sr-users July 2019

sr-users@lists.kamailio.org

68 participants
77 discussions

kamailio + asterisk integration RTP flow question
by Satish Patel 18 Jul '19

18 Jul '19

I am configuring kamailio as registar and asterisk for media services. in kamailio + asterisk integration how two SIP clients RTP traffic will flow, are they go via Asterisk or directly from kamailio ?

3 2

SIP-DTLS support
by Martin Meadows 17 Jul '19

17 Jul '19

Hello, Has there been any movement on the support of SIP-DTLS (datagram-TLS) within Kamailio since: * sr-users:2012-October:075185.html <https://lists.kamailio.org/pipermail/sr-users/2012-October/075185.html> I'm interested to see if Kamailio can proxy 5061/udp to a FreeSWITCH 5060/udp. In general, I have been unable to find any material on implementations of: * draft-jennings-sip-dtls-05.html <https://tools.ietf.org/id/draft-jennings-sip-dtls-05.html> Cheers - Martin

4 6

opt_reply(): called for non-OPTIONS request
by Alexandru Covalschi 17 Jul '19

17 Jul '19

Hello list, if KSR.is_OPTIONS() and KSR.is_myself_ruri(): KSR.x.modf("options_reply"); exit(); with a valid URI (no username) gives the subj output (opt_reply(): called for non-OPTIONS request) How could that be? Python KEMI, 5.2.3, docker. ________________________________ Regards, Alexandru Covalschi VoIP Engineer and System Administrator tel: +37367367850

2 5

Kamailio IMS to IP PBX - how?
by David Dean 16 Jul '19

16 Jul '19

I'm trying to connect a VoLTE smartphone to an IP PBX using Kamailio IMS. I've managed to get the smartphone to connect to Kamailio IMS and make calls with our devices registered with Kamailio. The problem is that I need the smartphones to register with the IP PBX which is a VigorBX 2000n. This is so they can use an internal extension number and make calls to other local extensions with hard wired desk phones. Should I be using Kamailio IMS to "proxy" the smartphone SIP connection to the VigorBX PBX? Or would it be better to connect the smartphone directly to the VigorBX using SIP? If so, how? I've read absolutely every how-to document I could lay my hands on but none of them describe integrating IMS with a PBX. Thanks for any help / advice you can offer. Dave

1 0

KEMI beginner question on alpine Linux
by Karsten Horsmann 16 Jul '19

16 Jul '19

Hi List, I play around with alpine Linux 3.10.x and Kamailio 5.2.x and I search for app_python3 and KEMI support. But I don't found it. AFAIK if I want to use KEMI with Python3 I need of course Kamailio and app_python3 (maybe something else?). Is the package maintainer on this list? Cheers Karsten

4 4

How to get total number of dialogs in profile with value
by Nathan Angelacos 15 Jul '19

15 Jul '19

In kamailio < 5.1.x it used to be possible to get total number of items in a dialog_with_profile by skipping the "value" paramter, such as: modparam ( "dialog", "profiles_with_value", "outbound" ) . . . get_profile_size ( "outbound", "$var(myvar)" ); However, 5.1.x and 5.2.x report: ERROR: dialog [dialog.c:912]: w_get_profile_size_helper(): invalid value parameter The documentation ( https://kamailio.org/docs/modules/5.2.x/modules/dialog.html#dialog.f.get_pr… ) Indicates the "value" paramter is still optional. I'm sure I must be missing something obvious... Any ideas? Thanks!

2 3

UDP fragmentation issues
by Gerry | x164 15 Jul '19

15 Jul '19

Hi, I have setup Kamailio on Centos 7 as a proxy in front of Asterisk in the same datacenter, using the dispatcher module. I have noted that large INVITEs sent by Kamailio do not arrive at the Asterisk box. It seems the network is dropping fragmented UDP traffic. This does not happen however when I contact the Asterisk box directly from outside the datacenter. Then large INVITES are arriving are Asterisk. So the problem is either in Kamailio or in the datacenter. I have analysed the problem with ngrep and tcpdump/wireshark. In normal mode (udp4_raw=no setting) Kamailio is producing UDP packets above 1500 bytes and leaves all fragmentation to the operating system. Enabling raw udp in Kamailio and lowering the MTU in the server didnt help. I even disabled offloading of UDP fragmentation to the NIC to get to the bottom of this. However nothing helps. Only forcing TCP in the dispatcher works when having large INVITES going to Asterisk. In any case, it appears that there is no guarantee that fragmented UDP arrives at all. As far as I am aware, the recommendation is to use TCP if messages are bigger than 1400 bytes. If so, how can I force Kamailio to send messages over 1400 bytes only in TCP mode? Or is there some other way how to deal with this? Thanks for any hints. Gerry

3 3

Kamailio Asterisk integration with docker DNS Problem
by Amir 15 Jul '19

15 Jul '19

Hi I have created a kamailio container with docker and two asterisk container And this is my dispatcher list: *1 sip:asterisk:50601 sip:asterisk2:5060* and this is my both asterisk SIP.conf *[Kamailio]host=kamailioport=5060insecure=invitetype=friendcontext=from-internal* The problem is that I use User-defined networks which has its own DNS when a container starts a dns record would be set If I start a kamailio first then start two asterisks the asterisk works fine because the name "kamailio" has been set in docker dns but I should restart the kamailio because at startup It could not resolve astersik dns records after the everything works fine but if the any of the asterisk instances crash/stop kamailio can not detect that because a dns record would not be available: *17(22) ERROR: <core> [core/resolve.c:1684]: sip_hostport2su(): could not resolve hostname: "asterisk2"17(22) ERROR: tm [ut.h:309]: uri2dst2(): failed to resolve "asterisk2"17(22) ERROR: tm [uac.c:452]: t_uac_prepare(): no socket found17(22) ERROR: dispatcher [dispatch.c:3110]: ds_ping_set(): unable to ping [sip:asterisk2:5060]* It is kind of loop , I think maybe I can fix asterisk problem with asterisk realtime and set the sip.conf in database. anyone has any suggestion for this problem?

8 14

Wrong $rm value
by Alexandru Covalschi 14 Jul '19

14 Jul '19

Hello list, I use kamailio 5.2.3 from repo's on stretch in docker. I use python KEMI + async and encountered weird behaviour: after t_continue $rm shows OPTIONS instead of REGISTER/INVITE. I wonder if that's correct behaviour, because for me it looks like a bug. Repeating is easy - just check $rm after t_continue. ________________________________ Regards, Alexandru Covalschi VoIP Engineer and System Administrator tel: +37367367850

1 2

Check if an INVITE comes from uac_reg endpoint
by Diogo Hartmann 14 Jul '19

14 Jul '19

Hello Kamailio experts, We were already able to register correctly at a SIP Provider using uacreg table. Now, we would like to validate inbound calls coming from this registered provider, sending them to a specific dispatcher group. I tried reading the uac module docs, found *uac_reg_lookup* function, but i'm not sure if that's we need. Could someone please help me to understand if UAC has a function to do that, or if we need to make some? Thanks!!

3 4

2026

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users July 2019