sr-users July 2015

sr-users@lists.kamailio.org

86 participants
150 discussions

A question to you, SIP providers: what is your policy on PBXes calling themselves

by Antonio Gómez Soto

Hello, This is question on PBX behavior, what is the right thing to do, and how do PBX's generally behave. If a user on a phone, dials a number, which happens to be configured on the same phone system (for example another tenant), there are two options: 1. The PBX notices this, and directly connects the phone to the DID on that system (breaking separation of tenants) 2. The PBX sends the call out on the SIP trunk, and the provider-routing sends the call back as an incoming call. What are the pros and cons of each option from the SIP provider point of view? How do PBX's generally behave? Thanks, Antonio PS: I reposted, because the original question was apparently not phrased in a clear way.

9 years, 3 months

IPSec supporting open source SIP Server

by Priyaranjan Nayak

Hi All, I need one IPSec supported opensource SIP server for testing my sip client endpoint.Could you please tell me which SIP server supporting IPSec ? Thanks in advance. Thanks Priyaranjan

9 years, 3 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

9 years, 3 months

acc Module missing BYE in database

by Dirk Teurlings - SIGNET B.V.

Hi again, Running kamailio 4.2.5 with the acc module to do accounting in a MySQL database. We are now running into the issue that sometimes there is no BYE line when there is an INVITE. Digging into this I noticed that the dialog:end event does get triggered, and that CNXCC consumes the call properly. So the BYE message does get sent, but acc doesn't seem to pick it up some of the times. It occurs around maybe once every 100 calls. Anybody got any ideas about this? -- Cheers, Dirk Teurlings

9 years, 3 months

Kamailio example trace needed UAR/UAA

by Jan Devai

Hi Experts, we have trouble with the last SIP Registration from I-CSCF to S-CSCF, see below: *The flow is as follows:* UE HSS P-CSCF I-CSCF S-CSCF -----------Sip Register-----------> -------Sip Register-------> <--------------------------UAR--------------- --------------------------UAA---------------> ----SIP Register---> Almost a week ago, I've open a thread about this issue called "Error forwarding to SCSCF", and so far we did not get any further with it :-( I was wondering does somebody have similar Signaling that he could share with us? It could be either with an external or OpenHSS, does not matter at this stage. It would definitely move us forward with at least the signaling comparison as a start point. Maybe the UAA answer and anything happening before is wrong. Any reference, similar working trace would be greatly appreciated. Thank you in advance Jan

9 years, 3 months

About Acc_radius rad_sttaus Function

by Yasin CANER

9 years, 3 months

SDP IPv4 has been concatenate - no audio

by Loic Chabert

Hello everyone, I'm trying put kamailio in front of asterisk server farm. Fow now, 2 asterisk servers are running and i'm trying to make some basic calls between two UACc. All asterisk servers has been ofuscaded from public internet using 10.189.122.0/24 network. All trafic must be passed throught asterisk so RTPproxy is used to (and used for rtp bridging). Kamailio and rtpproxy is running with public IP address, and private ip address (mhomed=1) But a wired thing append on my SDP body: c line have two rtpproxy public addresses concatenate (see my capture attached). Any reason for this ? Only invite method from my asterisk contains 2 publics IP addresses concatenated. Does it mean than rtp_manage as been executed twice ? Thanks in advance. Regards.

9 years, 3 months

Problem adding json module

by Joao Alves

Hi, Have just installed the kamailio 4.3 (from github) and was trying to include the additional modules that I require. I've edited the modules.lst and changed the extra modules to compile "include_modules= db_mysql utils json". During compilation (using "make all") getting the following warnings in relation with the json module. CC (gcc) [M json.so] json_funcs.o json_funcs.c: In function 'json_get_field': json_funcs.c:58:2: warning: implicit declaration of function 'is_error' [-Wimplicit-function-declaration] json_funcs.c:64:2: warning: 'json_object_object_get' is deprecated (declared at /usr/local/include/json-c/json_object.h:303) [-Wdeprecated-declarations] And later, when running the config file: ERROR: <core> [sr_module.c:574]: load_module(): could not open module </usr/local/lib64/kamailio/modules/json.so>: libjson-c.so.3: cannot open shared object file: No such file or directory 0(2344) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 195, column 12-20: failed to load module ERROR: bad config file (1 errors) I have already installed json-c modules and confirmed this installation by doing: root@ID14337:/usr/local/kamailio-4.3/kamailio# pkg-config --cflags json-c -I/usr/local/include/json-c root@ID14337:/usr/local/kamailio-4.3/kamailio# pkg-config --libs json-c -L/usr/local/lib -ljson-c root@ID14337:/usr/local/include/json-c# ls arraylist.h json.h json_inttypes.h json_object_private.h linkhash.h bits.h json_c_version.h json_object.h json_tokener.h printbuf.h debug.h json_config.h json_object_iterator.h json_util.h random_seed.h root@ID14337:/usr/local/include/json-c# cd ../../lib root@ID14337:/usr/local/lib# ls libjson-c.a libjson-c.la libjson-c.so libjson-c.so.3 libjson-c.so.3.0.0 pkgconfig python2.6 python2.7 root@ID14337:/usr/local/lib# So any ideas of what could be causing this to fail? Thanks Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

9 years, 3 months

Errors when starting Kamailio with Websocket Module

by Babu Raja

Hello, I incuded the module for websocket into my Kamailio 4.3 on my Ubuntu 12.04 server environement. For installing the packages and websocket I followed the instructions from the following link: http://www.kamailio.org/wiki/packages/debs http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket After starting the Kamailio I got 26 errors like: can not find command "handle_ruri_alias" or Syntax errors for: route[REGISTRAR] => square brackets.. ($du == "") => parameter "==" unknown.. the whole error log I will paste later.. Can anyone help me?

9 years, 3 months

RLS Notify sent to wrong transport

by M S

Hi, I have a Kamailio v4.3.0 with SVN Rev. c6aa95 running an RLS based presence setup. Everything works fine when end-user has UDP transport, however, if user has TCP or TLS transport then i aggregated NOTIFY sent by RLS gives send error, -- WARNING: <core> [forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:X.X.X.X:5061, to udp:Y.Y.Y.Y:12345) -- (X.X.X.X is server IP and Y.Y.Y.Y is end user IP). Not sure if it is configuration issue or some bug, so posting it to both mailing list. Please help. Thank you.

9 years, 3 months

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sr-users July 2015