Hi!
I am trying to use the app_java module with kamailio-4.4.0 SIP server.
Every time I load the app_java module, my users can't register and the
sever replies with destintion unreachable (port unreachable).
How can I fix this issue?
Also, how can I use the Kamailio.java file since it doesn't have a main
method?
Thanks
Hi all,
it seems that the AVPs not available when when processing CANCEL
message, even though they have been set for this transaction initially.
Is this the expected behavior?
P.S. kamailio 4.3.4
Andrew
Per my understanding the uac module stores the "vsf" parameter in
Record-Route and should be able to update the From/To URIs automatically
in all in-dialog requests that carry this parameter.
http://kamailio.org/docs/modules/stable/modules/uac.html#uac.p.restore_mode
“auto” - all sequential requests and replies will be automatically
updated based on stored original URI. For this option you have to set
“modparam("rr", "append_fromtag", 1)”.
What makes me wonder is: does that only work in From/To was changed by
uac_replace_from()/uac_replace_to() or also when assigning directly to
the $fu and $tu variables? I'm changing those variables and I am using
restore_mode auto but that does not change anything in in-dialog ACK.
I presume this is expected behavior because I'm not using uac when
assigning to variables, isn't it?
Thanks.
Andrew
Hi,
I've got an installation that is still running Kamailio 1.5.x. Yes, I
know: "They're running WHAT?" As far as I'm concerned, it was EOL'd in
early 2010. Nevertheless, that doesn't stop the customer from running
it. They're probably the most change-resistant customer in the history
of IP telecom, and, knowing this industry, that's a high bar. There's
not much I can do about it.
Anyway, I have a bit of a mystery:
GW Proxy PBX
==================================================
------- INVITE ------>
<---- 100 Trying ----
-------- INVITE -------->
<------ 100 Trying ------
<- 503 Service Unavailable
---------- ACK --------->
<- 500 Service Una...
-------- ACK -------->
There is no basis in the configuration for the translation of the "503
Service Unavailable" reply to "500 Service Unavailable", so I'm thinking
it has to be rooted in some default module behaviour.
Any insight would be appreciated!
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Hello everybody,
Can anybody help me out with sending a geolocation in the SIP messages that Kamailio sends? In order for the emergency telephone number to work properly, we need to send the location of our SIP trunk customers along with their INVITE when calling the emergency phone numbers.
Is there any module that will facilitate this, or is there another way to add this information in the INVITE message?
Thanks in advance!
Best regards,
Michael Jepson
Hello,I'm using Asterisk to originate a call via Kamailio.Request-URI in SIP INVITE coming from Asterisk looks like this<sip:kamailio@x.x.x.x>But my objective is to use Kamailio to forward the call to a remote endpoint.
What header should I put in the SIP INVITE from Asterisk to Kamailio to conveythat Kamailio should use this 'SIP URI' to route the call onwards.I tried 'Route' header, but it doesn't seem very clean, as kamailio doesn't updatethe Request-URI in the forwarded INVITE if I use the Route header.
Thanks,Nitesh
Good afternoon all,
I have two questions regarding my configs: kamailio.cfg<http://pastebin.com/32PCh8n0> tls.cfg<http://pastebin.com/gnWZeD9e> . I am using Kamailio 4.3.5 on Cent OS 7.2 and all relevant pakage up-to-date.
1.) When I try to start Kamailio I get the following warning:
WARNING: tls [tls_mod.c:287]: mod_init(): tls support is disabled (set enable_tls=1 in the config to enable it)
Q: As you can see (line 36) TLS is enabled in config. So what could be the problem?
2.) Lines 324-330 indicate that I want to redirect to 2031 if I'm called at 1. For some reason the SIP-Trace tells me its still building the INVITE with the To-Header to sip:1@PSTN:5060 and the PSTN can't find 1, because no 1 is registered.
Q: Why? (Outbound is set the same way and working perfectly).
Info: I tried already with uac_replace_to and uac_replace_from w/o success.
I would very much appreciate any hints on solving these issues.
Very Respectfully
Dimitry Nagorny
Trainee
Hi all,
I am trying to use an ldap directory for authentication and found this
tutorial.
https://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth
What I want is to use ldap bind with user's existing ldap password, not a
seperate sip password. I would like to use same password for simplicity.
There are two ways in my mind to achieve this and I need suggestions. One
way is to add a new field to the ldap tree and populate it with user
password's ha1 values but this needs an additional work and doesn't seems
nice to me.
Second way is to get clear-text password from auth request and bind to ldap
server with this password but I don't know if I can access to the password
or it this is a secure way.
Does anybody have any suggestions or experience to share?
Regards,
/Volkan
Hi,
Shouldn't http://deb.kamailio.org/kamailio point to 4.4 version?
And question about recent Ubuntu version support: Trusty is way to far
behind latest Willy
--
Serge S. Yuriev
Lead VoIP engineer
