sr-users May 2014

sr-users@lists.kamailio.org

96 participants
155 discussions

My kamailio.cfg has ofen not registred , please help me check my registrar process is right or wrong

by vast

Hi, My kamailio has ofen not registred ,please help me check my registrar process is right or wrong (I think the MSILO modle maybe tedious or incorrect).this is my kamailio.cfg file.

9 years, 11 months

Kamailio with Sqlite

by Ashwin Kumar R

Hi, I want to configure kamailio to use SQLITE instead of MYSQL.I made DB_ENGINE=SQLITE in kamctlrc file and loadmodule "db_sqlite.so" in kamctl.conf file. But when I changed these parameters modparam("auth_db", "db_url", "sqlite:///etc/kamailio/kamailio.db") #!define DBURL "sqlite:///etc/kamailio/kamailio.db" Im getting the following error ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I checked the log file ,following is the comment in syslog May 20 19:16:30 r13pc80-desktop /usr/local/sbin/kamailio[10920]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'etc/kamailio/kamailio.db' What kamailio.cfg and kamctlrc parameters do I need to change to do this? Regards Ashwin

9 years, 11 months

Accounting based on location/contact

by John Murray

Hello, I have subscribers that have 2 devices in the location table. These can make and receive call correctly and I can account for the calls. However I need to account which device has made or received the call as there are charges for one device type but not the other. I can modify the contact string from the device to add a parameter for example but I can't see how to add that into the accounting records using tm and acc modules. Thanks for your help. Regards John

9 years, 11 months

i use the csipsimple registrar kamailio get request time out

by vast

Hi, I registrar kamailio ofen can registrar .but at other wifi some time i use the csipsimple registrar kamailio get request time out . when i look up the log I have found ERROR: registrar [save.c:692]: update_contacts(): invalid cseq for aor <1008> I search on google but can`t find a good answer. some one tell me it's not a same callid and low cseq . what would i do for this error ?

9 years, 11 months

Asterisk Integration w Kamailio and Zoiper phone problems

by Brian Watson

Hi, I have a virtual environment setup and am trying to get the integration of Kamailio and Asterisk to a least be able to make a phone call based off the Asterisk Hello World example. I do the following: 1. I'm running Ubuntu 14.04 and Oracle Virtual Box 2. I Integrate Kamailio and Asterisk based off of the asipto kb: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb with both of them running on the same VM. 3. I've modified sip.conf and extensions.conf to have entries that tie together phones 101 and 102 3. I start asterisk with "asterisk" and "asterisk -rvvvvvv" 4. I install and setup the Zoiper Softphone as indicated in the tutorial. https://wiki.asterisk.org/wiki/display/AST/Hello+World 5. In Zoiper I hardcode the Domain: xxx.xxx.xxx.xxx:5080 and also enter the username and password. 6. I'm doing all of this without access to an internet connection so that's why I hardcode the IP address. 7. After setting of the Zoiper softphone I try to get it to communicate with Asterisk, but I get a 408 request timeout. Any ideas as to what might be wrong? Thanks, Brian

9 years, 11 months

Kamailio and RTPProxy

by Thyrus Gorges

Hello all, I am currently trying to configure Kamailio and rtpproxy to play back a short message to the user-agent. I’ve started rtpproxy using: rtpproxy -F -s udp:127.0.0.1:7722 -l 10.1.10.102 -d WARN:LOG_LOCAL0 -u root I created the necessary sound file using makeann from the source of rtpproxy. Below are the relevant pieces of kamailio.cfg # ----- rtpproxy parameters ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") Kamailio.cfg ————— # Validate that the User Agent via IP Authorization via a SQL table if (!allow_address("1", "$si", "$sp")) { sl_send_reply("403", "Forbidden"); xlog("**** $si is UNAUTHORIZIED ****"); exit; } xlog("Enter route 1"); if (has_body("application/sdp")) rtpproxy_answer(); if (is_method("INVITE")) { xlog("New Call for route [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]"); if (rtpproxy_offer()) { t_on_reply("1"); } else { t_on_reply("2"); } #t_on_branch("2"); #t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[1] { if (has_body("application/sdp")) rtpproxy_manage(); rtpproxy_stream2uas("/var/rtpproxy/prompts/connect", "-1"); exit; } onreply_route[2] { if (has_body("application/sdp")) rtpproxy_offer(); exit; } When starting Kamailio, it informs me that it found the proxy and it enabled support for it. May 21 20:44:50 localhost /usr/local/sbin/kamailio[23816]: INFO: rtpproxy [rtpproxy.c:1607]: rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled Now, once I start up a call, rtpproxy reports the following information: May 21 20:14:35 localhost rtpproxy[23379]: INFO:handle_command: new session MTFiNWU1N2NiZGQwNWMwYzVkN2MzZjUyNzhkNTU3YWE., tag 88fc4d50;1 requested, type strong May 21 20:14:35 localhost rtpproxy[23379]: INFO:handle_command: new session on a port 41236 created, tag 88fc4d50;1 May 21 20:14:35 localhost rtpproxy[23379]: INFO:handle_command: pre-filling caller's address with 10.8.0.10:8000 May 21 20:14:35 localhost /usr/local/sbin/kamailio[23528]: ERROR: <script>: rtpproxy_offer returned true May 21 20:14:35 localhost /usr/local/sbin/kamailio[23522]: ERROR: rtpproxy [rtpproxy.c:1771]: select_rtpp_node(): script error -no valid set selected May 21 20:14:35 localhost /usr/local/sbin/kamailio[23522]: ERROR: rtpproxy [rtpproxy_stream.c:113]: rtpproxy_stream(): no available proxies Any assistance is greatly appreciated ! — Thyrus Gorges

9 years, 11 months

Presence

by Joel White

I am working on enabling presence and bla in a Kamailio + FreeSWITCH environment All handsets are Polycom with a handful of Grandstream ATA's I enabled presence and presence_xml I have not got it to work as I keep seeing this meesage in the logs NOTICE: presence [subscribe.c:1030]: handle_subscribe(): Unsupported presence event call-info When I searched online for this error, the answers I got were saying this was a Linksys/Snom message.... How do I get presence to work properly? Thank you in advance

9 years, 11 months

uri parameter value case sensitiveness

by Daniel-Constantin Mierla

Hello, hopefully someone here with the RFC fresh in mind and its grammar for sip uri parameter can jump on to clarify if the value of the parameter can be changed to lower/upper case. For a specific example, is it allowed that the value of a parameter in contact header uri is changed to upper/lower case when building the sequential request (so this appears in request uri)? Same would be for Recor-Route uri parameter that would appear later in Route headers... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

9 years, 11 months

Realtime integration: Unregistered clients showing as registered?

by Olli Heiskanen

Hello, One of the tests I've been working with is Asterisk realtime integration according to Daniel's guide here: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb Weird thing is the client looks registered but I'm not sure if it really is registered. If I'm not mistaken I should see the peers when I issue 'sip show peers' on asterisk cli. Instead I get this: *CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] Also, calling between clients will fail; in Asterisk cli I get: *CLI> == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [661@default:1] NoOp("SIP/660-00000000", "Testing: Dialed 661") in new stack -- Executing [661@default:2] Dial("SIP/660-00000000", "SIP/661,3600,rt") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/661 == Everyone is busy/congested at this time (1:0/0/1) -- Executing [661@default:3] Hangup("SIP/660-00000000", "") in new stack == Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-00000000' In this setup I have SIP peers in an asterisk table added like this: INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser, fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com '); I have Kamailio and Asterisk on the same machine where Kamailio listens port 5060 and Asterisk listens 5070. Things that differ from the guide are Kamailio and Asterisk versions, which in my case are newer. Also, for another testing case I have MULTIDOMAIN enabled in Kamailio, does this interfere with the realtime integration? I'm using only one domain though. Please let me know if any configs or traces I can provide will help figure out what's going on. cheers, Olli

9 years, 11 months

Upgrade to version 4.1

by VOIP Tests

Hello, I understand that there are some bugs in ver 4.03. Can someone let me know how I can update my running version of Kamailio ( 4.03 ) to version 4.1? Will I need to do a fresh install of the new version or is there a way I can do an upgrade/update to ver 4.03? Thank you, Arun

9 years, 11 months

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sr-users May 2014