sr-users July 2011

sr-users@lists.kamailio.org

75 participants
119 discussions

Loading multiple htables from db on startup? (Kamailio 1.5)

by Robin Rodriguez

Hi all, Currently running on 1.5.4-notls and using a single htable that gets loaded at startup ala modparam("htable", "htable", "local_routes=>size=4;autoexpire=0;dbtable=TABLE_A;") modparam("htable", "key_name_column", "NAME_1") modparam("htable", "key_value_column", "VALUE_1") modparam("htable", "fetch_rows", 512) This works great, the problem I've run into is attempting to load a second htable on startup whose data comes from a different table/scheme ala modparam("htable", "htable", "inbound_acl=>size=4;autoexpire=0;dbtable=TABLE_B;") modparam("htable", "key_name_column", "NAME_2") modparam("htable", "key_value_column", "VALUE_2") modparam("htable", "fetch_rows", 512) With both of them in the config file Kamailio fails to start with the error INFO:core:init_mod: initializing module htable ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'NAME_2' in 'field list' 'NAME_2' does not exist in 'TABLE_1' so that seems to suggest that only the first dbtable value is used, the SQL side supports this: select NAME_2,key_type,value_type,VALUE_2 from TABLE_A order by key_type So it uses the first defined database table, but then the last defined key_name and key_value. Is there any way that I've overlooked in the docs or elsewhere about about loading multiple htables from different database tables? I'd like to use this htable in an ACL and would like to avoid database hits though can make it work with avpops if I have to. Any suggestions are greatly appreciated. Thanks, Robin

13 years, 4 months

Dialplan question

by alex pappas

Dear all, I have these 3 entries in the dialplan: dpid pr match_exp attr 2 50 ^38.+ porvider_A 2 50 ^3802.+ porvider_B 2 50 ^3806.+ porvider_C with the fifo command : *kamctl fifo dp_translate 2 3806328881* Output:: ATTRIBUTES::* porvider_A* I would expect that the dialplan module would choose the third choice which is the correct one for this call. >From the dialplan documentation i understand that the priority is needed when the prefixes are overlaping in terms of they are the same but with different attrs like : 2 50 ^38.+ porvider_A 2 50 ^3802.+ porvider_B *2 50 ^3806.+ porvider_C* *2 50 ^3806.+ porvider_D* * * here i understand that I should use priorities. Does anyone knnows how this works ?? Thank you in advance Alex * *

13 years, 4 months

acc table versions

by Klaus Darilion

Hi! Just as a note: I find it a bit strange that acc (4) and missed_calls (3) have different version numbers although they are identical. Anyway, it seems that acc module does not make version checks at all. regards Klaus

13 years, 4 months

Celebrating 10 years of SER

by Daniel-Constantin Mierla

Hello, the 3rd of September will mark the 10 years since the start of SIP Express Router (aka SER - also the origin for kamailio (openser) and now the two are back together). From last year: * http://sip-router.org/wiki/history/ser-9-years This is a good time to celebrate and I am thinking of at least a social event (e.g., dinner) in Berlin, Germany. Based of private discussion so far, other cities are interested in organizing something similar locally, e.g., Barcelona. If you are a group of people that can gather together for a drink in same place to celebrate the event, let me know and I will add in the list to coordinate - hopefully we will be even able connect between sites with some video/audio if the timezones are convenient. I'm looking as well to do a more formal event, something like a day long conference style, on Friday, 2nd of September. Berlin is the natural choice for location, being the place where the project started. I can take care of location when it is clear that there are people willing to travel. Berlin area alone should be able to get about 20 people, among them several long term developers of the project and companies offering SER/Kamailio-based services. Coming over gives you the opportunity to connect directly with them. At this event, along with presenting shortly the history and future plans, the event should be open to everyone that want to present their usage of SER/Kamailio. Attendance is going to be free, in the limit of available seats. If you think it would be interesting and want to attend, let me know. Also, if you have other ideas of how to celebrate, write us back. Best regards, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

13 years, 4 months

Re: [SR-Users] Playing with dlg_bridge()

by Stefan Broddesson

Brodde(a)yahoo.com Sent from my HTC ----- Reply message ----- From: "Roberto Fichera" <kernel(a)tekno-soft.it> To: "Henning Westerholt" <henning.westerholt(a)1und1.de> Cc: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> Subject: [SR-Users] Playing with dlg_bridge() Date: Thu, Jul 7, 2011 16:01 On 07/07/2011 02:51 PM, Henning Westerholt wrote: > On Thursday 07 July 2011, Roberto Fichera wrote: >>>> Do you know what does means the warning below? >>>> [..] >>>> 68.2.50 Jul 6 11:54:17 roberto /usr/sbin/kamailio[20457]: WARNING: >>>> dialog [dlg_handlers.c:884]: unable to find dialog for NOTIFY with >>>> route param '7d2.14447fc4' [727:1291273281] >>> This means that the module can't find an existing (stored) dialog for the >>> routed message in question, in this case a NOTIFY. >> Does this depends by the fact of own the dlg_bridge() works? And in case, >> how can I solve that problem? > Hi Roberto, Hi Henning, > I'm not 100% sure, I'm not the expert for dlg_bridge.. To be more specific > (after discussion with Timo..) - the warning is emmited if the module can't > find the dialog with the ID added as a route parameter to the respective > message. One particiular reason is (if you don't store the dialogs in a > database) a simple kamailio restart, then all existing dialog information is > lost. actually I've the following setup: #!define DBURL "postgres://openser:openserrw@localhost/openser" modparam("dialog", "dlg_flag", 2) modparam("dialog", "bridge_controller", "sip:controller@192.168.2.92") modparam("dialog", "db_url", DBURL ) Actually, my thought is that problem is more strictly related to the dlg_bridge() which might not track the bridged dialog. Anyway, I'm playing with dlg_bridge() because I want to realize the logic so that two peers, and if possible more, can talk together only if authorized. The authorization might be a unique tag or transaction id, stored in a database, and valid only for the give conversation {tag, caller, callee} once started, than the tag become invalid and can't be used again. I really like to have some suggestions about how to realize such logic, so which might be the best way to approach such problem since I'm actually a novice to Kamailio or in general with OpenSer logic. I'm actually more confident with Asterisk logic. > Best regards, > > Henning > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

13 years, 4 months

force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.

by MingHon

Hi List, im facing an issue that my kamailio proxy did not replace the ip address in the invite and 200OK sdp body. my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user my kamailio is listening on 192.168.1.3, also define: advertised_address="175.136.223.112"; & advertised_port=5060; and my asterisk is on 192.168.1.23. sip signalling and rtp port forwarded to kamailio. uacs from another nat register successfully. if i put 2 lines of force_rtp_proxy("fcow","175.136.223.112"); i will get double ip addr in c and o but kamailio ignore my ip addr. example i will get c=IN IP4 192.168.1.3192.168.1.3 here is part of my simple script. hope you can help. thank you very much. ---------------cfg------------------- route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy("fcow","175.136.223.112"); #force_rtp_proxy("fcow","175.136.223.112"); xlog("L_INFO","offer"); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } -------------------------------------- and here is the wireshark for uac INVITE and OK. -----------INVITE----------------- ve0 EE;p9INVITE sip:102@192.168.2.132:5062 SIP/2.0 Record-Route: <sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0 Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080 Max-Forwards: 69 From: "101" <sip:102@aextddns.dyndns.info>;tag=as032358a3 To: <sip:102@192.168.1.3:5060> Contact: <sip:102@192.168.1.23:5080> Call-ID: 416f6e09674ae9671bb7144a1cb11137(a)aextddns.dyndns.info CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.18 Date: Tue, 05 Jul 2011 07:20:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 327 v=0 o=root 1639709788 1639709788 IN IP4 192.168.1.3 s=Asterisk PBX 1.6.2.18 c=IN IP4 192.168.1.3 t=0 0 m=audio 10072 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes -----------200OK--------------- e90 ElE;pX4tSIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.200:5062 ;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416 Record-Route: <sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes> From: "101" <sip:101@aextddns.dyndns.info>;tag=1796959074 To: <sip:102@aextddns.dyndns.info>;tag=as2e4c0125 Call-ID: 1985782590(a)192.168.2.200 CSeq: 21 INVITE Server: Asterisk PBX 1.6.2.18 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:102@192.168.1.23:5080> Content-Type: application/sdp Content-Length: 286 v=0 o=root 403900934 403900934 IN IP4 192.168.1.23 s=Asterisk PBX 1.6.2.18 c=IN IP4 192.168.1.23 t=0 0 m=audio 14420 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------ My kamailio log. -----------LOG------------------ DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3 INFO: <script>: offer ------------------------------------- double force_rtp_proxy --------kamailio -> asterisk [INVITE]--------- Pyi-}E7V@:#pINVITE sip:102@aextddns.dyndns.info SIP/2.0 Record-Route: <sip:192.168.1.3;lr=on;ftag=640933430;nat=yes> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0 Via: SIP/2.0/UDP 192.168.2.200:5062 ;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648 From: "101" <sip:101@aextddns.dyndns.info>;tag=640933430 To: <sip:102@aextddns.dyndns.info> Call-ID: 1909950509(a)192.168.2.200 CSeq: 21 INVITE Contact: <sip:101@175.138.21.31:2788> Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 69 User-Agent: T20 9.41.0.80 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 334 v=0 o=20073 20073 IN IP4 192.168.1.3192.168.1.3 s=SDP data c=IN IP4 192.168.1.3192.168.1.3 t=0 0 m=audio 1006410064 RTP/AVP 0 8 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=nortpproxy:yes a=nortpproxy:yes -----------LOG------------------ DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3 DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3 INFO: <script>: offer -----------LOG------------------ -- Regards, MingHon

13 years, 4 months

Playing with dlg_bridge()

by Roberto Fichera

Hi All in the list, I want to implement a click and dial feature using the xhttp interface but also keeping working the XCAP functionality. Than I added some code about the dialog module (it's my first time I try it) so that the URL look like http://localhost:5060/sip-bridge/<from>/<to> #!ifdef WITH_XCAPSRV #!define WITH_XHTTPAUTH event_route[xhttp:request] { xdbg("===== xhttp: request [$rv] $rm => $hu\n"); if($hu=~"^/xcap-root/") { #!ifdef WITH_XHTTPAUTH if (!www_authorize("xcap", "subscriber")) { www_challenge("xcap", "0"); exit; } #!endif set_reply_close(); set_reply_no_connect(); # xcap ops $xcapuri(u=>data) = $hu; if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:" + $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; xlog("===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); if($xcapuri(u=>auid)=="xcap-caps") { $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "ok", "application/xcap-caps+xml", "$var(xbody)"); exit; } #!ifdef WITH_XHTTPAUTH # be sure auth user access only its documents if ($au!=$(var(uri){uri.user})) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>$si:$sp</body></html>"); exit; } #!endif switch($rm) { case "PUT": xcaps_put("$var(uri)", "$hu", "$rb"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } exit; break; case "GET": xlog("===== xhttp: get $var(uri) => $hu\n"); xcaps_get("$var(uri)", "$hu"); exit; break; case "DELETE": xcaps_del("$var(uri)", "$hu"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp del: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } exit; break; } } else if($hu=~"^/sip-bridge/") { if ( ! ( src_ip == 127.0.0.1 ) ) { xhttp_reply( "403", "Forbidden", "text/html", "<html><body>Invalid Access</body></html>" ); exit; } $var(uri) = $hu; $var(from) = $(var(uri){s.select,2,/}); $var(to) = $(var(uri){s.select,3,/}); xdbg( "=== GOT sip-bridge between $var(caller) => $var(callee) ===" ); *dlg_bridge( $var(from), $var(to), "sip:192.168.2.92:5060" );* } # http ops xhttp_reply("200", "ok", "text/html", "<html><body>OK: $si:$sp</body></html>"); exit; } #!endif but I got the error below: Not starting : invalid configuration file! 0(22256) : <core> [cfg.y:3409]: parse error in config file //etc/kamailio/kamailio.cfg, line 986, column 17-26: function dlg_bridge: parameter 1 is not constant 0(22256) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 986, column 63: bad command ERROR: bad config file (2 errors) so my question is how can I call the dlg_bridge() passing some parameters coming from the http request? Thanks in advance. Roberto Fichera.

13 years, 4 months

Problem with worst case scenario of accessing 15 gateways over the DISPATCHER module - MAX_BRANCHES is reached

by Klaus Feichtinger

Hi, I have a problem in my scenario where I use the dispatcher module for (round-robin based) distribution of incoming calls to 15 (!) different gateways. In idle case (= all gateways are up and running or just a couple of them does not accept calls) my solution is working fine. The solution is very primitive and based on the example of the dispatcher module's ReadMe. However, when e.g. 12 of 15 gateways are out of service (in detail: rejecting the call with "480 - Temporarily Unavailable") and only the 13th would be okay, I can not reach that gateway. This is because the TM module is interrupting the distribution with an error message indicating that the maximum number of branches (per transaction) is reached (ERROR:/tm/:/add_uac/: /maximum number of branches/ exceeded).... My first idea for solving this problem was/is in manipulating the definition of the "max_branches" constant in the config.h file. However, I prefer using the precompiled Debian packages. Therefore that solution is not optimal. I have not found a method for decreasing the destination_set on an easy way... Does anybody know an alternative solution / workaround? Thanks in advance, Klaus

13 years, 4 months

new to ser

by Pradeep Sharma

Hi, I am installed ser 0.9.6 onto my mac machine and with following command i get it to listen to 5060 port. but when i use a sip client and send register message i get a 501 not implemented as a response from the server. i am attaching ser.cfg. your help is appreciated. -- Thanks Paul

13 years, 4 months

updates to config preprocessor

by Daniel-Constantin Mierla

Hello, just to let you know about some recent additions regarding the config file preprocessor, hopefully they will help you keep the config cleaner and easy to maintain. - you can use either #! or !! as the start marker of preprocessor directive (to make happy those complaining about not able to distinguish quickly from comments) - subsdef was added some time ago to allow defining a token that will replace ID tokens as well as update inside string values. It has the same format as #!subst. For example:| #!substdef "!ABC!XYZ!g"| is pretty much same as: #!define ABC XYZ #!subs "/ABC/XYZ/" - substdefs, like substdef byt the defined value will be added enclosed in double quotes: |#!substdef "!ABC!XYZ!g"| is pretty much same as: #!define ABC "XYZ" #!subs "/ABC/XYZ/" - trydef - similar to define, but if the ID is defined already, ignore this time - redef - similar to define, but if the ID is defined, redefine its value (note that define itself throws error if the ID is already defined) Regarding the inclusion of files, a new option is available, 'import_file'. It is similar to 'include_file', but if the file is not found, it does not throw error. Bot include_file and import_file can be started as well with #! or !!. You can still use them without #! or !!, whatever you like more. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

13 years, 4 months

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sr-users July 2011