sr-users October 2010

sr-users@lists.kamailio.org

77 participants
157 discussions

Fwd: Tutorial: SIP SIMPLE Presence Made Simple with Kamailio 3.1

by Rouskol Andrey

Daniel, What whould you think about this variant of xcap authentication: event_route[xhttp:request] { xdbg("===== xhttp: request [$rv] $rm => $hu\n"); if($hu=~"^/xcap-root/") { # xcap ops $xcapuri(u=>data) = $hu; if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:"+ $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; if($xcapuri(u=>auid)=="xcap-caps") { ... skipped ... } #!ifdef WITH_XHTTPAUTH #!ifdef WITH_MULTIDOMAIN if (!www_authorize("$(var(uri){uri.host})", "subscriber")) { www_challenge("$(var(uri){uri.host})", "0"); #!else if (!www_authorize("xcap", "subscriber")) { www_challenge("xcap", "0"); #!endif exit; } set_reply_close(); set_reply_no_connect(); #!ifdef WITH_XHTTPAUTH # be sure auth user access only its documents #!ifdef WITH_MULTIDOMAIN if($aU=~".+@.+") $var(tmp) = "sip:"+$aU; else $var(tmp) = "sip:"+$aU+"@"+$(var(uri){uri.host}); if ($var(uri)!=$var(tmp)) { xdbg("===== xhttp: Forbidden!!! $var(tmp) - $var(uri)\n"); #!else if ($au!=$(var(uri){uri.user})) { xdbg("===== xhttp: Forbidden!!! $au - $(var(uri){uri.user})\n"); #!endif .. Regards, Andrey.

14 years, 4 months

Get registration information of a specific subscriber in script

by Adieu

Hello everybody, I just started to use kamailio. For the basic functions, I can get them working properly. But when I started to write my own script, I encountered some problem. The scenario is like this: I have a linksys spa3102 pstn-voip gateway at home which register itself at my server on the internet. I use adsl at home so the ip address change dynamically. When I want to route all my pstn calls to spa3102, I have to figure out the current ip address of my home connection. I tried to use dynamic dns, but my router does not update the ip address sometimes. So I was thinking about solving this problem in the kamailio configuration script. All I need is to get the registration information of spa3102 so that I can route my pstn call to it. I guess there might be a function doing this work. It expect a subscriber name as a variable and returns the registration information of the subscriber. I tried to dig the function out of kamailio documents but failed. And google doesn't seem to provide much help. I don't know if this function exist or not. Could someone confirm my guess or point out where I can find it? I will be much appreciated. //bow Adieu

14 years, 4 months

#!endif and SUBSCRIBE issues 3.0.x

by Jeremya

I've discovered - after quite a while - that #!endif doesn't work if there is extra text on the line #!endif works #!endif # someblock doesn't work. Could this be addressed in code or at least noted in documentation? I also have the case of SUBSCRIBE where it is a later (re)SUBSCRIBE after an initial SUBSCRIBE. This has a to-tag. This is not handled by the default script. Could the default script be altered to handle SUBSCRIBE with to-tag please? Thanks Jeremy

14 years, 4 months

Fwd: perl_exec() using parameter?

by "Nicolas Rüger"

Hello, any ideas for parameter passing to perl-scripts are appreciated!!! Regards, Nicolas -------- original-------- Hello, is there any chance to pass parameters (like from_uri or to_uri) from the kamailio routing logic to a perl script? I know that the whole sip message is provided as parameter but I wonder if there's a way to pass just the caller's uri (from tag) or the callee's uri (to tag) as a second parameter to the perl script. By using xlog it's clear that in $fu and $ru one can find the uri's. xlog("from: $fu to: $ru \n") OUTPUT: "from: sip:user01@mykamailio.de to:user02@mykamailio.de" BUT using these variables as a second parameter in perl_exec() leads to an error on kamailio start-up cause of bad config file. perl_exec("my_perl_script",$fu); Any suggestions or ideas??? Thank you... Regards, Nicolas -- GRATIS! Movie-FLAT mit über 300 Videos. Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome

14 years, 4 months

SER+nathelper+RTPproxy

by Medina Curafina

Dear Sirs, I am writing to you, 'cause I have some issues about SER work. I installed ser-0.9.4, and test it in my local network, and everything works fine with hello_world.cfg. After that I had different situation. I wanted to make a call from one of my local hosts to remote host (which I registered to your SERWeb in format "username"@iptel.org). Of course I changed my ser.cfg file from hello_world.cfg into example.cfg (which one I send you in Attachment, grey parts are changed lines, because I need to show work of nathelper module of SER). I Installed RTP Proxy from CVS, and followed basic instructions. so, in my local network I am host 102(a)192.168.1.103 (x-Lite4) my domain is default one* (localhost)*, my NAT box is Cisco Lynksis Router. When I make a call (host 102-->SER-->NAT translation-->SERWeb(213.192.59.75)--->external IP of my friends PC), I can reach him (He can hear a phone call, but when he answers can't hear me, and I still get ringing tone, and after 7 second it looks like he canceled a call). In Wireshark I captured INVITE, 100, 180,200, and 487 "Request Terminated" messages. In section of VOIP Calls Graph, I see that messages didn't pass further off SERWeb. I am just beginner in SER use, and I really need your help. I really tried to understand what to do, I get through documentation about RTP Proxy and I feel that I make some mistakes in that part. I was passing through this for so long time, and I don't have anybody to ask for help except you. Do I need to configure my Router, fix configuration of host, configuration file, or something about RTP proxy. Hope hearing from you soon, Very Grateful, Medina H.

14 years, 4 months

SER nathelper+RTPproxy

by Medina Curafina

I am writing to you, 'cause I have some issues about SER work. I installed ser-0.9.4, and test it in my local network, and everything works fine with hello_world.cfg. After that I had different situation. I wanted to make a call from one of my local hosts to remote host (which I registered to SERWeb in format "username"@iptel.org). Of course I changed my ser.cfg file from hello_world.cfg into example.cfg (which one I send you in Attachment, grey parts are changed lines, because I need to show work of nathelper module of SER). I Installed RTP Proxy from CVS, and followed basic instructions. so, in my local network I am host host(a)192.168.x.x <102(a)192.168.x.x.> (x-Lite4) my domain is default one* (localhost)*, my NAT box is Cisco Lynksis Router. When I make a call (host 102-->SER-->NAT translation-->SERWeb(213.192.59.75)--->external IP of my friends PC), I can reach him (He can hear a ringing tone, but when he answers can't hear me, and I still get ringing tone, and after 7 second it looks like he canceled a call). In Wireshark I captured INVITE, 100, 180,200, and 487 "Request Terminated" messages. In section of VOIP Calls Graph, I see that messages didn't pass further off SERWeb. I am just beginner in SER use, and I really need your help. I really tried to understand what to do, I get through documentation about RTP Proxy and I feel that I make some mistakes in that part. Is there anyone who can give me advice what I should do, and where am I wrong? Hope hearing from you soon, Very Grateful, Medy H.

14 years, 4 months

Kamailio as IPv6 to IPv4 Gateway

by Joe Uelk

Hello all, I'm looking to implement the following scenario: Step 1 SIP Server A sends INVITE to port 5060 over IPv6 to Kamailio: 2001::1 --udp/tcp--> 2001::2:5060 Step 2 Kamailio SIP NATs the INVITE and sends it out IPv4 to SIP Server B on port 6000 1.1.1.1 --udp/tcp--> 1.1.1.2:6000 Step 3 All SIP Server B responses within this SIP dialog are sent back to SIP Server A along the same path. There is more to this but I want to accomplish this first and go from there. Can someone point me toward a resource on how to get this created? Do I need to use the WITH_NAT function in kamailio.cfg? Thanks! Joe

14 years, 4 months

SIP Ping utility with kamailio

by JR Richardson

Hi All, Can someone point me in the right direction of a command line SIP Ping utility or how to invoke from Kamailio? I see there is a sip_ping.pl script in voip-hacks, does anyone have copy-paste text version of that, all I can find is the PDF? Thanks. JR -- JR Richardson Engineering for the Masses

14 years, 4 months

call for assistance

by Meftah Tayeb

hi folk, could you join irc irc.freenode.net #switchfin to start porting SR to BF? not exactly porting but integrating if you get there we will organise a conf call hosted by me i could provide SIP and evean PSTN access thank you -- Meftah Tayeb inum: +883510001288000 mobile: +213660347746

14 years, 4 months

how can string "0" be equal to int 0?

by Juha Heinanen

i have stored in htable a key_value=0 with key_type=1, i.e., string. then i have this piece of test code: if (defined($sht(htable=>$var(carrier_key)::id))) { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is defined\n"); } else { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is NOT defined\n"); }; $var(carrier_id) = $sht(htable=>$var(carrier_key)::id); if ($var(carrier_id) == (int)0) { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is not found\n"); } else { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is found\n"); }; and when the piece is executed, this is printed to syslog: Oct 15 12:20:33 squeeze /usr/sbin/trunk-proxy[2915]: INFO: 0 is defined Oct 15 12:20:33 squeeze /usr/sbin/trunk-proxy[2915]: INFO: 0 is not found i understand the first line, because the entry is in the htable, but how is it possible that string "0" is equal to int 0? -- juha

14 years, 4 months

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sr-users October 2010