sr-users March 2009

sr-users@lists.kamailio.org

92 participants
150 discussions

[Kamailio-Users] lcr-module, ping and load_gws_from_grp (kamailio 1.5)
by Stefan Prelle 12 Mar '09

12 Mar '09

Hi, I just noticed something weird when working with the lcr module. My setup has 5 gateways - 4 in group 1, one in group 2. The lcr-table has an entry for each group, both reacting on the same prefix. I am using load_gws_from_grp("1") to get a gateway for a target. This usually works fine. If now all 4 gateways in group 1 ar disabled because the alive check failed, load_gws_from_grp("1") returns the gateway from group 2 (or other groups where the prefix from the lcr-table matches. I would consider this a bug, since I explicitly requested gateways from a specific group. Regards, Stefan

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[Serusers] Problem with Record-Route not being added
by Frank Durda IV 12 Mar '09

12 Mar '09

I have recently noticed that despite having a $record_route_nated = true; in the route[PSTN_FORWARD] of ser.cfg, the INVITES and such sent on do not have a Record-Route header. In our NATed situation, probably the only thing this would cause problems with would be when the called party hangs up first, SER and the calling party never find out about it, since the calling party IP is not reachable from the PSTN switch NATed address space, so the single BYE message sent by the PSTN switch probably whizzes by to an unreachable destination address. So the calling party remains off-hook and just gets dead air. When the calling party finally disconnects, the SER passes the caller-initiated BYE to the PSTN switch which returns a 481 because it thinks the call was over ages earlier. Or at least this is what appears to be happening. Here is the PSTN_FORWARD in use, not that different from the PSTN gateway example ser.cfg: route[PSTN_FORWARD] { # Set where packets should be sent. If you don't do this here, # OPTIONS and other messages will be transmitted back to the # outside interface and processed all over again, and you eventually # get a Too Many Hops error on every OPTIONS message. # Code to make this point in multiple directions would be needed # to implement two-way origination. rewritehost("10.x.x.x"); #Send to SIP-TDM switch if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); } else if (method == "INVITE") { # Force record routing if either party is behind NAT $record_route_nated = true; # log(1, "NAT Mangle Force RTP Proxy\n"); force_rtp_proxy("e"); } t_on_reply("PSTN_REPLY"); # if this is called from the failure route we need to open a new branch if (isflagset(FLAG_FAILUREROUTE)) { append_branch(); } # if this is an initial INVITE (without a To-tag) we might try another # (forwarding or voicemail) target after receiving an error if (method=="INVITE" && @to.tag=="") { t_on_failure("FAILURE_ROUTE"); } # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); } drop; } The resulting INVITE that is being sent on to the PSTN switch looks like: INVITE sip:+1xxxxxxxxxx@10.x.x.x:5060;npdi=yes SIP/2.0 Via: SIP/2.0/UDP 10.x.x.x;branch=z9hG4bK6804.1cb4cb23.0 Via: SIP/2.0/UDP 10.x.x.x:5060;branch=z9hG4bK1kpu0b20e0fgogk9c6d0.1 f: <sip:+1xxxxxxxxxx;jip=xxxxxx@10.x.x.x:5060;user=phone>;tag=d0a-13c4-83e003-64bd22fa-83e003 t: <sip:+1xxxxxxxxxx@10.x.x.x:5060;user=phone> i: a249e890d8184d0a13c483e0033230d1ff3b57654c1d4018-0143-8166 CSeq: 1 INVITE User-agent: CS2000_NGSS/9.0 P-Asserted-Identity: <sip:+1xxxxxxxxxx;jip=xxxxxx@10.x.x.x;user=phone> Max-Forwards: 16 k: 100rel Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK m: <sip:10.x.x.x:5060;transport=udp> c: application/SDP l: 231 v=0 o=- 15109843 15109843 IN IP4 10.x.x.x s=- c=IN IP4 10.x.x.x t=0 0 m=audio 44464 RTP/AVP 0 101 b=AS:80 a=ptime:20 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=nortpproxy:yes So, any ideas as to why the "$record_route_nated = true" doesn't seem to be having any effect, eg no Record-Route shows up in outbound messages? I know SER is passing through the above code because I can uncomment the "NAT Mangle" comment and see it in the logs. Is there perhaps some module that must be present that we may have eliminated that makes Record-Route get emitted? Thanks again!

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[Kamailio-Users] DID assignment & routing
by Robert McGilvray 12 Mar '09

12 Mar '09

I'm deploying Kamailio in an enterprise environment and I'm looking for a better way to handle the mapping of external DIDs to SIP usernames, number re-writing etc. I'm currently using ENUM in my test setup but I'd prefer to go with SQL storage and have that data written out to a faster database for lookups by Kamailio. I have a mix of users that register directly to the proxy and also several PBXs that I need to route blocks of numbers to through Cisco routers. For location A I would need to route any four digit number that starts with a 3 (3759), EXCEPT ones specifically defined in my database since they might be mapped directly to a user. Also for any outbound call (9 prefix) I need to check the DB to see if it's local. If I move to DB storage how do I handle things like parallel/sequential forking? ENUM makes it very easy - for example: 1.0.0.0.1 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:8700@172.20.152.132!" . NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:901191986701####@172.20.152.132!" . NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:901191996700####@172.20.152.132!" . If I have a SIP user "bob(a)domain.com" a simple location lookup works for that, but I also need to be able to map a e.164 did to him, and a short form with just the last 4 digits for my internal dialing. I would need a database schema that does the equivalent to this: 0.1.5.3.3.9.2.4.1.9.1 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:bobm@domain.com!" . 0.1.5.3 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:bobm@domain.com!" . Any suggestions? Thanks Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. (W1)

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[Kamailio-Users] SQLOps Module
by Andreas Heise 12 Mar '09

12 Mar '09

Hi, first of all thanks for the great sqlops module. Is "|sql_result_free(result)|" needed before I leave the script by "exit;" to avoid memory trouble or is it only needed to query again in the same request processing? Is there any way to check if the query was not successful (db unreachable, systax error in query)? I've seen a warning in the syslog, but I need to check it in the script. BTW in the doc the example for 1.4.2. seems to be wrong? regards, Andreas

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[Kamailio-Users] Does Kamailio can be used as ptt server?
by doolin wu 12 Mar '09

12 Mar '09

Hi, I'm trying to find a open source ptt (push to talk) server? Does Kamailio can act as this role? Or do you know any open source ptt server? Thanks! -- Steven S.W

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[Kamailio-Users] Siremis: login problems
by Luntras Loredana 12 Mar '09

12 Mar '09

Helo, I've installes Siiremis and I have the following error when I try to log in. "An exception occurred while executing this script: Error message: #256, No Database information found in the config file. Script name and line number of error: /var/www/siremis-0.9.1/openbiz/bin/Configuration.php:42" I have only change the db_name in Config.xml from openser_siremis to openser. (the only data base that I have is openser that I've installed with opensedbctl create). Can anybody help me, please? Thanks, Loredana

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[Kamailio-Users] kamailio start with error Access denied for user..
by svenp 12 Mar '09

12 Mar '09

hello, ich have installed the latest version from kamailio kamailio-1.5.0-tls_src.tar.gz Compiled and installed with this parameters NICER=1 make all make include_modules="db_mysql" all make install the database tables created with /usr/local/sbin/kamdbctl create and then start kamailio with kamctl start the output on cmd is : gateway2:/usr/local/sbin # kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed my logfile: ... Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing record_route, line 310 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing is_method, line 313 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:parse_methods: methods 0x1 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:textops:fixup_method: using id for method [INVITE/1] Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing append_hf, line 320 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing is_method, line 334 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:parse_methods: methods 0x2200 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:textops:fixup_method: using id for methods [PUBLISH,SUBSCRIBE/8704] Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing is_method, line 337 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:parse_methods: methods 0x80 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:textops:fixup_method: using id for method [REGISTER/128] Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing save, line 339 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:db_do_init: connection 0x654220 not found in pool Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:db_mysql:db_mysql_new_connection: opening connection: mysql://xxxx:xxxx@localhost/openser Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:db_mysql:db_mysql_new_connection: driver error: Access denied for user 'openser'@'localhost' (using password: YES) Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:core:db_do_init: could not add connection to the pool Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:usrloc:register_udomain: failed to open database connection Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:registrar:domain_fixup: failed to register domain Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 339 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: CRITICAL:core:fix_expr: fix_actions error Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:core:main: failed to fix configuration with err code -1 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:destroy_modules: shutting down module auth_db Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:destroy_modules: shutting down module auth Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:destroy_modules: shutting down module acc .... username and pw ist default. i try it on console and login is ok: gateway2:/usr/local/sbin # mysql -uopenser -h localhost -popenser Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5894 to server version: 5.0.26 Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql> use openser Database changed mysql> Bye what is wrong in my config? -- View this message in context: http://www.nabble.com/kamailio-start-with-error-Access-denied-for-user..-tp… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] How terminate inbound pstn calls
by carl Lougher 12 Mar '09

12 Mar '09

Howdy, Can anyone help me configure my .cfg file to point inbound calls from the pstn gateway to my Trixbox on a sip trunk to Kamailio? Outbound calls work fine and trunk is registered in local DB. Cheers, Taff...

1 0

[Kamailio-Users] Modifying the URI in the header From more than once
by mayamatakeshi 12 Mar '09

12 Mar '09

Hello, I'm trying to add CallForwarding to my kamailio.cfg. Before relaying an INVITE from one subscriber to another, I'm required to modify the From username to the PSTN number of the caller. So I use uac_replace_from and this works fine. I can see kamailio sendind this record-route: Record-Route: <sip:192.168.33.2;lr=on;ftag=qunug47bd1hc74ir26iv;vsf=AAAAAERUW1NLXloEeFVZKRYIDAZKBwRPEQYEH1oLCQAEFS5qcA--;did=b76.5c17ed25> But then, when the callee fails to answer and failure_route is called, I need to modify the From username to the PSTN number of the callee before doing the forward. If I simply try to call uac_replace_from again inside failure_route, record-route is sent with a duplicate vsf parameter like this: Record-Route: <sip:192.168.33.2;lr=on;ftag=qunug47bd1hc74ir26iv;vsf=AAAAAERUW1NLXloEeFVZKRYIDAZKBwRPEQYEH1oLCQAEFS5qcA--;did=b76.5c17ed25;vsf=AAAAAERUW1NLXloEeFVeKRYIDAZKBwRPEQYEH1oLCQAEFS5qcA--> And this causes two problems: - the message becomes too big and the GW replies with "513 Message too large" - the reply is sent back to the caller with duplicate From header and some UAs do not acknowledge the reply. So, is it possible to modify the From header more than once? regards, takeshi

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[Kamailio-Users] PIKE module
by Juan Perez 12 Mar '09

12 Mar '09

Hello I need some clarification here; the Help file for this module says this: ///////////////////////////////////////////////////////////////////////////////////////////////////////////////////// The module keeps trace of all (or selected ones) incoming request's IP source and blocks the ones that exceeded some limit. Works simultaneous for IPv4 and IPv6 addresses. The module does not implement any actions on blocking - it just simply reports that there is a high traffic from an IP; what to do, is the administator decision (via scripting). ///////////////////////////////////////////////////////////////////////////////////////////////////////////////////// I understand from the first part that it blocks the IPs, but last sentence says otherwise. How is it supposed to work? txs jp

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sr-users March 2009