sr-users April 2016

sr-users@lists.kamailio.org

76 participants
114 discussions

by Clarence Sandjon

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

8 years, 10 months

should the AVPs be available in CANCEL?

by Andrew Pogrebennyk

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

8 years, 10 months

uac "restore_mode" is auto: effective when using core variables?

by Andrew Pogrebennyk

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

8 years, 10 months

A real mystery - 1.5.x vintage edition!

by Alex Balashov

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/

8 years, 10 months

RFC6442: geolocation

by Michael Jepson

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

8 years, 10 months

Kamailio setting the reqeust URI

by NITESH BANSAL

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

8 years, 10 months

Kamailio TLS and SIP-Package building

by Nagorny, Dimitry

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

8 years, 10 months

Ldap authentication

by Volkan Oransoy

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

8 years, 10 months

Misrouting ACK packet for 200 Ok messages.

by Yasin CANER

8 years, 10 months

Stable DEB

by Serge S. Yuriev

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

8 years, 10 months

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sr-users April 2016