sr-users February 2016

sr-users@lists.kamailio.org

80 participants
135 discussions

Kamailio 4.3 and Freeswitch 1.0 Stable Latest
by Barış Şekerciler 02 Feb '16

02 Feb '16

Hello everyone. I'm a student and I'm working on project which about SIP communication and PBXs with IP phones. So, my first test results were OK with JITSI. But I think I need Freeswitch for working with IP phones. In a word, I need configure an integration Kamailio and Freeswitch. First, I setup Debian 8.3 Jessie and installed Kamailio on this my local server, after that I installed another Debian 8.3 Jessie (from same image) as my second server and it was going to run Freeswitch on itself. Then I tried to send my communication requests to my Freeswitch server. I made some tests, my Jitsi accounts which are set in Kamailio server were done. I've read Kamailio 3.2 and Freeswitch integration from the Kamailio wiki and I applied exactly the same thing but I did not work. When I tried to reach another IP phone from my IP phone (I registered them) phone always say "NOT FOUND". I'm sure I did my configuration properly. So, how can I figure it out these things? How can I understand, does Kamailio routes to Freeswitch to my phones? Best regards. Thanks.

2 1

Critical issue: Kamailio throwing db errors
by Sharon Nathaniel 02 Feb '16

02 Feb '16

Hi, I am not sure, if its a kamailio issue or something else, but our Kamailio 4.2 server crashes everyday. Throwing out these logs: http://pastebin.com/z1MWdk7q. We have to restart the server to get this fixed. Please help us understand what is causing the issue. Thanks and Regards, Sharon Nathaniel The information in this email is confidential and may be legally privileged. It is intended solely for the addressee and access to it by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken based on it, is strictly prohibited and may be unlawful.

2 1

Kamailio v4.3 and Freeswitch Stable Latest
by Barış Şekerciler 02 Feb '16

02 Feb '16

4 27

Force transport from TLS to UDP when using dispatcher
by mayamatakeshi 01 Feb '16

01 Feb '16

Hello, I have this scenario: - subscriber registers using TLS - call arrives from PSTN gateway (in UDP) to subscriber - I relay the INVITE to subscriber but it doesn't answer the call - the subscriber has callforward enabled - we send the call to PSTN callforward destination using carrierroute In the above, after calling cr_route I use this: $du = "sip:" + $rd + ":" + $rp + ";transport=udp"; to force the call to PSTN gateway to go out as UDP because after trying to call the subscriber, the transport changes and stays as TLS. However, if instead of callforward the user has voicemail enabled, we use dispatcher to send the call to a cluster of media servers. So, after calling ds_select_dst() I call $du = "sip:" + $dd + ":" + $dp + ";transport=udp"; but this is unable to force the call to go as UDP but it still goes as in TLS In both cases I see this in the log files: /usr/local/src/git/kamailio-4.3/kamailio[19277]: WARNING: <core> [forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:XXX.XXX.XXX.XXX:5061, to udp:YYY.YYY.YYY.YYY:5060) But it only works for carrierroute, not for dispatcher. I am not sure if this is a bug in dispatcher. But anyway, is there any other way I could try to force transport=UDP? Regards, Takeshi

1 1

checking source domain
by Yuriy Gorlichenko 01 Feb '16

01 Feb '16

I need to understand where from packets received. Now I use something like If $si == "1.2.3.4" { xlog("L_INFO","bla bla bla"); } But I need to check source server not only by IP and PORT, but at Domain too For example if (some_pseudovariable=="pbx.server.com"){ xlog("L_INFO","bla bla bla"); } I can use $fu for example because for some packets it includes domain name of kamailio (i think details not important but this situations can be) Does kamailio have some mechanisms to do that? I searched it at cookbook but not found anything. Thank you

2 1

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sr-users February 2016