sr-users June 2018

sr-users@lists.kamailio.org

88 participants
121 discussions

TLS module and IPv6 (using tls.cfg).
by Joel Serrano 17 Jun '18

17 Jun '18

Hi all, I was playing around with the "config" modparam to specify and use a TLS configuration file instead of inline modparams inside kamailio.cfg. I did some testing and when I went to try with IPv6 I run into a little doubt?? tls.cfg config example: [server:1.2.3.4:5061] <----- How do you put an IPv6 in there? method = TLSv1 verify_certificate = yes require_certificate = no private_key = /usr/local/etc/kamailio/tls/local_key.pem certificate = /usr/local/etc/kamailio/tls/local_cert.pem verify_depth = 3 ca_list = local_ca.pem crl = local_crl.pem Applying the logic of the "listen" param, it should be: [server:[AAAA:BBBB:CCCC:DDDD:EEEE:FFFF:GGGG:HHHH]:5061] But it just seems totally wrong and I haven't found anything in the docs. Any suggestions? Thanks, Joel.

2 2

SDPC on SDP
by Duarte Rocha 17 Jun '18

17 Jun '18

Greetings, Please help me understand the reason and source of this behaviour on Kamailio. I receive a 200OK to a RE-INVITE that has : Session Name (s) : MySession Connection Information (c) : IN IP4 xxx.xxx.xxx.xxx Then, Kamailio routes the 200OK to it's destination, and it leaves Kamailio like this : Session Name (s) : MySession sdpc=IN xxx.xxx.xxx.xxx The line Connection Information got deleted and it's info is now on Session Name. How can i disable this ? Thanks

2 1

Alias listen
by kai.ohnacker＠cbc.de 17 Jun '18

17 Jun '18

Hello everybody, I have a question about the function "aliases". With my server no alias entry is registered and there is no DNS server (reverse DNS is not possible) in the network. Nevertheless, the SIP server listens to a defined alias. Is there next to the config file and the kamctlrc file another place where an alias can be entered? Cheers, Kai

2 1

Configuring Kamailio as a Proxy to another SIP trunk
by Ori Popowski 15 Jun '18

15 Jun '18

Hi, I'm making my first steps in Kamailio, and need help for a simple configuration. I am trying to configure Kamailio to forward SIP traffic from (and to) a SIP gateway to a Twilio SIP trunk: SIP gateway ----> Kamailio ----> Twilio SIP trunk How to configure Kamailio for this use case? How the kamailio.cfg file will look like in such case? Thanks.

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add_uri_param with variables
by Patrick Wakano 15 Jun '18

15 Jun '18

Hello list, Hope you all doing well! We've been attempting to add a URI parameter to implement the trunk group (tgrp and trunk-context) but discovered the add_uri_param() function only works with constant string.... we can't use a pseudovar to inform the value to be added. Anyone knows why such limitation? Can it be changed? It is possible to do the same thing via the subst_uri function, but it would be so much more elegant to use the function... Thank you, Kind regards, Patrick Wakano

1 0

outbound flow tokens and kamailio restart
by Pan Christensen 14 Jun '18

14 Jun '18

Hello all. I have created a WebRTC to SIP gateway. I implemented it using the outbound module. If I restart Kamailio during a call, subsequent messages fail to be routed. {1 322 BYE 218565972_26748892(a)x.x.x.x} INFO: outbound [outbound_mod.c:261]: decode_flow_token(): flow-token failed validation {1 322 BYE 218565972_26748892(a)x.x.x.x} INFO: rr [loose.c:519]: process_outbound(): failed to decode flow token {1 322 BYE 218565972_26748892(a)x.x.x.x} INFO: rr [loose.c:794]: after_loose(): failed to process outbound flow-token Is it possible to make the flow token survive a restart? With kind regards Pan B. Christensen Developer Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobile: 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

1 1

Re: [SR-Users] Using a var as timeout value in dlg_set_timeout_by_profile function
by Henning Westerholt 14 Jun '18

14 Jun '18

Am Donnerstag, 14. Juni 2018, 09:30:44 CEST schrieb jenus(a)cyberchaos.nl: > I do not see any errors in the logs, it looks like it just sets the > timer to 0 and disconnects the call. Hello Jan, I just looked quickly in the code, it seems that the timeout value is not interpreted as "pseudo-variable", therefore its not possible to use something like $var in this method right now. It is just read as a numeric value. dialog.c, line 1342: if(dlg_set_timeout_by_profile((struct dlg_profile_table *) profile, &val_s, atoi(timeout_str)) != 0) This can of course changed in the module with a change in the code. Best regards, Henning -- If you like the work that I do in Kamailio, please consider supporting me on Patreon: https://www.patreon.com/henningw

2 1

can't assign values to $fU
by Karthik Srinivasan 14 Jun '18

14 Jun '18

From the online documentation:$fU - From URI username *$fU* - reference to username in URI of 'From' header It is R/W variable (you can assign values to it directly in configuration file) But i am no able to assign values to $fU. Has anyone experienced this? Snippet of code in the configuration file of mine and its respective logs showing that $fU is not getting overwritten. I have tried to hard code values to it assign variables to it, all to no avail. configuration file code snippet: xlog("L_INFO", "[$ci] from header username pre mod: $fU"); $fU = "5555555555"; xlog("L_INFO", "[$ci] from header username post mod: $fU"); Logfile snippet: INFO: (s) [7EAB8FECD723E8444E36C659F331068B41EC6BB0] from header username pre mod: 3224567880 INFO: (s) [7EAB8FECD723E8444E36C659F331068B41EC6BB0] from header username post mod: 3224567880 Thanks, Karthik

4 8

Kamailio as outbound proxy for PBX
by Wilkins, Steve 14 Jun '18

14 Jun '18

Good Morning All! If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio actually make the SIP call? At some point I would like outbound calls to be controlled by Kamailio so that the outside endpoints never communicate with the PBX. Currently a call goes through Kamailio to Asterisk and then Asterisk communicates with the endpoint, and this is night ideal Thanks ALL

2 2

t_save_lumps() and failure_route
by Julien Chavanton 14 Jun '18

14 Jun '18

Hi, I have a scenario where I am using. t_save_lumps(), lookup() and then I can choose to drop some branches in branch routes. However, it is possible that I endup without any branch left, in this case t_relay() is returning and no failure route is called/created. At this point I need to recover the original message I believe it is not possible to restore/reset the message without passing by failure route ? Else, I could trigger a failure route using t_relay(), is there any "handy/clean" way to trigger failure_route() ? Regards

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sr-users June 2018