it keeps in
INVITE command, no response from the subscriber. Any help would be great!
This is what I've been trying but no luck yet:
if($rdir(name)=="downstream") {
xlog("DID request was incoming call, called number:
$rU caller_id: $fU and source_ip $si \n");
$sht(rtpProxy=>mip_$ci)=1;
# We should take the called_number as username in table
# Also validate if the number has active that DID,
balance if needed, etc.
# route(validate_customer);
# I know, hardcoded for testing
sql_xquery("ca", "select * from location where username
= 1212 ", "ra");
$avp(location_addr) = $xavp(ra=>received);
xlog("DID did_incoming location is $xavp(ra=>received)
\n");
# Validate SIP OK
#setflag(SIP_IP_OK);
#$rU=$avp(dnis);
#$rd=$avp(identity);
#$rp=$avp(port);
# uac_replace_to($xavp(ra=>contact));
uac_replace_to("sip:1212@voip.x.net:5060");
# $ru = $var(duri);
$ru=$xavp(ra=>contact);
$ru="sip:1212@voip.x.net:5060";
route(RELAY);
# t_on_failure("dst_failure");
# t_on_reply("dst_reply");
if(!t_relay()) {
sl_reply_error();
exit;
}
exit;
}
--
Regards,
Jose N Figueroa
Greetings,
I have two machines with Kamailio in a HA setup with replicated DB. For
simplicity let's say each machine has one HA IP and that IP can jump to the
other machine in case something happens (kamailio stopping, etc).
I'm using Dispatcher with load balance configuration. I have Dispatcher
configured so that each peer has one instance for socket with HA IP 1 and
HA IP 2.
In order for this to work correctly on the load balance scenario I must
disable via RPC command the peer which has the socket that doesn't belong
to the machine. I also must do it every time the IP jumps back and forth,
which adds complexety to my system.
Does Dispatcher has any sort of help on this? It could not include peers
with sockets IPs that don't belong to the machine in the destination set,
for example. Is this possible?
I could also work with failover support but i would rather avoid having so
many failovers.
Best Regards,A
Hi,
I have installed rtp engine inside pcscf pod and tried to make a call.
caller was unable to reach the pod environment with 2223 port exposed for
rtp. Kindly anyone help, if you have already done with pod based deployment
of ims.
Thanks,
Pavithra
Hi,
I have using kamailio with rtpengine, I have started rtpengine session
(rtpengine_offer() /rtpengine_answer() )only when call getting hold and then
play media file and on un hold detach rtpengine session(rtpengine_delete()).
during this activity I have received the log "No supported output codec
found in SDP" from rtpengine. And getting silence on other leg.
Above activity log are attach in mail.
if we do same step again (i.e. hold/unhold) in same call then second leg
listening the file perfectly and rtpengine working fine vice-versa.
I want to know , what else missing to configure here. Pls guide me to get
out this.
Regards
Amit Pal
Hi,
I have a dispatcher.list file like the following:
1 sip:127.0.0.1:5060 0 0 duid=abcd;maxload=5
1 sip:127.0.0.2:5060 0 0 duid=efgh;maxload=5
After calling the ds_select_dst("1", "4") function(ie, round-robin
selection), I want to read the value of the "duid" attribute of the
selected destination and then pass it to a database. How can I read this
value in the configuration file?
Jani Biju Babjan
I am seeing this error in the logs
i am using kamailio 5.4.1
any idea what the cause is?
ERROR: <core> [core/socket_info.c:2225]: parse_protohostport(): bad
protocol in sip:myp:5060
