sr-users April 2016

sr-users@lists.kamailio.org

76 participants
114 discussions

dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)
by TEG AMJG 04 Apr '16

04 Apr '16

Hi I am new to Kamailio and i am just trying to setup a kamailio (v4.2.3) dispatcher to distribute calls to 2 asterisk using the call load distribution algorithm (alg 10). I started to test and see how all it is working and in the logs appears that the call load record it cannot be found. - ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (call id generated by asterisk) I am getting this error trying a 2 call leg conversation when asterisk is creating the BYE message destinated to the callee after receiving a BYE message from the caller, and the BYE message by Asterisk reach Kamailio to then apply the ds_load_update function >From what i understand about ds_load_update it is that it sets the load to an specific destination if Kamailio receives a 2xx reply of an INVITE message, and removes it if Kamailio receives a BYE message based from the Call-id received in that message, where it is actually failing in my implementation at the moment Here are the blocks of code of the dispatcher parameters and where the function ds_load_update() is being used: #!ifdef WITH_LOADBALANCE #Parameters using Dispatcher from DB # ---------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") # ---------- #Enabling failover mechanism # ----------- modparam("dispatcher", "flags", 2) # ----------- #Setting dispatcher parameters in AVP # ----------- modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") # ----------- #Setting dispatcher result in PV variables # ----------- modparam("dispatcher", "attrs_pvname", "$var(attrs)") # ----------- #Enabling dispatcher algorithm according to call loads # ----------- modparam("dispatcher", "ds_hash_size",8) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) # ----------- # Setting parameters for probing dispatcher destinations # ----------- modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_ping_from", "sip:kamailio1@<Kamailio IP>") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshhold", 2) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404") # ----------- ... route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing #!ifdef WITH_LOADBALANCE if(is_method("BYE|CANCEL")){ xlog("L_ERROR","UBICAR BYE ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs\n"); ds_load_update(); } #!endif ... } ... onreply_route[MANAGE_REPLY] { #!ifdef WITH_LOADBALANCE if(is_method("INVITE")) { if(status=~"2[0-9][0-9]") { xlog("L_ERROR","UBICAR setting ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs"); ds_load_update(); }else if(status=~"[3-7][0-9][0-9]") { ds_load_unset(); } } #!endif if(status=~"[12][0-9][0-9]"){ route(NATMANAGE); } } This is what i am getting from logs after the 2 call legs conversation completes: kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR: FROMASTERISK 70175143146f92975a5bff8876435911(a)192.168.10.194 kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=sip:103@192.168.10.194 method=INVITE callid=28d0f6e71625c9e87baec07f46167863(a)192.168.10.194 to= sip:104@192.168.10.194 reply=2 rrcode=OK rcode=200 kam1 /usr/sbin/kamailio[4083]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=sip:103@192.168.10.194:5060 method=INVITE callid=1-3692(a)192.168.20.2 to=sip:104@192.168.10.194:5060 reply=2 rrcode=OK rcode=200 kam1 /usr/sbin/kamailio[4084]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.10.190:5060 from=sip:103@192.168.30.1:5060 method=BYE callid=1-3692(a)192.168.20.2 to=sip:104@192.168.30.1:5060 reply=1 rrcode=<null> rcode=<null> kam1 /usr/sbin/kamailio[4085]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.30.2:22523;transport=UDP from= sip:103@192.168.10.194 method=BYE callid= 28d0f6e71625c9e87baec07f46167863(a)192.168.10.194 to=sip:104@192.168.10.194 reply=1 rrcode=<null> rcode=<null> kam1 /usr/sbin/kamailio[4085]: ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for ( 28d0f6e71625c9e87baec07f46167863(a)192.168.10.194) Kamailio IP: 192.168.10.194 Asterisk IP: +----+-------+-------------------------+-------+----------+-------------------------------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+-------------------------+-------+----------+-------------------------------+-------------+ | 1 | 0 | sip:192.168.10.190:5060 | 2 | 0 | duid=PBX1;my=pbx1;maxload=200 | pbx1 | | 2 | 0 | sip:192.168.10.191:5060 | 2 | 0 | duid=PBX2;my=pbx2;maxload=200 | pbx2 | +----+-------+-------------------------+-------+----------+-------------------------------+-------------+

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get_redirects with t_next_contacts
by Marrold 04 Apr '16

04 Apr '16

Hi all, I am having some issues using get_redirects() from the UAC_Redirect module in conjunction with t_next_contacts from the TM Module. I've searched Google and the list but the only references I can find are several years old. I am attempting to configure Kamailio so it if receives a 3XX redirect message, it will store the Contacts it contains, and attempt them in 'Q' parameter order. So for the Contact below- Contact: <sip:10.0.97.21:5060>;q=0.2,<sip:5.29.13.139:5060>;q=0.5. I want Kamailio to try 5.29.13.139, and if that times out, or is rejected in someway, try 10.0.97.21. If I use t_relay rather than t_next_contacts, kamailio branches the calls and relays them in parallel regardless of Q value - this is not what I want. If I use t_next_contacts, it attempts 5.29.13.139 which is rejected with a 401, but it doesn't attempt 5.29.13.139 and responds "408", "Request Timeout MH 2" Relevant script section is below, has anyone been able to get this working? failure_route[302] { if (t_check_status("302")){ xlog("L_ALERT","Failure Route: 302\n$mb"); get_redirects("*"); if (!t_load_contacts()) { send_reply("500", "Server Internal Error - Cannot load contacts MH"); exit; }; if (!t_next_contacts()) { t_reply("408", "Request Timeout MH"); } else { t_on_failure("302"); t_relay(); exit; }; } else if (t_check_status("[456][0-9][0-9]")){ if (!t_next_contacts()) { t_reply("408", "Request Timeout MH 2"); } else { t_on_failure("302"); t_relay(); exit; } } } Thanks, Matthew

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Using 302 redirects with Dispatcher Module
by Marrold 04 Apr '16

04 Apr '16

Hi All, I'd like to use the Dispatcher Module along with 30X redirects, to monitor gateway availability with SIP options and to provide redundancy / fail over. Some sources online suggest it's as easy as - if ( method=="INVITE" ) { dst_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } However when the 30X reply is sent, it contains no contact header. Please could someone advise if it's possible to use the Dispatcher module with 302 redirects, and what they syntax is? Thanks, Matthew

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kamailio number translation and route
by liwei＠neulinx.com 04 Apr '16

04 Apr '16

hi all; I want to use kamailio achieve: First phase: 1) inbound caller number translation; 2) outbound caller number translation; 3) inbound callee number translation; 4) outbound callee number tanslation; Example : Call Flow backend server ------------------> kamailio --------------------------> GateWay ------------------->PSTN caller -->callee caller ---> callee 1000 --> 2000 111000--->222000 inbound outbound caller --->callee caller ---> callee 1000 ---> 2000 11000 ---> 22000 | translation | | translation | 11000 ---> 22000 111000---> 222000 Second phase: / ---->gateway1 ----->pstn / backend server ---> kamailio ----->gateway2 -----> pstn \ \ ----> gateway3 ------->pstn Could you give me some suggest or example configuration？ liwei(a)neulinx.com

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Load balancing traffic based on SIP URI
by NITESH BANSAL 04 Apr '16

04 Apr '16

Hello, I want to use Kamailio to load balance traffic across multiple asterisks.The problem is that my SIP traffic is related to conference, so if a SIP callfor a particular URI is sent to an Asterisk instance, any subsequent calls for thatSIP URI should be sent to the same Asterisk instance.Is there anyway to achieve this in Kamailio? Thanks,Nitesh

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SIP Authentication
by Girish 04 Apr '16

04 Apr '16

Hello Sir, I am a newbie to Kamailio and I am trying my first setup. I would like to use Kamailio to take in traffic using a SIP Trunk and Route the calls to another SIP trunk based on prefix. I have added the incoming SIP trunk IP address to the address table on the MYSQL but I am unable to get a call routed out. Is there any sample configuration for Authentication via IP address and route the incoming SIP INVITE. Regards, Phoenix

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RPID and Auth DB
by Alberto Sagredo 04 Apr '16

04 Apr '16

Hi! I plan to use rpid field but when trying to get from database, subscriber table has rpid field filled, i get always NULL. Is there any more needed than: modparam("auth_db", "load_credentials", "rpid") ? Just a check of variable xlog(" Valor de RPID es : $avp(s:rpid)"); Shows me NULL all times Thanks

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KAMAILIO tcp_send query, connection problem
by Ed Todd 04 Apr '16

04 Apr '16

KAMAILIO tcp_send query, connection problem I have a KAMAILIO SIP server on a static IP from Comcast, using a modem/router Cisco DPC39412B. I have a PC application server which is successfully signing on to the SIP server, via a LAN cable on the same network as the SIP server (Cisco). In my second test, the PC application server is connected to the internet via a ZTE Falcon Z-917. In a third test, the PC application server is connected to the internet via a Talk-Talk fiber broadband. The connection does not occur. >From the log files, the successful connection says: Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (7817 (14)), reusing fd Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f6270b8a4a0 n=565 fd=1 On the failed connection: Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1818]: tcp_send(): tcp_send: no open tcp connection found, opening new one Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 192.168.0.146 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 55133, type 3 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 1776:802:0, 803 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: tls [tls_server.c:184]: tls_complete_init(): Using TLS domain TLSc<default A while later, I get: Apr 1 09:41:37 SIP-Server kamailio[7804]: DEBUG: usrloc [urecord.c:357]: wb_timer(): Binding 'hvs_us_iphone','sip:hvs_us_iphone@172.56.8.182:44341;transport=TLS;line=75dc89bd64369c7' has expired Apr 1 09:41:38 SIP-Server kamailio[7820]: DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7f6270b8a4a0 (ticks=862851529, timeout=862909176 (3602 s), wr_timeout=862851529 (0 s)), write queue: 197 bytes Apr 1 09:41:38 SIP-Server kamailio[7820]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.0.146:55133 failed (timeout) Apr 1 09:41:38 SIP-Server kamailio[7820]: DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7f6270b8a4a Any clues or tips would be greatly appreciated! Ed

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Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 03 Apr '16

03 Apr '16

Sure it is! This is exactly what I was looking for! Actually I wasn't able to see that there was a way to perform this action! Thanks for your help, Olle! I will try to create this, and when I have it done, I will post my script here, so if someone else need the same problem I can help. Now, I have just one more question about my scenario: when I register a user, is there a way to create something like a "custom field" on the location? To explain better: If possible, I can add something like a custom header that says if the user is registered from a phone device or the web interface, then I can perform the loop and replace only the oldest contact using that device... *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > This is my problem... I want to have 2 contacts for AOR, and not only one. > > When the 3rd arrives, it must remove the oldest and continue with 2... > > *If I am using the 0x04 parameter, it will save just one contact, and the > user will not be able to get the Invite both, in the mobile device and web > interface.* > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Hello, *Olle!* >> >> *Thanks for your help in this case, and don't worry about the time >> (actually I wasn't even expecting to get an answer today).* >> >> >> *I know the max_contacts parameter. But the problem is that if I set the >> max_contacts to 2, when the user tries to register again, he gets an error >> message 503. The point is that I want to get the user registered, and >> replace the oldest location entry for the newest one.* >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> Hello all. >>> >>> I have problem here and I tried to find a solution and search over >>> internet, but without success. >>> >>> My scenario is the following: I have an application that must be >>> registered in Kamailio when a user logs in the web interface, so he can get >>> calls (something like a web softphone using webrtc). At this point, we are >>> OK, and everything is working fine. >>> >>> To get these register functions I am using the parameter >>> "save("location", "0x04")" as described in the REGISTRAR module >>> documentation, so if a user logs in another web browser or computer, only >>> the last one will continue registered and all calls will be forwarded to >>> him. >>> >>> The point is that now we are creating a phone app that will do almost >>> the same thing as the web interface, allowing users to receive calls using >>> the mobile device, and here is my problem: I want to allow my users to be >>> registered on two devices at the same time, but if a user logs into another >>> device, I don't want to reply with a 503. I want to allow the user to >>> register again, deregistering the oldest contact. >>> >>> I saw that there is a module named "ims_usrloc_scscf" and on its >>> description it says "implemented overwrite oldest contact behaviour", >>> but I couldn't find any documentation about it. >>> >>> So, is there a way to get this working today? >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> >> >> >

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kamailio proxy
by liwei＠neulinx.com 03 Apr '16

03 Apr '16

hi ， I want to use kamailio as a proxy ; / gateway1 ----->pstn / 1、 call from backend server ---> kamailio ----->gateway2 -----> pstn \ \ gateway3 ------->pstn 2、caller and callee number translation ; Could you please give me some suggest; Wade From China liwei(a)neulinx.com

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sr-users April 2016