- sr-users - kamailio.org

Re: [SR-Users] What is q-value ??
by Daniel Grotti 09 Jul '14

09 Jul '14

Hi, q value is used in location table for serial forking. If you have a subscriber with 2 contact registered you can use the q value to load one contact first (the one with the higher q) and afterwards load and call the second one. Highest is the value stored, highest is the priority. Daniel On 9 Jul 2014 06:20, aawaise <owaisulhaq(a)hotmail.com> wrote: > > There is a mention of highest "q value" in documentation of registrar > module's function "lookup(domain [, uri])". > Can some one please explain what this "q value" is ?? > It is used in 4.2 section of following link. > http://kamailio.org/docs/modules/stable/modules/registrar.html#idp1976344 > > Any help will be appreciated. Thanks. > > > > -- > View this message in context: http://sip-router.1086192.n5.nabble.com/What-is-q-value-tp128895.html > Sent from the Users mailing list archive at Nabble.com. > > _______________________________________________ > 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

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What is q-value ??
by aawaise 09 Jul '14

09 Jul '14

There is a mention of highest "q value" in documentation of registrar module's function "lookup(domain [, uri])". Can some one please explain what this "q value" is ?? It is used in 4.2 section of following link. http://kamailio.org/docs/modules/stable/modules/registrar.html#idp1976344 Any help will be appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/What-is-q-value-tp128895.html Sent from the Users mailing list archive at Nabble.com.

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Using 'reg_fetch_contacts' in Kamailio-3.1.1
by aawaise 09 Jul '14

09 Jul '14

I want to extract domain of a user from Location List For this I found "reg_fetch_contacts(table, uri, profile)" function to be the most suitable function. I want to know when was it added to registrar module ? Can it be used in kamailio version 3.1.1 ? Or I should update my kamailio version ?? Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Using-reg-fetch-contacts-in-Kamaili… Sent from the Users mailing list archive at Nabble.com.

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Access the attributes of a location record
by aawaise 09 Jul '14

09 Jul '14

I want to extract domain attribute of a user from Location Table. I have gone through the readme of registrar module, that has an example of using reg_fetch_contacts() and how to access the attributes of a location record ? But I get the callee domain = NULL when I log using command xlog("callee=>domain: $(ulc(callee=>domain)))\n". Can anyone please help me with the correct syntax. Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Access-the-attributes-of-a-location… Sent from the Users mailing list archive at Nabble.com.

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Help me to configure kamailio on EC2
by Jayaraman, Kamalakannan 09 Jul '14

09 Jul '14

Hi, I had installed kamailio on my AWS EC2, by following the steps in http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x . Since I installed in EC2, i changed the listen=<my-private-ip> advertise=<my-public-ip> in kamailio.cfg file too. I open the 5060 port in AWS console too. Kamailio starts without any error, but I can't able to connect in Jitsi client. Please help me out to do this as soon as possible.

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No sound between chrome and firefox
by Yuriy Gorlichenko 08 Jul '14

08 Jul '14

Hello. I have Kamailio 4.1.3+rtpengine_rtpproxy-ng as module for rtpengine. Kamailio installed as frontend (registration, auth, proxy ) of asterisksk servers. WEBRTC users registred at kamailio and asterisk works as media server. When I try to call from Jssip from Firefox to chrome to way audio is fine. When I call from chrome - I see rtp packets only from firefox. Not from chrome. At kamailio log when I call from chrome log the same as whe i try to call from firefox (I can not see anithing wrong) So somebody else have same issue and may be somebody know reasons of this. Thanks for advice

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sip_trace and registration responses
by Rafal Skrzeczkowski 08 Jul '14

08 Jul '14

Hi, I'm trying to make configuration that will trace registrations. I can trace REGISTER messages using sip_trace. But how can i trace responses generated by pv_www_authenticate using sip_trace? Rafal Skrzeczkowski

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Kamailio, CPL, SIP CGI and alternatives.
by LAA 08 Jul '14

08 Jul '14

Hi all, I'm looking for a mechanism to let users personalize the behaviour of their profile, uploading configuration changes to Kamailio. For example voice mail redirecion, time-based routing, etc. I'm testing CPL scripts (cpl-c module with Kamailio 4.1.3), but I have problems with the behaviour of some features. "reject" and "redirect" seem to be working fine, but "proxy" is not working. ¿Has anybody experienced the same problems? I could hardly find any post regarding CPL scripting with Kamailio. What are the most commonly used alternatives for this purpose with Kamailio? ¿SIP CGI? ¿any other? Best regards. Luis.

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SDP_NAT on Retry
by Timo Klecker 08 Jul '14

08 Jul '14

Hello everybody, we have got a problem with sdp-nat on retries of a 200 OK packet. Due to slow response of our trunk the 200 ok is resend (up to 3 times) but sdp-nat is only done on the first two packets. We are running Kamailio 4.1.0 with the following setup: Trunk --> NAT --> Kamailio --> Voip Bridge The rtp stream from the trunk (public ip) is supposed to go to one of our media gateways (behind 1:1 nat). We use sdp_mangle_ip to replace the private IPs of our Mediagateway with the public ones. This goes well with the first 200 OK packet but not with a retry. A.A.A.A (public trunk media IP) B.B.B.B (our public media IP - 1:1 natted to I.I.I.I) INVITE Trunk --> Kamailio Connection Information (c): IN IP4 A.A.A.A INVITE Kamailio --> Voip Bridge Connection Information (c): IN IP4 A.A.A.A 200 OK Kamailio <-- Voip Bridge Connection Information (c): IN IP4 I.I.I.I 200 OK Trunk <-- Kamailio Connection Information (c): IN IP4 B.B.B.B 200 OK Kamailio <-- Voip Bridge Connection Information (c): IN IP4 I.I.I.I 200 OK Trunk <-- Kamailio Connection Information (c): IN IP4 I.I.I.I ACK Trunk --> Kamailio --> Voip Bridge Our configuration used is the following: route[RELAY] { xlog("L_INFO"," In route Block RELAY \n"); # enable additional event routes for forwarded requests if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (!t_relay()) { xlog("L_NOTICE", " (ROUTE_RELAY) Relay failed, sending error \n"); sl_reply_error(); } exit; } onreply_route[MANAGE_REPLY] { xlog("L_INFO"," In route Block MANAGE_REPLY \n"); if ( isflagset(20) ) { -- we have to do nat route(SDP_NAT); force_send_socket(KAMAILIO_TO_TRUNK_IP); } else { force_send_socket(KAMAILIO_TO_BRIDGE_IP); } } How can we handle the 200 OK retry? Is there a way to do this? Kind regards Timo Klecker

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SIP domain translation between providers
by Jeremy Visser 08 Jul '14

08 Jul '14

G'day folks Struggling to work out whether Kamailio can help me or not in this situation, and I hope someone can tell me whether I'm barking up the wrong tree. I have hundreds of users which register to an old legacy Asterisk box that needs retiring. The "users" are customers consisting of VoIP ATAs, PBXes, softphones, and other hardware phones. I've been told I need to retire the Asterisk system in the next month and move customers to our new provider, who provide the SBC themselves (it's a Broadsoft-based system). Obviously a customer once ported to the new provider requires new SIP proxy and SIP domain settings configured in their CPE. For obvious reasons, I cannot require hundreds of customers to change their configuration overnight. However, there is not enough time to assist hundreds of customers changing their SIP proxy/registrar/domain details. So in the meantime I'm trying to work out a transition solution. I cannot simply change my SIP registrar DNS to be a CNAME for the new provider's registrar, as the domain is different, and the new provider doesn't allow domains other than their own to register with them (even if the username/password portion is correct). My new provider has previously told me they are unwilling to accept my old domain whilst authenticating (they are much bigger than me, and I have no power to change them). So I was hoping to use Kamailio and possibly the Path or Dispatch modules to proxy SIP registration to the new provider, rewriting the domain part of the SIP register string as I go But I can't for the life of me work out whether that's even possible. I have spent weeks poring over various Kamailio configurations posted to the interwebs, but nobody seems to have done this before. Specifically the "translation proxy" -- which I am hoping to make Kamailio -- will need to be stateless, and perform no local authentication. Individual registrations will need to be passed through to the new provider (including authentication!), while still translating the domain in the Request-URI, To, From, and various other SIP headers. Because the current legacy Asterisk box has nat=yes set for all users, I will need to proxy media, because if I start proxying SIP traffic to the new provider, I will be confusing their own NAT hacks. I don't understand the documentation for either the Path or Dispatch modules, and the information on the wiki is confusing and poorly written. I am very familiar with SIP, and have no trouble configuring Asterisk, but trying to work with Kamailio is leaving me feeling like an idiot. However, Kamailio's config is very flexible, so despite this I suspect what I'm trying to achieve may be possible. Any pointers in the right direction? Cheers, Jeremy.

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Strange avp_check behavior
by Klaus Darilion 08 Jul '14

08 Jul '14

Hi! Kamailio 4.1.4. When using avp_check with fnmatch and AVP the result is always TRUE if it is true for 1 time. It only happens with $avp(), but not with $var(). Is this a bug or some special behavior of AVPs? xlog("L_ERR", "======= with $$avp(...)================"); $avp(rpid) = "+499539110"; $avp(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$avp(rpid)", "fm/$avp(pattern)/g")) { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): YES"); } else { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): NO"); } $avp(pattern) = "+499539*"; # MATCH if (avp_check("$avp(rpid)", "fm/$avp(pattern)/g")) { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): YES"); } else { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): NO"); } $avp(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$avp(rpid)", "fm/$avp(pattern)/g")) { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): YES"); } else { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): NO"); } xlog("L_ERR", "======= with $$var(...)================"); $var(rpid) = "+499539110"; $var(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$var(rpid)", "fm/$var(pattern)/g")) { xlog("L_ERR", "$var(rpid) matches $var(pattern): YES"); } else { xlog("L_ERR", "$var(rpid) matches $var(pattern): NO"); } $var(pattern) = "+499539*"; # MATCH if (avp_check("$var(rpid)", "fm/$var(pattern)/g")) { xlog("L_ERR", "$var(rpid) matches $var(pattern): YES"); } else { xlog("L_ERR", "$var(rpid) matches $var(pattern): NO"); } $var(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$var(rpid)", "fm/$var(pattern)/g")) { xlog("L_ERR", "$var(rpid) matches $var(pattern): YES"); } else { xlog("L_ERR", "$var(rpid) matches $var(pattern): NO"); } --> The result is: <script>: ======= with $avp(...)================ <script>: +499539110 matches +499539[2-5]: NO <script>: +499539110 matches +499539*: YES <script>: +499539110 matches +499539[2-5]: YES <script>: ======= with $var(...)================ <script>: +499539110 matches +499539[2-5]: NO <script>: +499539110 matches +499539*: YES <script>: +499539110 matches +499539[2-5]: NO Thanks Klaus

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Where can I get kamctl.oracle
by Konstantin Khatskevich 08 Jul '14

08 Jul '14

Hello! I attempt to dispose the bond ODBC-Oracle, but I cannot find kamctl.oracle (I have any other)& Where can I download this kamctl.oracle? -- Best regards, Konstantin Khatskevich

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Re-Compiling and Re-Installing kamailio server
by aawaise 08 Jul '14

08 Jul '14

I am trying to make changes in code and recompile kamailio server project. Problem is when I try the following command make PREFIX='/usr/local/kamailio-3.1/' FLAVOUR=kamailio.cfg I get the message; 'kamailio' is up to date. I want to know, how to make my machine forget about the older installation, so that new compilation goes good. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Re-Compiling-and-Re-Installing-kama… Sent from the Users mailing list archive at Nabble.com.

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Help me to configure Kamailio on EC2, not able to connect through Jitsi
by Jayaraman, Kamalakannan 08 Jul '14

08 Jul '14

Hi, I had installed kamailio on my AWS EC2, by following the steps in http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x . Since I installed in EC2, i changed the listen=<my-private-ip> advertise=<my-public-ip> in kamailio.cfg file too. I open the 5060 port in AWS console too. Kamailio starts without any error, but I can't able to connect in Jitsi client. Please help me out to do this as soon as possible. -- *Kamalakannan Jayaraman* *Associate Software Engineer* Pearson English kamalkannan.jayaraman(a)pearson.com +91 8754 7754 53 - Mobile kamalakannan.jayaraman.ge - Skype 6th Foor, 10th Block, DLF IT park, Chennai - 600089 India Pearson English is the world's largest provider of English language learning.

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need help on kamailio-snmpstats module
by Allen Zhang 07 Jul '14

07 Jul '14

Hi, I'm having trouble on installing kamailio-snmpstats module. I'm using kamailio 4.0. I want to install kamailio's snmpstats module from RPM. I compiled net-snmp 5.4 from source and built some RPMS from it. I've installed net-snmp 5.4 from RPM on the target machine. And when I run 'yum install kamailio-snmpstats', I get: Error: Package: kamailio-snmpstats-4.0.0-0.el6.x86_64 () Requires: net-snmp-agent-libs Where do I get this net-snmp-agent-libs? Can I buid this from net-snmp source somehow? Regards, Allen

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Measuring Response Time for a SIP Message
by AliReza Khoshgoftar Monfared 07 Jul '14

07 Jul '14

Hi, I am trying to do two types of measurements on a packet (say an INVITE or BYE) that arrives at a kamailio proxy. 1) The time that packet is spending in Kamailio to be processed 2) The total time from when the packet arrives at the machine that runs the proxy till it leaves that machine. I believe kamailio shall be able to help me with (1) and for (2) , I may need to use some linux tools Have anybody tried to measure response time with either definition? and in case that you have, how have you done it? Thanks Alireza

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$avp(received)
by Kelvin Chua 07 Jul '14

07 Jul '14

is there any way of getting the received ip/port/proto from nathelper? modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") set_contact_alias(); xlog("$avp(RECEIVED)"); this gives me <null> Kelvin Chua

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kamailio server over internet
by Gaurav Chaudhary 07 Jul '14

07 Jul '14

sir , i have set-up a kamailio server on a machine , every thing is in default configuration. but the issue is only local clients are able to register. i have forwarded the port 5060 correctly but the clients from some other network over the internet are not able register every request just got time out at the client side. do i nee to enable some specific setting to make it happen.

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Extract value from a function
by aawaise 07 Jul '14

07 Jul '14

I came across this function of registrar module reg_fetch_contacts (domain, uri, profile) How can I log domain of a user using this function ?? Need help in syntax. Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Extract-value-from-a-function-tp128… Sent from the Users mailing list archive at Nabble.com.

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Re: [SR-Users] UAC remote registration - refreshing database
by Dan Christian Bogos 07 Jul '14

07 Jul '14

Hey Daniel, I totally agree on improving versus adding new module. The only thing which made me going towards porting was not knowing how easy would be enhancing vs porting something which is implemented. Also the fact that Alexa has mentioned it as an often requested feature made me think that it could take a while until someone would be able to do it in the existing module. Anyway, thanks for your input! DanB PS: Ovidiu, I appreciate your feedback as well. Good to know that you are around ;).

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MSILO - dumped messages get duplicated
by Marc M. 06 Jul '14

06 Jul '14

Hi, I have a Kamailio 3.2 setup up and running. Recently I configured msilo for offline message store. Essentialy the setup runs fine. Messages to users without location info getting stored. In case the user has location info, but cannot be reached, I made failover_route, so the messages getting stored in this case too. Messages getting dumped at the next REGISTER. However, I have a group of new mobile clients, with unstable connection. It happens many times that they send a REGISTER, but the connection brakes afterwards. It this case the message is getting dumped, but since it cannot be delivered, it runs on the failover_route which stores the message again. However, the previous message is not removed from the db, so the same message is now twice in the db. Next time the message either gets delivered in double, or in case of a new failure, it gets stored 2x2= four times. I need to avoid this duplicates but still make sure the message get delivered sooner or later. Any help is highly appreciated. Here is my routing table: thanks Marc ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } if (is_method("MESSAGE")) { log(1,"-------------------------------------------------------->WE HAVE A NEW MESSAGE\n"); route(MSILO_MESSAGE); } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } ##storing offline messages route[MSILO_MESSAGE]{ if (lookup("location")){ t_on_failure("FAIL_MESSAGE"); t_relay(); xlog("L_NOTICE","---------------------------------MI: ($mi), MB: ($mb)\n"); }else{ xlog("L_NOTICE","------------------MESSAGE PROCESSING: location not found, storing offline message ID: ($ru)\n"); m_store("$ru"); sl_send_reply("202", "ACCEPTED"); xlog("L_NOTICE","------------------MESSAGE PROCESSING: location not found, stored offline message ID: ($ru)\n"); }; exit; }; failure_route[FAIL_MESSAGE]{ xlog("L_NOTICE","------------------MESSAGE PROCESSING: user cannot be reached, storing offline message ID: ($ou)\n"); m_store("$ou"); t_reply("202", "ACCEPTED"); xlog("L_NOTICE","------------------MESSAGE PROCESSING: user cannot be reached, stored offline message ID: ($ou)\n"); exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); # MSILO - dumping user's offline messages m_dump(); log(1,"-------------------------MESSAGE PROCESSING: dumping offline messages\n"); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rtpproxy_offer(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { rtpproxy_answer(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

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RTP to SRTP encryption with Kamailio.
by Dani Kamailio 04 Jul '14

04 Jul '14

Hello, I need to build a VoIP system who receives SIP and RTP traffic in a public IP and encrypt both of them with TLS and SRTP respectively. The main point is to have security inside of the local network (I know it may sound unuseful). So, I was trying to build the whole system in Kamailio but I got stucked with the RTP to SRTP bridge and I do not really know how to do it. I know that there are some modules like rtpproxy-ng and rtpengine as media relay that can handle SRTP... any ideas? I just want to make sure that I am in the right way. If it could be done in Kamailio, have I to write the code to encrypt RTP, like the algorithm or something? Would it be convenient to send the RTP packets to Asterisk? (and Asterisk would somehow encrypt them?) I am pretty new with all of this, any help will be apreciate, thank you.

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no rtpengine.so module
by Yuriy Gorlichenko 04 Jul '14

04 Jul '14

Hello. I install kamailio 4.1.3 and it works fine. But I neen proxy DTLS SRTP to backend media servers through kamailio. rtpproxy no not work with dtls srtp? so I installed rtpengine but can not find rtpengine.so file for copiyng to kamailio modules directory. Does anyone can install rtpengine as kamailio module?

3 4

FW: Module TOPOH: Does not unmask CALL-ID on request BYE in one direction
by Samuel F. 04 Jul '14

04 Jul '14

Hi, We are setting up a Kamailio Proxy with the Topoh module in place masking everything including CALL-ID. It seems to be masking everything just fine except for CALL-ID in BYE requests (verified with PCAPs) when received from downstream. Upstream works fine. Looking at the source code for the module 4.1.2 we see the unmasking being done like this: if(dialog) { /* dialog request */ th_unmask_ruri(&msg); th_unmask_route(&msg); th_unmask_refer_to(&msg); if(direction==1) { th_unmask_callid(&msg); } } It only seems to be unmasking for one direction. From my understanding BYE can be initated from both directions? Might be something I don’t understand here how the source code works but I wanted to ask for any pointers of help here.. been troubleshooting this for a good while now. Kind regards, Samuel

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Re: [SR-Users] UAC remote registration - refreshing database
by Dan Christian Bogos 04 Jul '14

04 Jul '14

Hey Alex, Many thanks for so fast answer. Could be I have missed the past interest (hence re-posting). Anyway, I wonder what would be the chances that we get Ovidiu's interest so he can port registrant module to kamailio maybe? Unfortunately without reloads it is hard to push remote registration in enterprise solutions. Thanks again, DanB On 03.07.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > On 07/02/2014 06:22 AM, Dan Christian Bogos wrote: >> >Anybody aware if it is possible to refresh the list of remote >> >registrations from the database without restarting the whole server? > This gets asked a lot, and the answer is no. But it is probably a widely > desired feature set by now. > > -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please > be kind to the English language: > http://www.entrepreneur.com/article/232906

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