sr-users August 2015

sr-users@lists.kamailio.org

79 participants
123 discussions

Start a nNew thread

upgrade V4.2.5 to V3.1 -Build error

by Yasin CANER

8 years, 9 months

XCap Server + Postgre

by Igor Olhovskiy

Hi! I'm experience really strange behavior of Kamailio. Right now it's using postgre database When trying to implement XCAP server capabilities, according to http://kamailio.org/docs/modules/4.3.x/modules/xcap_server.html#idp76784 on a block if (!www_authorize("xcap", "subscriber")) { www_challenge("xcap", "0"); exit; } While trying to test it via browser (user test, pass test) I've got these errors in log Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012] Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012] Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012] Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: ERROR: db_postgres [km_dbase.c:247]: db_postgres_submit_query(): 0x7f3fb0660b10 PQsendQuery Error: ERROR: relation "subscriber" does not exist#012LINE 1: select password from subscriber where extension='test' AND u...#012 ^#012 Query: select password from subscriber where extension='test' AND user_context='xcap' Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Aug 12 12:34:57 sip01 /usr/sbin/kamailio[7655]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Without this block, I don't get user auth. Is it a bug or it's mine misconfiguration? -- Best regards, Igor

8 years, 9 months

kamailio as SIP Agent

by Sandeep Chakravarthi

Hi, We have a requirement with one of our telco We are using asterisk in our servers and we are planning to implement SIP-I protocol and we choosed kamailio for it. In Kamailio website, i came to know that kamailio will be supporting both SIP-I and SIP-T protocols Below is what we need and pls confirm whether it is possible or not? Asterisk PBX <-------> Kamailio <--------> Telco MSC Telco will be forwarding the calls to kamailio on sip-i protocol and kamailio server has to forward the calls to our Asterisk server by converting sip-i to standard sip protocol Similiarly Asterisk will be initiating sip call to kamailio server and kamailio server should convert it into SIP-I and should forward the call to Telco MSC 1. I am able to establish the SIP trunk [sending OPTIONS from asterisk and kamailio acknowledges with 200 OK] between Asterisk and Kamailio using dispatcher module in kamailio and sip.conf in asterisk. How to establish the SIP trunk between kamailio and telco MSC? [Generally MSC will act as SIP server and kamalio should send OPTIONS packet and MSC will acknowledges with 200 OK] My telco MSC has only provided me the MSC SIP IP and there were no username/passwords provided. Means i need to use IP based authentication for the SIP Trunk establishment. In Kamailio how to achieve it? Please help and any suggestions/feedback will be highly appreciated and thankful Regards, Sandeep

8 years, 9 months

sdp manipulation in connection with rtpengine_offer/answer calls

by Juha Heinanen

it was not clear to me based on rtpengine module readme if rtpengine_offer/answer are same kind of sdp manipulation functions as, for example, sdpops module functions, i.e., will rtpengine functions pass to rtpengine the sdp as it is after possible earlier sdpops function calls and will possible sdpops functions calls after rtpengine function call keep on manipulating the sdp returned from rtpengine function call. -- juha

8 years, 9 months

RTPProxy incorrect port 0 in reply from rtp proxy

by Grant Bagdasarian

Hello, I've been load testing kamailio(4.2) with rtpproxy(2.0) using sipp, and when I hit 50+ CPS I start getting the following error: "incorrect port 0 in reply from rtp proxy". I'm running 4 instances of rtpproxy on the same instance as kamailio. Could somebody tell me why I'm receiving this error? Do I need to run more instances of rtpproxy? # <#----------RTPPROXY Parameters----------#> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7721 udp:127.0.0.1:7722 udp:127.0.0.1:7723 udp:127.0.0.1:7724") In request_route: if (is_method("INVITE")) { record_route(); if (has_body("application/sdp")) { rtpproxy_manage(); } } In onreply_route: if (has_body("application/sdp")) { rtpproxy_manage(); } rtpproxy -s udp:127.0.0.1:7721 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 10.0.0.1 rtpproxy -s udp:127.0.0.1:7722 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy1.pid -l 10.0.0.1 rtpproxy -s udp:127.0.0.1:7723 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy2.pid -l 10.0.0.1 rtpproxy -s udp:127.0.0.1:7724 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy3.pid -l 10.0.0.1 Regards, Grant

8 years, 9 months

Re: [SR-Users] LCR Routing - Cost based routing

by DanB

Hi Alexandru, Well the concept is simple: on INVITE, you will issue a special format evapi request. You will receive the list of suppliers in the reply (which is received in a separate route) and you continue processing from there. These suppliers are simple tags so your creativity will be the limit of what you can do with them - eg: direct ip addresses received or tags of suppliers which you can further translate with drouting module or any alias related one. For more information on LCR please find this year's presentation on Kamailio World (which I strongly recommend to be attended): https://www.youtube.com/watch?v=Hsvcwleb-fY You can also find the kamailio configuration which I have used in the worshop here: https://github.com/cgrates/cgrates/tree/master/data/tutorials/kamevapi/kama… Let me know if further questions (or join our irc or mailing list so we do not create too much noise here). DanB

8 years, 9 months

Help with sip balancer

by Bruno

Hello, i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP registration from VoIP provider and route the calls to the asterisk boxes where an IVR will take care to answer. Here's my network topology: +---> [asterisk1] [public_ip] | 10.50.10.131 [router] <---NAT---> [kamailio] <---+ 10.50.10.1 10.50.10.120 | +---> [asterisk2] 10.50.10.132 In my setup i planned to use UAC and DISPATCHER modules. I started from the "kamailio-basic.cfg" and added some extra lines to handle UAC and DISPATCHER. All is working fine when i do a test call from a softphone inside network 10.50.10.0/24. When a call is coming from the sip carrier, troubles occurs because asterisk boxes are sending their internal ip in SDP. I understand that i need to rewrite SDP in that case, but i actually don't know how/where. I've attached kamailio configuration and a sip trace taken with sngrep where the problem is visible. For security reasons, i would like to force the RTP through RTPProxy. I'm missing something, and need your help me to understand my errors. Best Regards, Bruno

8 years, 9 months

Trailing newline with {re.subst,expression}

by Daniel Tryba

I'm unable to get rid of the \n in "multiline" content of a var. xlog("var: $var(sdp)"); $var(sdp)=$(var(sdp){re.subst,/^a=rtcp:.*//}); xlog("var: $var(sdp)"); a=sendrecv#015#012a=rtcp:31543#015#012a=X-foo:bar becomes: a=sendrecv#015#012#012a=X-foo:bar Using the s flag results in no substitution at all, /^a=rtcp:.*//s should result in a=sendrecv#015#012 according to http://www.kamailio.org/wiki/cookbooks/4.3.x/transformations#resubst_expres… Is this a bug or a feature? If it is a feature, how to get rid of the newline (without additional regexps substitutions).

8 years, 9 months

force_send_socket/$fs behaviour when binding to INADDR_ANY

by Alex Balashov

Hello, When binding Kamailio to 0.0.0.0, Kamailio no longer recognises any specific IP address homed on the system as being a "socket" for purposes of forcing traffic out of any specific interface (i.e. setting $fs), or any other purpose for which ingress and egress "sockets" are tracked (e.g. double RR). As I understand it, this is because Kamailio considers only addresses explicitly specified via the 'listen' core config directive to be valid "send socket" arguments. Kamailio _will_ accept addresses already attached to the interface if it is bound interface-wise, e.g. listen=eth1:5060 but not when addresses are added to the interface without restarting the proxy: # ip addr add dev eth1 172.30.110.10/24 Aug 10 17:31:35 centosity6 /usr/local/sbin/kamailio[28627]: WARNING: pv [pv_core.c:2285]: pv_set_force_sock(): no socket found to match [udp:172.30.110.10:5060] Is there any straightforward way to modify this behaviour, so that it would be possible to dynamically add addresses to existing physical interfaces and get Kamailio to utilise them as if they were bound at boot time? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 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, 9 months

Re: [SR-Users] Unable to limit call duration with timeout_avp

by Admin

Thanks Alex. That worked.

8 years, 9 months

← Newer
1
...
6
7
8
9
10
11
12
13
Older →

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 August 2015