- sr-users - kamailio.org

Getting Started with SIREMIS
by David Wilson 19 Sep '13

19 Sep '13

Hi All, I have Kamailio 4.0.3 installed via APT onto Ubuntu 12.04 and have Siremis 4.0 installed and running nicely. Is there an example config file that demonstrates how to use the features configurable through Siremis? For example, when I populate a UACReg record via Siremis, I had to adapt the kamailio.cfg to use that information. Similar adaptations are required for LCR or CR records and these are where I'm struggling; there is some routing code that is implied by the data structures, but the documentation for the corresponding modules doesn't provide an overall example. Regards, Dave.

2 1

Modifying path to kamailio.cfg
by Grant Bagdasarian 19 Sep '13

19 Sep '13

Hello, I compiled and installed Kamailio 4.0.3 like usual on a new test machine, but changed the CFGFILE variable in /etc/init.d/kamailio to /usr/local/etc/kamailio/kamailio.cfg. Kamailio started normally. When I reset CFGFILE to CFGFILE=/etc/kamailio/kamailio.cfg and start Kamailio again, I'm getting an error saying it can't find the file at /usr/local/etc/kamailio/kamailio.cfg. I renamed the config file at /usr/local/etc/kamailio/ to make sure it's not using that one, but it still is. Why is the change not being accepted? Regards, Grant

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siremis 4.0 problem
by Marek Cervenka 19 Sep '13

19 Sep '13

hi, i'm trying siremis 4.0 with kamailio 4.0.3. installation with web wizard was successfull i have strange problem with login. login succed but then i'm immediately logged out. i see MSG_LOGIN_SUCCESSFUL in log/DEBUG.log can you point me where i can enable additional DEBUG in siremis? or any other ideas what's the problem? thanks centos6, mysql 5.1, php 5.3 -- --------------------------------------- Marek Cervenka =======================================

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SCA Module Drops Shared Line if Barge-In Aborted on Polycom
by Robert Boisvert 18 Sep '13

18 Sep '13

SCA Developers, I ran into a problem using the SCA module and was looking for some assistance for the following situation. Can you help? Here is the scenario: 1. When a conversation is occurring on a shared line with Polycom phones; 2. Another caller on the same shared line picks up the phone and uses the barge-in feature; 3. The caller aborts the barge-in before being connected; 4. SCA or Polycom, I don't know which at this point, turns off the LEDs on the shared line and then the shared line no longer behaves like a shared line. 5. Even though the existing shared call continues, if the new caller picks up the shared line he gets a green LED, hears a dial tone and is not prompted to barge-in by the soft keys. 6. When all parties end the conversation the shared line returns to its previous correct behavior. Have you seen something like this? Do you have any suggestions on how to troubleshoot or workaround the problem? Thanks again for your help. The SCA module remains a very useful product for us, Bob

1 0

Store dispatcher weight attribute in dispatcher table
by Grant Bagdasarian 18 Sep '13

18 Sep '13

Hello, Is it possible to store the weight attribute in the dispatcher table? If so, which column must I use? I see it is possible when using the list file. The attrs_avp is the closest thing, but I think this value has to be loaded from the database also? Regards, Grant

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Authentication and outbound proxy
by Isaac McDonald 18 Sep '13

18 Sep '13

Hello, Is there any way to authenticate requests when using Kamailio as an outbound proxy? For example, if all the phones are configured to use Kamailio as an outbound proxy for NAT traversal, the credentials on the phone authenticate against the destination SIP proxy and not the outbound SIP proxy (Kamailio). Is there a way to have the credentials on the phone authenticate to BOTH the outbound proxy and the destination SIP proxy? Thanks, Isaac

3 4

Regd. Presence_xml module
by Shankar 18 Sep '13

18 Sep '13

Hi, While reading the module documentation of 'PRESENCE_XML' module, I came across 'intergrated_xcap_server'. If it is set to value '0', then it has to use xcap_client module to fetch documents from XCAP server. But as per documentation, it says this mode is not supported. So can I assume that only intergrated xcap server is supported in kamailio and not standalone. If YES, module xcap_client cannot be used at all? Can someone please confirm the same. Regards, Shankar

1 0

Re: [SR-Users] (no subject)
by Keith 17 Sep '13

17 Sep '13

Hi, I am trying to use SIPP to load test Kamailio. I am using a combination with Freeswitch and Kamailio as an SBC with the calls being delivered to Asterisk. The problem seems to be with the ACK sent back from SIPP on response to the 200 OK. This ACK is not being forwarded by Kamailio and therefore FS drops the call. I have scoured every page and tried everything so any help would be appreciated. Thanks, Keith

2 1

TEXTOPS parser problem with binary data in MIME body
by Klaus Feichtinger 17 Sep '13

17 Sep '13

Hello, I have a problem with Kamailio version 3.2.4 (tested also with 3.3.5) regarding "binary data" in message bodies. According RFC3261 it is explicitly allowed using binary data within MIME bodies: RFC 3261 section 7.4.1: SIP messages MAY contain binary bodies or body parts. When no explicit charset parameter is provided by the sender, media subtypes of the "text" type are defined to have a default charset value of "UTF-8". However, the Kamailio function "Regular Expression Transformation" (re.subst), which is based on the transformation class (exported by the textops module) is causing problems in our customer system. In regular scenarios, Kamailio is receiving SIP INVITE messages with a MIME body, which is containing following parts: - application/sdp (standard conform) - application/x-q931 (Cisco proprietary with BINARY data!) - application/gtd (Cisco proprietary with ASCII strings) The content of the "x-q931" and "gtd" body parts is depending on (UUS1) data that were received on the ISDN line. Kamailio has to forward User-User specific information from the ISDN line to SIP user agents (in form of the "User-to-User" header field). The content of these UUS1 data may contain also byte values "00", which is legitimate. In general, Kamailio is in every INVITE message searching specific content in the last body (application/gtd) and copying this content to a config variable. As soon as the x-q931 body contains nul values (0x00 in binary format), the parser stops at this position and cannot parse the rest of the message. Therefore, I am missing the information that should be copied to the SIP header field, as the parser stops before the end of the message body! As long as the message body does not contain 0x00, it is working fine! My question is: - is this a bug in Kamailio parsing functions? - is this a design issue of Kamailio text parsers (as binary data are allowed acc. RFC3261) - does anybody know a solution for this problem? This "bug" is causing big troubles.... Thanks in advance, Klaus Feichtinger P.S. trace data (1)...(3) of my problem (1) exemplary content of the x-q931 body ======================Sending: Binary Message Body Sep 13 10:50:19: Content-Type: application/x-q931 08 01 B4 05 04 03 80 90 A2 18 01 89 1E 02 82 88 6C 05 B1 35 30 30 34 70 05 B1 35 30 30 31 7D 02 91 81 7E 18 04 1D 42 75 25 92 43 31 62 94 00 00 2C 68 20 64 00 62 F2 10 41 B9 D7 BD 0D 0A (2) SIP INVITE message (debugger view) =============================INVITE sip:115101@<ipv4>:5060 SIP/2.0 Via: SIP/2.0/UDP <ipv4>:5060;branch=z9hG4bK171164E From: <sip:1101015004@<ipv4>>;tag=29E16410-2541 To: <sip:115101@<ipv4>> Call-ID: 1DBCDACB-1B9911E3-8992FF70-D2BED293@<ipv4> Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 90 CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:1101015004@<ipv4>:5060> Expires: 1800 P-Asserted-Identity: <sip:1101015004@<ipv4>> Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 820 --uniqueBoundary Content-Type: application/sdp Content-Disposition: session;handling=required v=0 o=CiscoSystemsSIP-GW-UserAgent 1770 5294 IN IP4 <ipv4> s=SIP Call c=IN IP4 <ipv4> t=0 0 m=audio 16384 RTP/AVP 8 0 c=IN IP4 <ipv4> a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 --uniqueBoundary Content-Type: application/x-q931 Content-Disposition: signal;handling=optional Content-Length: 62 4" l15004p15001}~Bu%C1b..,h d.brA9W --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,, USI,rate,c,s,c,1 USI,lay1,ulaw TMR,00 CPN,08,,1,5001 CGN,08,,1,,,5004 UUS,3,1d427525924331629400002c6820640062f21041b9d7bd CPC,09 FCI,,,,,,,y, GCI,185ef1291b9911e381d500270dff3fa0 --uniqueBoundary-- (3) config excerpt ============if (has_body("multipart/mixed")) { if (filter_body("application/sdp")) { remove_hf("Mime-Version"); remove_hf("Content-Type"); append_hf("Content-Type: application/sdp\r\n"); } else { xlog("L_WARN", " <route> message body part 'application/sdp' not found\n"); } $var(UUS)=$(rb{re.subst,/^(.*)UUS,.,([a-z0-9,]*)..[A-Z][A-Z][A-Z],(.*)/\2/s}); append_hf("User-to-User: $var(UUS)\r\n", "CSeq"); if !(subst_body('/Content-Disposition..session.handling=required....//s')) { xlog("L_WARN", " <route> substituting Content-Disposition FAILED!!! \n"); } msg_apply_changes(); if (search_body("a=[a-z]+:.+[\r\n]{4}$")) { #!ifdef WITH_XLOGDEBUG xlog("L_INFO", " <route> double CRLF at the end of the message body detected - is reduced to one now. \n"); #!endif $var(sdp) = $(rb{s.striptail,2}); set_body("$var(sdp)", "application/sdp"); } }

2 2

Registering "upstream" with the uac module, but with authentication
by Steve Davies 17 Sep '13

17 Sep '13

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

4 15

Calling users on another kamailio server
by Brian Wallen 17 Sep '13

17 Sep '13

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.

4 15

kamailio presence and XCAP related issue
by Rupayan Dutta 17 Sep '13

17 Sep '13

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.

1 0

Asterisk and Kamailio Integration CAPACITY ISSUE
by Hassan Wajahat 16 Sep '13

16 Sep '13

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

2 1

Kamailio realtime fails to forward register with Asterisk.
by imperium broadcast 15 Sep '13

15 Sep '13

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

2 6

issues with dispatcher rerouting to next server when getting a 500 from freeswitch
by Joseph Watson 14 Sep '13

14 Sep '13

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

2 1

json module / quotes in result
by Florian Lohoff 13 Sep '13

13 Sep '13

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

3 2

New Developer: Charles Chance
by Daniel-Constantin Mierla 13 Sep '13

13 Sep '13

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 -

3 2

NAT function
by Kethzer Docteur 13 Sep '13

13 Sep '13

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

2 3

Pseudo-variables and branches
by Steve Davies 13 Sep '13

13 Sep '13

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

3 2

Htable simple CPS Limit
by Gareth Rylance 13 Sep '13

13 Sep '13

I was inspired by Daniels ClueCon 2013 talk and have decide to try to add simple rate limiting using a HTable. $var(srcgroup) is an id that identifies the invites source/ I wish to limit invites from a given source. $var(rateHashMin) was added just to allow simple monitoring. I query the table via XML-RPC to get an idea of current load. I think it would be better to add it to a seperate table and then I could increase the exprires time to capture more of the course grain points. I am interested in any comments. ---------------------- modparam("htable", "htable", "htable_source_group_rates=>size=8;dbtable=htable_source_group_rates;initval=0;autoexpire=300;") modparam("htable", "htable", "htable_source_group_limits=>size=8;dbtable=htable_source_group_limits;initval=500;") ---------------------- ---------------------- route[SOURCE_GROUP_LIMIT] { if (is_method("INVITE")){ $var(rateHashSec) = (str)$var(srcgroup)+":sec:"+$timef(%Y/%m/%d_%H_%M_%S); $var(rateHashMin) = (str)$var(srcgroup)+":min:"+$timef(%Y/%m/%d_%H_%M_00); $sht(htable_source_group_rates=>$var(rateHashSec)) = $sht(htable_source_group_rates=>$var(rateHashSec)) + 1; $sht(htable_source_group_rates=>$var(rateHashMin)) = $sht(htable_source_group_rates=>$var(rateHashMin)) + 1; if($sht(htable_source_group_rates=>$var(rateHashSec)) > $sht(htable_source_group_limits=>$var(srcgroup))) { sl_send_reply("503", "$si over source Limit of $sht(htable_source_group_rates=>$var(rateHashSec)) / $sht(htable_source_group_limits=>$var(srcgroup)"); exit; } } } ---------------------- Thanks Gareth

2 2

Error in Kamailio_ims v 4.0.3
by Sai Krishna Kota 13 Sep '13

13 Sep '13

Hello Carsten, Yha I have not included 'xmlrpc' for compilation. Can you please list all the required modules to be compiled with the ims modules so that I can know more about kamailio-ims. Thank you, Kota Sai Krishna

2 1

Cross-Compilation Problems for mips (gcc 3.4.2)
by Tirant Lo Blanc 12 Sep '13

12 Sep '13

Hi, I've been exploring the possibility to port Kamailio 3.3 (SER) to some MIPS boards. I didn't have any problem with the first one, with a GCC 4.3.4 toolchain. But on my second board (gcc 3.4.2) I am having problems when compiling. Are there any requirements for gcc/binutils/kernel versions? This is the log I am getting: CC (mipsel-linux-uclibc-gcc) [ser]sip_msg_clone.o In file included from atomic_ops.h:181, from sip_msg_clone.c:43: atomic/atomic_unknown.h:59:2: warning: #warning no native memory barrier implementations, falling back to slow lock based workarround CC (mipsel-linux-uclibc-gcc) [ser]socket_info.o In file included from socket_info.c:836: /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:22: error: parse error before "__u32" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:28: error: parse error before "__u32" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:30: error: parse error before "nlmsg_flags" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:31: error: parse error before "nlmsg_seq" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:32: error: parse error before "nlmsg_pid" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/netlink.h:83: error: field `msg' has incomplete type In file included from socket_info.c:837: /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:253: error: parse error before "__u32" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:255: error: parse error before "rta_expires" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:256: error: parse error before "rta_error" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:257: error: parse error before "rta_used" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:260: error: parse error before "rta_id" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:261: error: parse error before "rta_ts" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:262: error: parse error before "rta_tsage" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:333: error: parse error before "__s32" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:359: error: parse error before "__u16" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:361: error: parse error before "ndm_type" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:404: error: parse error before "__u32" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:406: error: parse error before "ndm_updated" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:407: error: parse error before "ndm_refcnt" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:519: error: parse error before "__u32" /opt/buildroot-gcc342/bin/../lib/gcc/mipsel-linux-uclibc/3.4.2/../../../../mipsel-linux-uclibc/sys-include/linux/rtnetlink.h:521: error: parse error before "tcm_info" socket_info.c: In function `addattr_l': socket_info.c:874: error: dereferencing pointer to incomplete type socket_info.c:878: error: dereferencing pointer to incomplete type socket_info.c:882: error: dereferencing pointer to incomplete type socket_info.c:882: error: dereferencing pointer to incomplete type socket_info.c: In function `nl_bound_sock': socket_info.c:891: error: storage size of 'la' isn't known socket_info.c:891: warning: unused variable `la' socket_info.c: In function `get_flags': socket_info.c:929: error: field `nlh' has incomplete type socket_info.c:953: error: dereferencing pointer to incomplete type socket_info.c:957: error: dereferencing pointer to incomplete type socket_info.c:967: error: invalid application of `sizeof' to incomplete type `nlmsghdr' socket_info.c:967: error: dereferencing pointer to incomplete type socket_info.c:967: error: invalid application of `sizeof' to incomplete type `nlmsghdr' socket_info.c:967: error: dereferencing pointer to incomplete type socket_info.c:967: error: dereferencing pointer to incomplete type socket_info.c:967: error: dereferencing pointer to incomplete type socket_info.c:967: warning: left-hand operand of comma expression has no effect socket_info.c:968: error: invalid application of `sizeof' to incomplete type `nlmsghdr' socket_info.c:970: error: dereferencing pointer to incomplete type socket_info.c:970: error: invalid application of `sizeof' to incomplete type `nlmsghdr' socket_info.c: In function `build_iface_list': socket_info.c:996: error: field `nlh' has incomplete type socket_info.c:1043: error: dereferencing pointer to incomplete type socket_info.c:1047: error: dereferencing pointer to incomplete type socket_info.c:1057: error: invalid application of `sizeof' to incomplete type `nlmsghdr' socket_info.c:1057: error: dereferencing pointer to incomplete type socket_info.c:1057: error: invalid application of `sizeof' to incomplete type `nlmsghdr' socket_info.c:1057: error: dereferencing pointer to incomplete type socket_info.c:1057: error: dereferencing pointer to incomplete type socket_info.c:1057: error: dereferencing pointer to incomplete type socket_info.c:1057: warning: left-hand operand of comma expression has no effect socket_info.c:1058: error: invalid application of `sizeof' to incomplete type `nlmsghdr' socket_info.c:1060: error: dereferencing pointer to incomplete type socket_info.c:1060: error: invalid application of `sizeof' to incomplete type `nlmsghdr' socket_info.c:1067: error: dereferencing pointer to incomplete type socket_info.c:1067: error: invalid application of `sizeof' to incomplete type `nlmsghdr' make[1]: *** [socket_info.o] Error 1 make: *** [bin] Error 2 Do you have an idea on what could be producing the error? Thanks and kind regards

3 3

tls nat
by hiro 12 Sep '13

12 Sep '13

After many failures because of broken libraries I managed to decrypt my problematic TLS sessions providing the private key of kamailio to wireshark. With TLS+SRTP enabled my nokia phones send session progress and pracks with rtp port. This breaks NAT/rtpproxy in kamailio, which replaces the rtpproxy port in session progress, but then forgets about it for the 200 OK. Attached is a tree overview and the conversations of each phone with kamailio. hiro

2 5

Re: [SR-Users] Carrierroute
by Keith 12 Sep '13

12 Sep '13

Hi, I am trying to use the carrierroute module, however the carrierroute.so file is not in the module directory. I installed Kamailio version 4 from yum. Thanks, Keith

2 1

How to convert recorded video call in .rtp file using kamailio and rtpproxy in to any video file format. ??
by Manas Mohanty 12 Sep '13

12 Sep '13

Hi All, How to convert recorded video call in .rtp file using kamailio and rtpproxy in to any video file format. ?? Thanks, Manas

2 3

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