sr-users September 2013

sr-users@lists.kamailio.org

89 participants
127 discussions

Registering "upstream" with the uac module, but with authentication

by Steve Davies

Hi, I'm making slow but steady progress with my Kamailio project. My next task: I need to "relay" a registration to an upstream service. IE when one or more devices registers against my local registration service, I need to initiate a corresponding registration upstream. If all my local devices unregister, all all the registrations lapse, I want to drop the upstream registration. There is an example a bit like this in the Asterisk integration example. However in that case the Asterisk doesn't challenge for authentication. My upstream does. I can't find anything in the uac module docs about authenticating. Is there any way to handle this. One thing I did try was to create "uacreg" as a view, and use that as the source of upstream registrations: CREATE VIEW `uacreg` AS select distinct `s`.`id` AS `id`,`l`.`username` AS `l_uuid`,`l`.`username` AS `l_username`,`s`.`domain` AS `l_domain`,`l`.`username` AS `r_username`,`s`.`domain` AS `r_domain`,`s`.`domain` AS `realm`,`l`.`username` AS `auth_username`,`s`.`password` AS `auth_password`,concat('sip:',`s`.`domain`,':5060') AS `auth_proxy`,timestampdiff(SECOND,now(),`l`.`expires`) AS `expires` from (`location` `l` join `subscriber` `s`) where (`l`.`username` = `s`.`username`) This kinda worked - but didn't quite track the registration status. Do you think this idea has promise? Maybe something similar that works using triggers rather than a straight view might work better? Thanks, Steve -- Steve Davies: Technical Director, Connection Telecom (Pty) Ltd

11 years, 1 month

Calling users on another kamailio server

by Brian Wallen

I currently have two independent kamailio servers. I'd like to set them up in a way that user1 on server1 can make a call to user2 on server2. After searching I've come up with two ways that this might be able to be done. Can someone please sanity check these or point me in the right direction? 1. Have one registrar server and convert the other server to a proxy 2. Keep them both as registrars and somehow make them each aware of the users on the other server I like 2 better because if one server went down users on the other server would still be up. The only thing is I don't know how to set the servers up to communicate with each other.

11 years, 1 month

kamailio presence and XCAP related issue

by Rupayan Dutta

Dear All, .May be I couldn't express my doubt properly ijn my earlier mails.Actually we were testing OfficeSIP server with SipML5 client for multiparty videoconferencing with presence enabled for showing buddy list and it is working properly.Now as we moved to kamailio with same client and configuring kamailio with Websocket and presence support, I have noticed that the presentity and xcap tables are not populated but location,watchers and active watchers tables are showing correct entries after a user logs in using sip over websocket from SIPML5. I have followed the steps as described in " http://kb.asipto.com/kamailio:presence:k31-made-simple". As Sipml5 sends PUBLISH/NOTIFY request in pidf format is this is the possible cause?As I am a person from java based web application technology it is really hard for me to detect the root cause.Please give me some idea regarding this issue.

11 years, 1 month

Asterisk and Kamailio Integration CAPACITY ISSUE

by Hassan Wajahat

Hi, I followed your tutorial and the setup is running seamlessly. But the question is that I read somewhere that asterisk can make up to 300 to 400 concurrent sip calls. Higher than that will cause problems. Will that issue appear here as well since you are using asterisk in the sip calling process?? Regards, Hassan Wajahat

11 years, 1 month

Kamailio realtime fails to forward register with Asterisk.

by imperium broadcast

Hi all I have been trying to figure out where Im going wrong with following this guide: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb The only thing that is different is I am using Asterisk 1.8.22.0 Kamailio works fine and I am able to register a phone to it but the forward register fails to register with Asterisk. Its like Asterisk doesn't see the request even running sip debug on the Asterisk Console it doesn't show the register attempts. I have to use ngrep to see any thing. Realtime works fine when I register the phone on Asterisk. Although I can get round the register using AGI I get the same issue when passing a call to asterisk, Asterisk doesn't see the invite. Am I missing something so obvious :-/ This is what I have in sip.conf "IP change to protect the innocent!" [kamailio-2] type=peer host=1.1.1.1 fromdomain=1.1.1.1 context=outgoing insecure=invite,port directmedia=nonat qualify=yes disallow=all allow=all and this is the sip trace for the register U 2013/09/13 16:54:50.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000(a)127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: <sip:102@172.16.0.112:5060>. Expires: 120. . U 2013/09/13 16:54:54.697484 1.1.1.1:5060 -> 172.16.0.110:5080 REGISTER sip:172.16.0.110:5080 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKd5d1.80da8915.0. To: sip:102@172.16.0.110. From: sip:102@172.16.0.110;tag=533cb9e91f4b999cf76861cbb9ed54ed-65e5. CSeq: 10 REGISTER. Call-ID: 4670378d1185f5b6-25000(a)127.0.0.1. Content-Length: 0. User-Agent: kamailio (3.3.1 (x86_64/linux)). Contact: <sip:102@172.16.0.112:5060>. Expires: 120. and I have this for my Asterisk binds in kamamilio.cfg #!ifdef WITH_ASTERISK asterisk.bindip = "172.16.0.110" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "172.16.0.112" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif -- Regards Mick

11 years, 1 month

issues with dispatcher rerouting to next server when getting a 500 from freeswitch

by Joseph Watson

Hello Everyone, I am really having an issue getting kamailio to try another dispatch route when I get a 500 back from one of my freeswitch servers. Currently I have a list of freeswitch servers located in my dispatch.list file. If one of then is to busy it will respond with a 500 Maximum Call in Progress. I am trying to get Kamailio to go to the next route in the dispatch.list when it get one of these 503 and if it should exhaust the whole list respond with a 503. Here is my kamailio config ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=4 log_stderror=yes #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=10 /* comment the next line to enable TCP */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:eth0:5060 sip_warning=no ####### Modules Section ######## #set module path mpath="/usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("xlog", "buf_size", 8192) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) modparam("tm", "remap_503_500", 0) # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") #modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") #modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") #modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") #modparam("dispatcher", "ds_append_branch", 1) ####### Routing Logic ######## # main request routing logic route { append_hf("X-VSIP-CUST-IP: $si\r\n"); route(DISPATCH); route(RELAY); } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } # xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); return; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("DISPATCH"); route(RELAY); exit; } } } route[RELAY] { if (!t_relay()) { send_reply("503", "LB01 - All Circuits Are Busy"); # sl_reply_error(); } exit; } I am very new to kamailio and still learning my way around. Any help you can provide with be fantastic! Thank you, Joe Watson

11 years, 1 month

json module / quotes in result

by Florian Lohoff

Hi, is it correct that the json_get_field returns the quotes from the json document? Example: { "foo": "bar" } json_get_field("$var(json)", "foo", "$var(result)"); result will contain "bar" with quotes ... I was a little astonished to see that. Flo -- Florian Lohoff f(a)zz.de

11 years, 1 month

New Developer: Charles Chance

by Daniel-Constantin Mierla

Hello, I want to announce that a new person got developer GIT write access to repository: Charles Chance. He is for long time in the community, sending patches in the past to modules such as memcache. His immediate goal is to care of dmq module, having a set of patches to be committed as well as plans to integrate it within more modules. His git commit id is: cchance My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013 - more details about Kamailio trainings at http://www.asipto.com -

11 years, 1 month

NAT function

by Kethzer Docteur

I need help to fixed my NAT configuration I have nat module load but nat configuration is not working. ANY HELP PLease #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" loadmodule "nat_traversal.so" #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@67.215.8.130") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } -- Kethzer Docteur

11 years, 1 month

Pseudo-variables and branches

by Steve Davies

Hi - Can someone help me understand how pseudo-variables interact with branches? I've written this logic in my failure route to do a reroute of calls if the primary destination fails: if ( is_method("INVITE") && !t_check_status("407") && ($rd != "127.0.0.1") ) { # Maybe should also check its not just a Busy or similar xlog("L_NOTICE","SLD: INVITE failed $rm ($mt / $rs $rr): $ci::$ft; try via asterisk instead\n"); append_branch("sip:$rU@127.0.0.1:5070"); send_reply("181", "Rerouting call via local failover"); $avp(senttoast) = 1; if (!t_relay()) { sl_reply_error(); # or reply_error ? } I'm trying to check if the call was already rerouted - by looking at $rd. But in fact I find that $rd keeps the domain from the ORIGINAL invite even if the call that just failed was the one to 127.0.0.1:5070 . So I'm obviously missing something about how pseudo-variables interact with the branching etc. Thanks, Steve -- Steve Davies: Technical Director, Connection Telecom (Pty) Ltd Email is preferred, but: Phone: 0878200200

11 years, 1 month

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sr-users September 2013