- sr-users - kamailio.org

REGISTER request and save() from onreply_route
by Dmitry Sytchev 19 Jan '15

19 Jan '15

Hi all! Seems to be an old topic but still unclean is it possible or not, please correct me if I'm wrong. There are some old threads like http://lists.sip-router.org/pipermail/users/2007-May/011278.html I use Kamailio 4.2.1 trying to create intercepting registrar proxy, which handles NAT (rewrites contact), forwards REGISTER to backend and then stores REGISTER in its internal location on 200 OK from backend, so I call save() from reply route in noreply mode. The problem is, it stores IP source of reply as 'received' and modified contact which i sent out to backend server as 'contact', so it loses original information from initial REGISTER. Is there a way to make save() use original request fields like T_req($ct), or some way to call save() with full set of custom params? Of course, I can save() all REGISTERs in request_route and then delete on backend fail, but it seems to be waste of resources, especially when REGISTER traffic is high :( I'd be glad to hear any advice.

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REGISTER request and save() from onreply_route
by Dmitry Sytchev 19 Jan '15

19 Jan '15

Hi all! Seems to be an old topic but still unclean is it possible or not, please correct me if I'm wrong. There are some old threads like http://lists.sip-router.org/pipermail/users/2007-May/011278.html I use Kamailio 4.2.1 trying to create intercepting registrar proxy, which handles NAT (rewrites contact), forwards REGISTER to backend and then stores REGISTER in its internal location on 200 OK from backend, so I call save() from reply route in noreply mode. The problem is, it stores IP source of reply as 'received' and modified contact which i sent out to backend server as 'contact', so it loses original information from initial REGISTER. Is there a way to make save() use original request fields like T_req($ct), or some way to call save() with full set of custom params? Of course, I can save() all REGISTERs in request_route and then delete on backend fail, but it seems to be waste of resources, especially when REGISTER traffic is high :( I'd be glad to hear any advice. -- Best regards, Dmitry Sytchev, IT Engineer

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Diameter problem on Kamailio
by Kowalski Piotr 3 - Hurt 19 Jan '15

19 Jan '15

Dear sr-users! We are facing serious problem. We built openIMS based on Kamailio. Moreover, we have RCS users registered, but some users are automatically unregistered on the openIMS core: 1) We don't know what a piece of log on S-CSCF means: /usr/sbin/kamailio[17392]: ERROR: --- EXPLOIT --- We need to do an UNREG server SAR assignemnt 2) In log files there is "404 Not Found: Unknown destination" or "404 user offline". Probably there is some diameter problem. Has anyone of you already faced such problem? Many thanks for your answers. Piotr Kowalski Starszy Architekt Techniczny Tel. +48 22 699 50 93 Kom. +48 519 123 289 Orange Orange Polska Hurt Badania i Rozwój Sieci i Platform Usługowych Wydział Usług Konwergentnych Obrzeżna 7 /p. 309, 02-691 Warszawa Fax. +48 22 857 99 86 www.hurt-orange.pl<http://www.hurt-orange.pl/> - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Czy musisz drukować tę wiadomość? Pomyśl o środowisku. Korzystaj z e-faktury. Treść tej wiadomości jest własnością Orange Polska i zawiera informacje stanowiące tajemnicę przedsiębiorstwa Orange Polska. Jeżeli nie jesteście Państwo jej adresatem, bądź otrzymaliście ją przez pomyłkę, prosimy o powiadomienie o tym nadawcy oraz trwałe jej usunięcie. Orange Polska Spółka Akcyjna z siedzibą i adresem w Warszawie (02-326) przy Al. Jerozolimskich 160, wpisana do Rejestru Przedsiębiorców prowadzonego przez Sąd Rejonowy dla m.st. Warszawy XII Wydział Gospodarczy Krajowego Rejestru Sądowego pod numerem 0000010681; REGON 012100784, NIP 526-02-50-995; z pokrytym w całości kapitałem zakładowym wynoszącym 3.937.072.437 złotych.

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Search Un-Register Expires header
by Emilio Soler Muñoz 19 Jan '15

19 Jan '15

Hi, I find a problem when I try to search the header "Expires" with value 0. At my kamailio.cfg the configuration is: if ( search_hf("Expires", ":0@", "a") ) { xlog("L_INFO", "Encontrado expires 0 ($rm)"); } This way find nothing. I test with 0 (no wildcards regular expression), but this make matches in any Register with header Expires content 0. Example: ... if ( search_hf("Expires", "0", "a") ) .... I saw some people control expires=0 at contact header, but not my case. Example: if (search("expires=0")) { xlog("L_INFO", "Encontrado expires 0 ($rm)"); } Regards, -- Emilio Soler emiliosoler(a)gmail.com

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RTPengine
by Kalala Alexander 19 Jan '15

19 Jan '15

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TCP connectivity issue
by Mickael Marrache 19 Jan '15

19 Jan '15

Hi, I use Kamailio as a stateless proxy with UDP + TCP. When TCP is used, I get the following TCP errors: WARNING: <core> [receive.c:265]: receive_msg(): error while trying onreply script ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect xxxxxxxxx failed (timeout) WARNING: <core> [receive.c:265]: receive_msg(): error while trying onreply script ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect xxxxxxxxxx failed (timeout) The TCP config in my script: tcp_connection_lifetime=3605 I get this issue when I receive replies to INVITE from the UAS. I can see the replies using ngrep but Kamailio doesn't seem to be able to process them correctly. I precise that the UAS is behind NAT and NAT configuration is correct (the INVITE does reach the UAS). I also precise that the UAS sends keepalive every 5 sec (I can see the ping-pong using ngrep). I set it to 5 sec to eliminate the possibility that the NAT closes the connection. Thanks, Mickael

1 0

call control for 4.2?
by Peter Villeneuve 18 Jan '15

18 Jan '15

Hi, I want to experiment with the following setup and was wondering which module(s) you all recommend. I have a working simple kamailio 4.2 setup where registered users can only call each other on the same domain (ie no outbound calling to PSTN or other domains). Now I want to explore the possibility of limiting call times for certain users but allowing unlimited call times for others (ie user 5000's calls are limited to 2 minutes while user 6000 has unlimited calling time). There are some tutorials available on the internet but they seem to be quite outdated. Given my simple scenario, should I be looking into using the callcontrol module and install the CDR tool from here http://callcontrol.ag-projects.com/ (I don't really need any complex billing scheme with various rates but rather a simple scheme were user x is either time limited or not). Before I dig in any further, I'd appreciate it of someone could tell me if this is the way to go or is there's a simpler solution. Are there any recent tutorials that I should be looking at? Cheers, Peter

3 4

UID-* modules
by Olle E. Johansson 18 Jan '15

18 Jan '15

hi! Frankly, I don't really know the difference between the uid* modules and the "regular" modules. I know the history, but would like to understand and maybe some text I can add to the README files of these modules to explain to other users what the difference is. /O

1 0

Load balancer - Path and Route header issue
by Mickael Marrache 18 Jan '15

18 Jan '15

Hi, I'm trying to use one Kamailio instance as load balancer for multiple Kamailio instances. The load balancer only performs NAT handling and dispatching of messages. The other instances handle registrations, routing, RTP proxy, authentication, etc. When a REGISTER request is received by the load balancer, a Path header is added to the request which is then forwarded to one of the registrars. The registration request is successfully handled and the Path header is saved in database. All subscribers register through the load balancer. When an INVITE is received from one subscriber by the load balancer, a Record-Route header is added by the load balancer which then forwards the INVITE to one of the other instances. The selected instance authenticates the subscriber, looks for the callee using the REGISTRAR module, adds a Record-Route header, adds a Route header according to the Path header associated with the callee, and forwards the INVITE to the load balancer. The load balancer performs loose routing (removing the Route header), adds another Record-Route header, and forwards the INVITE to the callee using the right transport address (indicated by the received parameter present in the Route header). The flow is right for the INVITE and its replies. However, when the caller sends the ACK, the route set is: load balancer, proxy, load balancer - without any information about how to reach the callee. Therefore, when the ACK reaches the "second" load balancer, the latter forwards it according to the RURI which is the Contact URI present in the 200 OK (which is not routable because of the NAT). I expected Kamailio to reuse the information that was present in the Path header to route the ACK correctly. What am I doing wrong? Thanks, Mickael

1 0

Kamailio with rtpproxy and asterisk in internal lan
by Willian Mazzardo - SYSSVOIP 17 Jan '15

17 Jan '15

Hi all. I want to create a Kamailio server who have dispatcher module, to loadbalance some asterisk servers who have my billing system. But, if I call from internet to my Public IP on Kamailio server, and then call goes to internal IP asterisk, who execute Echo App, I cant hear myself. If I call to my Public IP on Kamailio server, and call goes to EXTERNAL IP on Asterisk, everything works nice. I want this : Sip Client > Public IP Kamailio > Internal IP Kamailio > Internal IP Asterisk Server > Echo() I have rtpproxy running on same machine that Kamailio, and Im running with this command: rtpproxy -F -s udp:127.0.0.1:7722 -l PU.BL.IC.IP/172.16.1.1 -d DBUG:LOG_LOCAL0 My asterisk have 172.16.1.2 as internal IP. https://www.cloudshark.org/captures/15724d2d7e51 This is my tcpdump file captured. Thanks for any help. Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030

1 0

Using the iptel Switch for both home & office environments...
by H. KAIMARAH 15 Jan '15

15 Jan '15

Hello there, Kindly advise me on how i can download and install the iptel soft switch for both Windows and Linux platforms. I intend to install a hosted Sip VoiP service using your open source switch for both home and office environments. I have jst been checking out the documentation from the site but seams some documentation pages are failing to load (timing out). Kindly advise or share with me some of the download links for the application so i can make a successful installation. Sincere regards, -- ***/Happy New Year/*** *HENRY K. Kaimarah* /Cell:/ +256-754462010 /Now in Kampala/ - UGANDA.

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REGISTER replication using t_replicate with ds_select_dst
by Spencer Thomason 15 Jan '15

15 Jan '15

Hello all, We are moving our registration and location services to dedicated instances geographically separate from our proxies and then using redirects to locate the registered user. The idea is to protect the proxies from a potential thundering herd of registrations if there was some network outage I’m using dispatcher to find a registrar that has a registration for the user, if a user is not locally registered a 503 is returned and the next server is contacted. Registrations are replicated to other servers when the client is not behind NAT. If the client is behind NAT, the registrar where the client initially registered performs keep-alives and NAT traversal and also acts as an outbound proxy instead of sending a redirect. I’m attempting to simplify the configuration by using a dispatcher group to classify incoming registrations and to select replication targets but the registrar that initially received the request is in the group as well. Is there an elegant way to chose all destinations except the local instance? I’m currently doing this with the hack below but any suggestions would be greatly appreciated. Also, if anyone has any comments or recommendations on the topology, please share! Thanks, Spencer if (isflagset(FLT_FROMREG)) return; subst('/^From:(.*)sip:.*@[a-zA-Z0-9.:]+(.*)$/From:\1sip:$au@DOMAIN\2/ig'); subst('/^To:(.*)sip:.*@[a-zA-Z0-9.:]+(.*)$/To:\1sip:$au@DOMAIN\2/ig’); $rd = "DOMAIN"; $var(local_uri) = "sip:" + $Ri + ":" + $Rp; ds_select_dst("10", "4"); if ($du != $var(local_uri)) { t_replicate($du); } while (ds_next_dst()) { if ($du != $var(local_uri)) { t_replicate($du); } }

1 0

Changing SDP with subst after rtpproxy_answer corrupts SDP
by Matthias van der Vlies 15 Jan '15

15 Jan '15

Dear all, I have an on-reply route that needs to change the SDP version for the reply coming in. The use case is that I have a mobile originated call and there is some Ericsson switch that doesn't like it when the SDP version is updated (in this case by asterisk) although nothing has changed to the actual SDP (183 session progress and then OK.) Funny thing is that Asterisk will actually drop a call if it receives a re-INVITE with same version... That's why they invented ignoresdpversion, but now it's the other way around.... :) Mobile phone -> Ericsson MSC -> ACME packet -> (18X.4X.XXX.XX) Kamailio (10.41.0.21) -> Asterisk The issue is that the asterisk sends a reply 200 OK, with an updated version because it already sent SDP for 183 session progress. This can be patched in asterisk, but in my scenario I can unfortunately not do that. Thus trying to fix this on Kamailio. I am able to 'fix' this currently by performing a subst on the sdp owner variable: onreply_route[WITHSDP] { if (has_body("application/sdp")) { if(ds_is_from_list()) { rtpproxy_answer("wrei"); *if(subst("/^o=someowner ([0-9]+) ([0-9]+) IN IP4 (.*)$/o=someowner \1 \1 IN IP4 \3/")) {* xlog("L_INFO", "Fixed Asterisk incorrect version number in SDP"); } # tried the answer here as well, but that corrupts it even more } else { rtpproxy_answer("wrie"); } } exit; } However this corrupts the SDP: v=0 o=tismi 652858233 652858233 IN IP4 10.41.0.21*18X.4X.XXX.XX* s=Some server c=IN IP4 18X.4X.XXX.XX t=0 0 m=audio 57644 RTP/AVP 8 0 18 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes When I do not substitute the SDP looks perfectly fine and the external address shows as the IN IP4. But of course the version is incremented: v=0 o=tismi 1606876535 *1606876536* IN IP4 *18X.4X.XXX.XX* s=Some server c=IN IP4 18X.4X.XXX.XX t=0 0 m=audio 55410 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=sdpmangled:yes The ericsson is now accepting this (although it's corrupt, I know.... probably the ACME doing something funky with it), but it causes issues with another unknown piece of equipment that fails on parsing the session owner. I hope there is something wrong with my subst, but I'm afraid I can not do this from the on_reply route because SDP is only updated once it finishes? I know it's dangerous to alter the session version like this, so I made sure the Asterisk will never send a re-INVITE. Now I need a way to not corrupt the o= Kind regards, Matthias van der Vlies

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websocket convert and forward to asterisk
by jaflong jaflong 15 Jan '15

15 Jan '15

Hi Everyone, Does anyone have a example of the config where I can get the following to work I want Kamailio to process websocket converting wss to tcp and srtp to rtp and forward to asterisk as tcp and rtp incoming call on websocket (wss rtp/savpf) --> kamailio/rtpproxy (protocol - convert wss to tcp) and (sdp - convert rtp/savpf to rtp/avp) and (audio - convert srtp to rtp) --> asterisk (recieve protocol: tcp, sdp: rtp/avp and audio:rtp) This application would only have incoming call from websocket to forward on to asterisk.

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Fosdem 2015
by Alexandr Dubovikov 15 Jan '15

15 Jan '15

Hello everybody, what is about Fosdem this year ? Who is going there ? Wbr, Alexandr

2 1

Kamailio with a third party operator
by Jyaim Jyaim 15 Jan '15

15 Jan '15

Hello all, I cannot find documentation on how to configure Kamailio to interwork with another operator's IMS Core. I think the s-cscf should be in charge of detecting a user is not known locally, in order to route requests toward other operator. Do you know where I should look to get infos on this topic ? Thank you!

1 0

Redirect Server Including Path/Recieved Information
by Asgaroth 15 Jan '15

15 Jan '15

Hi All, I am attempting to setup a standalone redirect server which will lookup contact info and redirect to appropriate outbound proxy. The problem I am having is that the registrar is storing the path and recieved information, however, when I perform a lookup and reply with 302, the recieved and path information is not included. Are there module parameters in the registrar module that will include these contact parameters (recieved) or create the appropriate headers (path/route) in the 302 response, or, is this something I need to do manually. Currently I have the following lookup code: route { t_check_trans(); if ( method == "INVITE" ) { xlog("route[MAIN] : $rm : ruri=$ru"); xlog("route[MAIN] : $rm : lookup=sip:$rU@registered.domain"); if ( lookup("location", "sip:$rU@registered.domain") ) { send_reply("302", "Moved Temporarily"); exit; } } } I had a look at the path_mode parameter but this looks like it only takes affect for REGISTER methods. The INVITE that comes in supports the PATH header but I dont see this being passed back. In fact, I'm not entirely sure if the path header is a supported header in the 302 response message. I had a quick google and I cant seem to easily find what headers are suported in the 302 message, and where I need to put the path/recieved information. I presume I can add the recieved info as a parameter to the contact header, but how would i specify the outbound proxy to use, would that be in a route or a path header in the 302 response? I can see that the $du pseudo variable is set with the appropriate outbound path and received information, now I just need to include this information in the 302 response. Any suggestions/comments to assist in how I get this info into the 302 message would be greatly appreciated. Thanks in advance.

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Crash in usrloc, get_all_db_ucontacts
by Kristian F. Høgh 14 Jan '15

14 Jan '15

Hi, On an ancient kamailio 3.1, we occasionally experience crash in the function get_all_db_ucontacts in usrloc. A backtrace reveals the following: #0 0x00002b505f2fb4ff in get_all_db_ucontacts (buf=0xadd270, len=<value optimized out>, flags=64, part_idx=1, part_max=30) at dlist.c:176 176 if (VAL_NULL(ROW_VALUES(row)+4) || p1==0 || p1[0]==0){ The lines around line 176 (which look similar to trunk, except p and p1 are str today) 159 for(i = 0; i < RES_ROW_N(res); i++) { 160 row = RES_ROWS(res) + i; 161 162 /* received */ 163 p = (char*)VAL_STRING(ROW_VALUES(row)); 164 if ( VAL_NULL(ROW_VALUES(row)) || p==0 || p[0]==0 ) { 165 /* contact */ 166 p = (char*)VAL_STRING(ROW_VALUES(row)+1); 167 if (VAL_NULL(ROW_VALUES(row)+1) || p==0 || p[0]==0) { 168 LM_ERR("empty contact -> skipping\n"); 169 continue; 170 } 171 } 172 p_len = strlen(p); 173 174 /* path */ 175 p1 = (char*)VAL_STRING(ROW_VALUES(row)+4); 176 if (VAL_NULL(ROW_VALUES(row)+4) || p1==0 || p1[0]==0){ 177 p1 = NULL; 178 p1_len = 0; 179 } else { 180 p1_len = strlen(p1); 181 } The client from frame #2 (bt full below), send a REGISTER, which was replied "200 OK" at the same second as the crash. Is it a timing issue between REGISTER and nathelper ping? I have a trace and the core. Regards, Kristian Høgh Uni-tel bt full #0 0x00002b505f2fb4ff in get_all_db_ucontacts (buf=0xadd270, len=<value optimized out>, flags=64, part_idx=1, part_max=30) at dlist.c:176 sock = 0x0 dbflags = 11088 host = {s = 0x375ef8e "PROXY_IP:5060", len = 13} i = <value optimized out> row = 0xa13488 dom = 0x2b5062723278 p1 = 0x2 <Address 0x2 out of bounds> now_s = "'2015-01-13 12:58:51'\177\000\000)" port = 11088 p_len = 73 cp = <value optimized out> needed = 16 res = 0xa2ed38 p = 0xcfcdffaf <Address 0xcfcdffaf out of bounds> now_len = 21 proto = 1642847400 p1_len = 0 shortage = 0 query_buf = "select received, contact, socket, cflags, path from location where expires > '2015-01-13 12:58:51' and cflags & 64 = 64 and id % 30 = 1", '\000' <repeats 376 times> query_str = {s = 0x2b505f515520 "select received, contact, socket, cflags, path from location where expires > '2015-01-13 12:58:51' and cflags & 64 = 64 and id % 30 = 1", len = 135} #1 get_all_ucontacts (buf=0xadd270, len=<value optimized out>, flags=64, part_idx=1, part_max=30) at dlist.c:398 No locals. #2 0x00002b5061eb6b1a in nh_timer (ticks=<value optimized out>, timer_idx=<value optimized out>) at nathelper.c:1623 rval = <value optimized out> buf = 0xadd270 cp = <value optimized out> c = {s = 0x0, len = 0} opt = {s = 0x0, len = 0} path = {s = 0x0, len = 0} curi = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0xadd836 "CLIENT_IP:62252\250u\220", len = 14}, port = {s = 0xadd845 "62252\250u\220", len = 5}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = { s = 0x7fff110134b0 "", len = 566862356}, port_no = 50336, proto = 70, type = ERROR_URI_T, flags = 65536, transport = {s = 0x8cbfa0 "", len = 5441948}, ttl = {s = 0x7fff1101352f "", len = 4448297}, user_param = {s = 0x375ef8e "PROXY_IP:5060", len = 13}, maddr = {s = 0x9204b0 "sip3", len = 10677560}, method = {s = 0x0, len = 0}, lr = {s = 0xadd84a "\250u\220", len = 0}, r2 = {s = 0xa2ebc0 "\250\272\252", len = 285291296}, transport_val = {s = 0x40 <Address 0x40 out of bounds>, len = 335544320}, ttl_val = { s = 0x3221c302d0 "H\307\300\017", len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}} he = <value optimized out> send_sock = <value optimized out> flags = <value optimized out> proto = <value optimized out> dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {in6_u = {u6_addr8 = '\000' <repeats 15 times>, u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, id = 0, proto = 0 '\000', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}} iteration = 1 __FUNCTION__ = "nh_timer" #3 0x0000000000521c54 in fork_dummy_timer (child_id=<value optimized out>, desc=<value optimized out>, make_sock=<value optimized out>, f=0x2b5061eb6a70 <nh_timer>, param=0x0, interval=1) at timer_proc.c:71 pid = <value optimized out> #4 0x00002b5061eb42ee in child_init (rank=<value optimized out>) at nathelper.c:673 i = 58055190 #5 0x00000000004db1d0 in init_mod_child (m=0x917dc0, rank=0) at sr_module.c:829 No locals. #6 0x00000000004db1a9 in init_mod_child (m=0x918420, rank=0) at sr_module.c:826 No locals. #7 0x00000000004db1a9 in init_mod_child (m=0x918aa0, rank=0) at sr_module.c:826 No locals.

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One way audio & video - Kamailio + rtpproxy ( NAT call )
by Chirag Ajmera 14 Jan '15

14 Jan '15

Kamailio - 4.2.2 ( SIP server ) Rtpproxy - Git Compiled ( miconda patched version ) Issue: Remote NAT Call Bria Rmt Iphone SIP Extn (3G) ----> Kamailio Server -----> Desktop Bria Client ( Wifi ) Audio and Video packets are sent from iPhone to desktop client .. but nothing otherway Also, being totally new to kamailio I can't figure out logging SIP messages ( INVITE etc ) .. Kindly if anyone can help Thank You Best, Chirag A.

1 0

Fwd: Need a support for Siremis and kamailio server
by Gaurav Rai 14 Jan '15

14 Jan '15

Hi I am using a siremis and *kamailio * hosted server with lylix.net and they give a link for getting a support about Siremis. I need a support to how to make OUT call on PSTN with registered subscriber on Softphone. Currently we have registered user on subscriber table with their location and can make a PEER to PEER audio call and text messaging also. To make OUT call on PSTN we had create a LCR gateway and LCR rule with Prefix but still can not make a OUT call getting a error 404 can u please help me out to make a PSTN call Waiting for your reply Thanks Gaurav Rai +919716238804

2 1

Making rtp streams appear unidirectional
by aft 14 Jan '15

14 Jan '15

Hi, Is it possible to make the rtp stream appear unidirectional? By that i mean, The rtp stream from client to proxy will go through one rtpproxy and proxy to client stream will go through another rtpproxy instance? If not, is it possible to mimic something like that by running rtpproxy in bridge mode where both IPs in bridge mode will be public IP? Thanks in advance cheers arif

2 4

Dispatcher 8 algorithm based on priority strange works
by Yuriy Gorlichenko 14 Jan '15

14 Jan '15

Daniel. I added 8 algorithm to our server and it works with 2 asterisk now but it works strange because: While works server with priority 1 - all ok. When this server goes down dispatcher choose next server with lowes priority. But when server with highest priority waking up dispatcher use server with lowes priority until this server not goes down. here id my config for dispatcher modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:kamailio1@10.0.1.12") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("tm", "reparse_on_dns_failover", 0) (!ds_select_dst("$var(setid)", "8")){ sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","{$rm} from [$fU@$si:$sp] But NO destinations available for $rd \n"); t_on_failure("DISPATCHER_ROLLOVER"); } if (!t_relay()) { sl_reply_error(); we use 4.3 master branch kamailio

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Unexpected evaluation to false on a phonenumber in $avp
by Daniel Tryba 14 Jan '15

14 Jan '15

Not a really important question, just an observation that caused some headscratching on my first day after a nice vacation. I'm using an avp to make a redirect decision. Normally this avp contains Dutch telephone numbers (+31[0-9]{7,10}). With a simple: if($avp(dst_redirectnumber)) { #change $ru } this works flawless (normally). But a client decided, to forward to a Swiss number (+4141743xxxx) but with this number the if condition results in a false. A German number just works fine (+49) The avp is stuffed in an avp_db_query loop with $avp($avp(i:1))=$avp(i:2) as key/value pair from the table. I changed above condition to check if the string length of dst_redirectnumber is larger than 0. But I'm at a loss why the false occurs, first guess was an octal conversion (number only contains the digits 0-7) which results in a number "just" above 2^32. But changing a digit to 8 still fails. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

4 5

Dispatcher Failover algorithm
by Yuriy Gorlichenko 14 Jan '15

14 Jan '15

I as wrote before - we find dispatcher algorithm than can do mechanism something like this: Try call to fist server with max priority or weight. OIf this server unavailible then call second server with less weight and etc. Does anyone know what ling of algorithm we can use for this?

2 6

event on table location Duplicate entry for key 'ruid_idx'
by Yuriy Gorlichenko 14 Jan '15

14 Jan '15

Hello. We use 2 kamailio servers cluster and we have porblems with db. Database failed pecause of error: Could not execute Write_rows_v1 event on table production.location; Duplicate entry 'uloc-54aae947-86d-a67' for key 'ruid_idx', Error_code: 1062; handler error HA_ERR_FOUND_DUPP_KEY; the event's master log FIRST, end_log_pos 380, Internal MariaDB error code: 1062 But a location table no row 'ruid_idx' and no entry uloc-54aae947-86d-a67.

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