I need to hang up a call after some seconds after received 183 provisional
response but before any other SIP message.
I am using "rtimer" module to set a callback on time, but how can I search
for a call on transaction module in order to terminate the call?
I think I am missing something to prepare the transaction data before I can
use this data to terminate the call.
I am thinking of storing the call-Id of all transactions to use on timeout
callback, I don't know if it's possible.
Thanks in advance.
Antonio
Hello all
we are using the phonenum_match_cn function to detect valid phone numbers,
but seems there are some numbers like 576015145834 which are no detected as
valid ones,
$phn(ani=>valid) --> 0
$phn(ani=>ccname) --> <null>
$phn(ani=>cctel) --> 0
$phn(ani=>error) --> Invalid number
but using
https://libphonenumber.appspot.com/phonenumberparser?number=576015145834&co…
seems it gives a valid result
Validation Results
Result from isPossibleNumber() true
Result from isValidNumber() true
Result from isValidNumberForRegion() true
Phone Number region CO
Result from getNumberType() FIXED_LINE
we are using kamailio 5.4 and libphonenumber tag v8.12.30 (also tested v8.9)
the command is used with 'ZZ' cnc and passing the number with +
maybe we are missing something in the configuration/setup?
thanks and regards
david
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NOC engineer
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Hi,
I'm running a cloud infrastructure with multiple sip domains and I have
some kamailio's as registrars with dmq for usrloc replication.
The kamailio registrars are configured without db, so the aors are in ram.
For some gui's I need to get all aors for one sip domain, If I execute
ul.dump I get all aors, but I want only a few. In my situation it would be
perfect to get all registrations for one sip domain in one request (memory,
performance, cache at gui...)
I thought I could save the registers with save("$rd") but ul.dump doesn't
allow any parameter that restricts the "location" domain.
Is there any solution for this?
Best regards
--
Jose Fco. Irles Durá
hello i have problem on receive source port rtp packet
client-ip = 5.202.105.250
server-ip = 194.5.205.33
client2-ip = 188.212.241.46
In the 200 ok packet from client2 ,The value 4002 is set in the audio port
[image: invite.jpg]
I expect rtp packets to be sent to me from source port 4002, but these
packets are sent to me from another source , which is port 55755 in the
image below, and I just opened this port(4002) in the firewall.
If I turn on my firewall service, all packets will be dropped. I have to
have this firewall
[image: rtp.jpg]
thanks for help me
Kamailio allows the use of an interface name (eth0 for example), as the identifier for the listen directive. Additionally it allows assignment of a name to be used as an identifier. This can be used, as indicated in the documentation:
A unique name can be set for sockets to simplify the selection of the socket for sending out. For example, the rr and path modules can use the socket name to advertise it in header URI parameter and use it as a shortcut to select the corresponding socket for routing subsequent requests.
I'm wondering if there is a way to get allow either of these parameters to be used to define DMQ's server_socket parameter. This would be useful in containerization situations where the local IP address is assigned dynamically. Here's what I'm attempting:
#!KAMAILIO
loadmodule "tm"
loadmodule "sl"
loadmodule "dmq"
loadmodule "textops"
listen=udp:eth0:5060 name "sip5060"
listen=udp:eth0:5090 name "sip5090"
modparam("dmq", "server_address", "sip:dmq_quicktest:5090")
modparam("dmq", "notification_address", "sip:dmq-bus:5090")
modparam("dmq", "server_socket", "sip5090" )
## Above line doesn't work, neither does the line below:
#modparam("dmq", "server_socket", "udp:eth0:5090" )
modparam("dmq", "multi_notify", 1)
modparam("dmq", "num_workers", 3)
modparam("dmq", "ping_interval", 30)
request_route {
exit;
}
I get the following error during startup:
0(1) ERROR: dmq [dmq.c:241]: mod_init(): server_uri is not a socket the proxy is listening on
Thoughts? I realize that I can configure and pass static networking with the containers, but I'm looking for a way to have this set on the fly - especially when testing, etc.
Ben Kaufman
Hello !
I have this issue with the telco provider they give the sip trunk on a
private conection with private ips, the problem is that i am trying to
relocate the sip trunk in remote server and I've tried though basic
networking/tunnels/vpn and no go; their sbc is dropping requests and
rejects if they are more via headers, they force me to use that conection
and the ip allocated.
The challenge is how to this is in kamailio, basically i just want to
transport the trunk (similiar to a reverse proxy for web) to its new
destination and from there handle everything.
Any hints on how to achieve this?
Dear List,
I’m looking at using Kamailio as SBC.
This would replace our current FreeSWITCH-based SBCs.
What are your thoughts on this move? Would there be any known reasons why Kamailio would be superior to FreeSWITCH as an SBC?
Thank you so much
Unai Rodriguez
Hello guys,
I have a failure route which calls another route, and on this route I need
to
t_drop_replies();
But i'm getting:
[core/cfg.y:3510]: yyerror_at(): parse error in config file
/etc/kamailio/include/relay_routes.cfg, line 143, column 20: Command cannot
be used in the block
Any way to fix this without me having to use it in the failure_route?
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
