- sr-users - kamailio.org

Re: [SR-Users] sr-users Digest, Vol 67, Issue 33
by Nina Bersamina 13 Dec '10

13 Dec '10

remove --- On Mon, 12/13/10, sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org> wrote: From: sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org> Subject: sr-users Digest, Vol 67, Issue 33 To: sr-users(a)lists.sip-router.org Date: Monday, December 13, 2010, 6:48 AM Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Dialog module with 2 servers and 2 separate databases. (Daniel-Constantin Mierla) 2. First public release of sip:provider Community Edition (Andreas Granig) 3. First public release of sip:provider Community Edition (Andreas Granig) 4. Re: Crash (michel freiha) ---------------------------------------------------------------------- Message: 1 Date: Mon, 13 Dec 2010 15:12:13 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Dialog module with 2 servers and 2 separate databases. To: "Pan B. Christensen" <pan(a)ibidium.no> Cc: sr-users(a)lists.sip-router.org Message-ID: <4D0629BD.8050501(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, one option you can do is to write to db immediately when a call is active (see dialog module parameters) and do a query to the other server database in addition to counting the local instance active calls. Another one, different, is to use memcache for a communication system between two or more instances. Cheers, Daniel On 12/13/10 2:37 PM, Pan B. Christensen wrote: > Thanks for your reply, Daniel. > The purpose is to do busy forwarding without querying the client when > the user already has >= X active calls. X will normally be 1 (call > waiting inactive) or 2 (call waiting active). Advanced users may > possibly set a higher value. Counting the number of calls on > the server and doing busy forwarding based on that rather than waiting > for a "486 Busy here" from the client has several advantages. > Currently, I've written code to do this with the dispatcher module, > and it's working great with only one server. Here's a code snippet: > $var(dlg_busy) = 0; > get_profile_size("busy", "$avp(s:uid)", "$var(dlg_busy)"); > if ( $var(dlg_busy) >= $avp(s:busy_level) ) { > if ($avp(s:cfb_status) == "on") { > $rU = $avp(s:cfb_number); > xlog("L_INFO", "-------------------- $avp(s:uid) has > $var(dlg_busy) active calls. Treshold $avp(s:busy_level). Forwarding > on busy to $rU --------------------\n"); > route(10); > } > ... > } > Based on your reply, I guess one way to solve this would be to write > the get_profile_size function in sqlops, query the two dialog > databases and add the numbers. This would still require the customer > to change their database design. Is there an easier or better way to > do this? > I also wote code to do busy forwarding if the client replies with 486 > (do not disturb activated), 603 (call rejected) etc. > This code works for normal busy forwarding if Polycom is set to 1 call > per line key (default is 8). We'll then have to provision the > $avp(s:busy_level) variable to the clients instead of handling it > server-side. If a user now wants to change the setting, he'll have to > reboot his phone after doing so. Changing the > reg.x.callsPerLineKey setting in the phone also limits the number of > outgoing calls the user can make. We'll also have to make code for all > the other hardphones the customer is planning to use plus make guides > on how to change the setting for all kinds of softphones. We want to > avoid all this. > With kind regards, > Pan > > ----- Original Message ----- > *From:* Daniel-Constantin Mierla <mailto:miconda@gmail.com> > *To:* Pan B. Christensen <mailto:pan@ibidium.no> > *Cc:* sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > *Sent:* Monday, December 13, 2010 12:26 PM > *Subject:* Re: [SR-Users] Dialog module with 2 servers and 2 > separate databases. > > > > On 12/10/10 2:17 PM, Pan B. Christensen wrote: >> Hello, >> My customer has the following database design. >> Voip server 1 talks to SQL server 1. >> Voip server 2 talks to SQL server 2. >> Voip 1 and Voip 2 are load-balanced. >> Each SQL server has two databases. Database 1 contains >> semi-static data like call forwarding properties for users and is >> read-only. This is replicated from a third SQL server which the >> web interface writes to. Database 2 is read/write, is not >> replicated and contains data that is updated frequently like user >> location and now dialog info. >> Voip server 1 is not allowed to talk to SQL server 2 and vice versa. >> I'm using forward() to send authenticated REGISTERs to the other >> server so that it'll write this to RAM and its own SQL server. >> Thus, both servers are aware of clients authenticated and >> registered by the other server. >> How can I make both servers be aware of active calls on the other >> server? > what is the purpose? > > Practically, it is not possible to track a call in two instances, > because, unlike registration where is just a storage of mappings > between contact and aor, call states of dialog module involve more > processing logic, including timeouts and sending BYEs. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > Kamailio (OpenSER) Advanced Training > Jan 24-26, 2011, Irvine, CA, USA > http://www.asipto.com > > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.5.449 / Virus Database: 271.1.1/3312 - Release Date: > 12/12/10 19:34:00 > > > _______________________________________________ > 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 Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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First public release of sip:provider Community Edition
by Andreas Granig 13 Dec '10

13 Dec '10

Hi all, We at Sipwise are excited to announce the first public release of the sip:provider Community Edition (http://www.sipwise.com/products/spce/). It is a fully open-source SIP based Class5 VoIP soft-switch, providing every component an operator needs to offer VoIP services. It comes as a communication platform leveraging the capabilities of Kamailio, SEMS and Asterisk, complemented by our own open-sourced building blocks to provide consistent and easy-to-use provisioning, billing and configuration maintenance. The different parts are carefully integrated with each other to form a fully featured VoIP soft-switch. Please check http://www.sipwise.com/news/announcements/spce-first-release/ for more information on this release. Have fun playing with it, we hope it's as useful to you as it is for us. Andreas

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First public release of sip:provider Community Edition
by Andreas Granig 13 Dec '10

13 Dec '10

Hi all, We at Sipwise are excited to announce the first public release of the sip:provider Community Edition (http://www.sipwise.com/products/spce/). It is a fully open-source SIP based Class5 VoIP soft-switch, providing every component an operator needs to offer VoIP services. It comes as a communication platform leveraging the capabilities of Kamailio, SEMS and Asterisk, complemented by our own open-sourced building blocks to provide consistent and easy-to-use provisioning, billing and configuration maintenance. The different parts are carefully integrated with each other to form a fully featured VoIP soft-switch. The platform will make it much easier for new Kamailio users to get started with VoIP, and will provide missing parts in the open-source VoIP eco-system for more experienced users. Please check http://www.sipwise.com/news/announcements/spce-first-release/ for more information on this release. Have fun playing with it, we hope it's as useful to you as it is for us. Andreas

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Dialog module with 2 servers and 2 separate databases.
by Pan B. Christensen 13 Dec '10

13 Dec '10

Hello, My customer has the following database design. Voip server 1 talks to SQL server 1. Voip server 2 talks to SQL server 2. Voip 1 and Voip 2 are load-balanced. Each SQL server has two databases. Database 1 contains semi-static data like call forwarding properties for users and is read-only. This is replicated from a third SQL server which the web interface writes to. Database 2 is read/write, is not replicated and contains data that is updated frequently like user location and now dialog info. Voip server 1 is not allowed to talk to SQL server 2 and vice versa. I'm using forward() to send authenticated REGISTERs to the other server so that it'll write this to RAM and its own SQL server. Thus, both servers are aware of clients authenticated and registered by the other server. How can I make both servers be aware of active calls on the other server? The customer is looking into possibly changing their design, but this may be a lengthy process involving lots of people. With kind regards, Pan

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Trying to drop invite 200ok reply in onreply_route
by nikita 13 Dec '10

13 Dec '10

Hello, I'm using kamailio 3.0 with mediaproxy and I want to cancel calls if I have no mediaproxy relay connected to my mediaproxy dispatcher. In my request route I'm checking if "use_media_proxy()" is returning me a positive result and it's working fine : use_media_proxy(); if($rc < 0) { sl_send_reply("480","Temporarily Unavailable"); exit; } But I'm also want check in my onreply_route If I still have an active relay because I need to rewrite the sdp in the 200/OK reply. My problem is that I don't know how to end the transaction, I have tried : if (is_method("INVITE") && status == "200") { #!ifdef WITH_MEDIA_PROXY use_media_proxy(); if($rc < 0) { xlog("L_ERR", "invite reply error $rc \n"); dlg_bye("all"); } #!endif } and also : if (is_method("INVITE") && status == "200") { #!ifdef WITH_MEDIA_PROXY use_media_proxy(); if($rc < 0) { xlog("L_ERR", "invite reply error $rc \n"); drop(); } #!endif } In both case I'm seeing in my log file that use_media_proxy() returned me -1, but the reply is routed to the caller ... Any idea on how I can cancel my call in the onreply_route ? Also, I don't know if it's a good thing to make a second call to "use_media_proxy" in the onreply_route, mediaproxy module couldn't reply me with a different relay address if I have multiple mediaproxy relay? Regards, -- Nikita

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Dispatcher Crash when all hosts are invalid
by Eric Hiller 13 Dec '10

13 Dec '10

Dispatcher crashes everytime guaranteed if there is a single dead host kamailio exits with a core dump (see below). If there are two dead hosts entire kamailio locks up (no core dump). Any ideas on this one? The invalid flag is never marked in the DB btw. Crashes at ds_select_dst having never reached "COMPLETED FIRST DST LOOKUP" # ------------------------- request routing logic ------------------- # main routing logic # ------------------ route{ # initial checks if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; xlog("[$TF] VALID RECV $rm\n $fu ($si:$sp) to $ru\n"); $avp(orig_du) = $du; if($(fU{s.select,1,-})){ # from a user-formed message $avp(user) = $(fU{s.select,0,-}); xlog(" USER $avp(user) DETECTED\n"); } if(pv_isset("$avp(user)") && !ds_is_from_list()){ # from client # route to client PBX if(method == "REGISTER"){ route(3); } else { route(1); } } if(ds_is_from_list()){ # from client, outbound from pbx xlog(" IS FROM PBX\n"); if(is_e164($rU)){ xlog(" IS E164"); # all pbx-pbx traffic should be in user format # since this is pbx-non-pbx send it to gateway # gateway only takes E164 route(2); } if($(rU{s.select,0,-})){ # send directly from pbx to client route(4); } } if($si == "GATEWAY_IP_ADDRESS"){ # lookup enum # route to appropriate pbx via DISPATCHER based on user returned xlog(" CALL FROM GATEWAY\n"); } } route[1]{ route(3); xlog(" ROUTED to $avp(user) PBX\n"); ds_select_dst($avp(user),"7"); xlog(" COMPLETED FIRST DST LOOKUP\n"); if($avp(orig_rd) == $rd){ exec("kamctl dispatcher reload"); ds_select_dst($avp(user),"9"); } xlog(" host is now $rd; all is $ru || $du\n"); if($avp(orig_du) != $du){ t_on_failure("1"); if(!t_relay()){ xlog(" T_RELAY() FAILED\n"); sl_reply_error(); } } else { xlog(" RELOAD STILL YIELDED NOTHING\n"); } } route[2]{ xlog(" Sending to gateway..."); #$rd = "GATEWAY"; } route[3]{ #register if(!www_authorize("domain.net","subscriber")){ www_challenge("domain.net","0"); return; } else { xlog(" AUTHENTICATED $avp(user)\n"); save("location"); return; } } route[4]{ $rU = "2-1"; #$rd = "10.1.40.7"; #$rp = "16314"; lookup("location"); xlog(" ROUTING TO USER $rU\n"); forward(); } failure_route[1]{ xlog(" FAILED FAILURE_ROUTE[1]\n"); if(t_any_timeout()){ xlog(" MARKING DEAD HOST $du\n"); ds_mark_dst(); xlog(" TIMEOUT!\n"); } } 0(2934) ERROR: <script>: [Thu Dec 2 20:15:34 2010] VALID RECV REGISTER sip:3-1@domain.com (10.1.40.7:43899) to sip:domain.com 0(2934) ERROR: <script>: USER 3 DETECTED 0(2934) ERROR: <script>: ROUTED to 3 PBX 0(2934) ERROR: <script>: mark#1 0(2934) ERROR: <script>: host is now domain.com; all is sip:domain.com || sip:xxx.xxx.xxx.166:5060 0(2934) ERROR: <script>: [Thu Dec 2 20:15:35 2010] VALID RECV REGISTER sip:3-1@domain.com (10.1.40.7:43899) to sip:domain.com 0(2934) ERROR: <script>: USER 3 DETECTED 0(2934) ERROR: <script>: ROUTED to 3 PBX 0(2934) ERROR: <script>: mark#1 0(2934) ERROR: <script>: host is now domain.com; all is sip:domain.com || sip:xxx.xxx.xxx.166:5060 1(2935) ERROR: <script>: FAILED FAILURE_ROUTE[1] 1(2935) ERROR: <script>: MARKING DEAD HOST sip:xxx.xxx.xxx.166:5060 1(2935) ERROR: <script>: TIMEOUT! 0(2934) ERROR: <script>: [Thu Dec 2 20:15:50 2010] VALID RECV REGISTER sip:3-1@domain.com (10.1.40.7:29372) to sip:domain.com 0(2934) ERROR: <script>: USER 3 DETECTED 0(2934) ERROR: <script>: ROUTED to 3 PBX 0(2934) ERROR: <script>: mark#1 0(2934) ERROR: <script>: host is now domain.com; all is sip:domain.com || sip:xxx.xxx.xxx.165:5060 0(2934) ERROR: <script>: [Thu Dec 2 20:15:50 2010] VALID RECV REGISTER sip:3-1@domain.com (10.1.40.7:29372) to sip:domain.com 0(2934) ERROR: <script>: USER 3 DETECTED 0(2934) ERROR: <script>: ROUTED to 3 PBX 0(2934) ERROR: <script>: mark#1 0(2934) ERROR: <script>: host is now domain.com; all is sip:domain.com || sip:xxx.xxx.xxx.165:5060 1(2935) ERROR: <script>: FAILED FAILURE_ROUTE[1] 1(2935) ERROR: <script>: MARKING DEAD HOST sip:xxx.xxx.xxx.165:5060 1(2935) ERROR: <script>: TIMEOUT! 0(2934) ERROR: <script>: [Thu Dec 2 20:15:55 2010] VALID RECV REGISTER sip:3-1@domain.com (10.1.40.7:29373) to sip:domain.com 0(2934) ERROR: <script>: USER 3 DETECTED 0(2934) ERROR: <script>: ROUTED to 3 PBX :::::Core file from crash with 1 invalid host::::: Program terminated with signal 8, Arithmetic exception. #0 0x0069e92b in ds_select_dst (msg=0x82ffb00, set=2, alg=7, mode=0) at dispatch.c:1720 1720 i = (i+1)%(idx->nr-1); Thanks for any help on this one! -Eric

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Problems with Siremis 2.0
by Kurt Mullen 13 Dec '10

13 Dec '10

After completing the web install, I am getting the following error after login. Not Found The requested URL /siremis/system/general_default was not found on this server. Any help out there? Kurt A. Mullen Practical PC, LLC (O) 830.542.4102 x204 (F) 210.767.3912

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Re: [SR-Users] Error opening Kamailio's FIFO
by Kurt Mullen 13 Dec '10

13 Dec '10

This issue has not been resolved. I am still getting the same error when trying to use kamctl for anything. I have tried all of the suggestions posted to date with no resolution. Is there anything more I can provide to help resolve this problem? Kurt -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: Monday, December 06, 2010 4:02 PM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 67, Issue 20 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. presence_xml and xpath (Andreas Granig) 2. Re: presence_xml and xpath (Daniel-Constantin Mierla) 3. Re: presence_xml and xpath (Andreas Granig) 4. Re: presence_xml and xpath (Daniel-Constantin Mierla) 5. Re: presence_xml and xpath (Andreas Granig) 6. Re: Error opening Kamailio's FIFO (Kurt Mullen) ---------------------------------------------------------------------- Message: 1 Date: Mon, 06 Dec 2010 15:13:06 +0100 From: Andreas Granig <agranig(a)sipwise.com> Subject: [SR-Users] presence_xml and xpath To: sr-users(a)lists.sip-router.org Message-ID: <4CFCEF72.4040308(a)sipwise.com> Content-Type: text/plain; charset="iso-8859-1" Hi, I'm having troubles fetching an xpath from an xml document using presence_xml, so I probably do something wrong (using kamailio 3.0.x): $xml(x=>doc) = '<?xml version="1.0" encoding="utf-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:test@mydomain.com"><tuple id="86ae65b7-42de-4399-b635-295caad13aac"><status><basic>none</basic></statu s></tuple></presence>'; xlog("L_INFO", "all=$xml(x=>doc)\n"); -> prints the whole document, which is ok xlog("L_INFO", "tmp1=$xml(x=>xpath:/)\n"); -> also prints the whole document (including the xml tag in the beginning, not sure if this is ok) xlog("L_INFO", "tmp2=$xml(x=>xpath:/presence)\n"); -> prints an empty string How is this actually supposed to work? What I'd like to do in the end is extracting for example the "none" from the "basic"-tag by specifying "xpath:/presence/tuple/status/basic", which also returns an empty string now, also when appended with "/text()". Thanks a lot, Andreas

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Re: [SR-Users] Purple Error.
by mayamatakeshi 10 Dec '10

10 Dec '10

On Fri, Dec 10, 2010 at 10:57 AM, David J <david(a)styleflare.com> wrote: > I will test it. Seems to have worked > OK. After confirmation, please notify the developers: http://sip-router.org/tracker/ > On Dec 9, 2010 8:40 PM, "mayamatakeshi" <mayamatakeshi(a)gmail.com> wrote: > > On Thu, Dec 9, 2010 at 11:55 PM, David J <david(a)styleflare.com> wrote: > > > >> Yes. That module is loaded. > >> On Dec 9, 2010 7:48 AM, "mayamatakeshi" <mayamatakeshi(a)gmail.com> > wrote: > >> > >> > On Thu, Dec 9, 2010 at 9:36 PM, David J. <david(a)styleflare.com> > wrote: > >> > > >> >> If anyone can direct me how to resolve. > >> >> > >> >> > >> >> I load presence.so,pua.so,purple.so > >> >> > >> >> When I run kamailio I see. > >> >> > >> >> > >> >> Dec 9 14:36:05 localhost /usr/local/sbin/kamailio[1605]: ERROR: > purple > >> >> [purple.c:148]: can't import load_tm > >> >> > >> > > >> > Function load_tm is defined at module tm.so. Are you loading this > module? > >> > > > > > > I took a look at the code. > > It seems there was a change in the API but module purple was not updated. > It > > calls load_tm this way (purple.c): > > > > if((load_tm=(load_tm_f)find_export("load_tm", 0, 0))==NULL) > > > > But all other modules call it this way: > > > > if((load_tm=(load_tm_f)find_export("load_tm", NO_SCRIPT, 0))==NULL) > > > > NO_SCRIPT is defined as -1. > > > > So I believe just changing the above will permit to load the module. >

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lcr in 3.1 - no reload without Siremis?
by Anders 10 Dec '10

10 Dec '10

I am upgrading from 1.5 to 3.1, but ran into some issues. The one right now is that I can’t seem to use the lcr and gws, although I’ve spent quite some time looking at the documentation. When it hits the next_gw() inside my route[PSTN] it finds nothing, and I have already added to the tables. But I added it manually to the tables, so maybe I needed to load them to memory – but “kamctl lcr reload” has been discontinued, so no loading. Is Siremis necessary as the input tool for lcr and gws? Just to make sure that it's not because of my cfg - here is the part of the cfg where it returns false in next_gw(): route[PSTN] { if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); if (!load_gws(1)) { sl_send_reply("500", "Our mistake - cannot load gateways"); exit; } if (!next_gw()) { sl_send_reply("503", "Service not available (no more gateways)"); exit; } } if (!t_relay()) { sl_reply_error(); } exit; } Any ideas what is wrong? //Anders

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Re: [SR-Users] Purple Error.
by mayamatakeshi 10 Dec '10

10 Dec '10

On Thu, Dec 9, 2010 at 11:55 PM, David J <david(a)styleflare.com> wrote: > Yes. That module is loaded. > On Dec 9, 2010 7:48 AM, "mayamatakeshi" <mayamatakeshi(a)gmail.com> wrote: > > > On Thu, Dec 9, 2010 at 9:36 PM, David J. <david(a)styleflare.com> wrote: > > > >> If anyone can direct me how to resolve. > >> > >> > >> I load presence.so,pua.so,purple.so > >> > >> When I run kamailio I see. > >> > >> > >> Dec 9 14:36:05 localhost /usr/local/sbin/kamailio[1605]: ERROR: purple > >> [purple.c:148]: can't import load_tm > >> > > > > Function load_tm is defined at module tm.so. Are you loading this module? > I took a look at the code. It seems there was a change in the API but module purple was not updated. It calls load_tm this way (purple.c): if((load_tm=(load_tm_f)find_export("load_tm", 0, 0))==NULL) But all other modules call it this way: if((load_tm=(load_tm_f)find_export("load_tm", NO_SCRIPT, 0))==NULL) NO_SCRIPT is defined as -1. So I believe just changing the above will permit to load the module.

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Purple Error.
by David J. 09 Dec '10

09 Dec '10

If anyone can direct me how to resolve. I load presence.so,pua.so,purple.so When I run kamailio I see. Dec 9 14:36:05 localhost /usr/local/sbin/kamailio[1605]: ERROR: purple [purple.c:148]: can't import load_tm Thanks.

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TCP Overload Congestion
by Jijo 09 Dec '10

09 Dec '10

Hi Andrei, I'm observing TCP Recv buffer getting full when we are doing a load run (30cps) on Proxy with TCP. Basically the congestion is happening on the TCP connection from SIP Server to Proxy. I beleive kamailio is not processing the message fast enough, due to this congestion is occuring. Did anyone observe this behavior? How do we resolve it? Thanks Jijo

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Kamailio 3.1 compile issue on Sparc Solaris 10
by KevinJin 09 Dec '10

09 Dec '10

Hi All, Trying to upgrade the Kamailio to the latest 3.1 version, but got the issue below when compile it, any help is appreciated. AppSer01:root@/usr/local/src/kamailio-3.1.0/kamailio$ gmake Q=0 all Makefile.defs defs skipped Makefile.defs defs skipped gmake[1]: Entering directory `/usr/local/src/kamailio-3.1.0/kamailio/modules/db_mysql' gcc -fPIC -DPIC -g -O9 -funroll-loops -mcpu=ultrasparc -mtune=ultrasparc -Wall -m64 -DNAME='"kamailio"' -DVERSION='"3.1.0"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"gcc 3.4.6"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001000 -DCFG_DIR='"/usr/local/kamailio-3.1/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER_MOD_INTERFACE -I/usr/local/include -I/usr/local/include/mysql -I/usr/local/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_cmd.c -o my_cmd.o -MMD -MP In file included from ../../globals.h:37, from ../../pt.h:42, from ../../counters.h:43, from mysql_mod.h:39, from my_cmd.c:40: ../../ip_addr.h:86: error: field `sin6' has incomplete type ../../ip_addr.h: In function `ip_addr_loopback': ../../ip_addr.h:283: warning: implicit declaration of function `htonl' ../../ip_addr.h:286: warning: implicit declaration of function `IN6_IS_ADDR_LOOPBACK' ../../ip_addr.h: In function `sockaddr2ip_addr': ../../ip_addr.h:353: error: dereferencing pointer to incomplete type ../../ip_addr.h: In function `su_getport': ../../ip_addr.h:399: warning: implicit declaration of function `ntohs' ../../ip_addr.h: In function `su_setport': ../../ip_addr.h:418: warning: implicit declaration of function `htons' gmake[1]: *** [my_cmd.o] Error 1 gmake[1]: Leaving directory `/usr/local/src/kamailio-3.1.0/kamailio/modules/db_mysql' gmake: *** [modules] Error 1 Thanks, Kevin

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using purple module with latest 3.1 release
by Meftah Tayeb 08 Dec '10

08 Dec '10

hi guys, i have installed/Configured kamailio 3.1.1 with db, auth, and everyhting is up evean the purple module is loaded but i don't know how to handl the mod_purple route i read the purple module wiki page but didn't understand in what place i should put it thanks for any help -- Meftah Tayeb inum: +883510001288000 Phone: +13602276297 Fax: +12538020313

1 0

Dialplan help
by Phillman25 Kyriacou 08 Dec '10

08 Dec '10

Hi I was wondering if you could help, I am trying to match more than one expression in the dialplan list when a request for an invite is received. My purpose is to authenticate source ip with the correct prefix so I want to match both prefix+source ip before allowing to proceed with a specific entry in dialplan list. Can that be done using Siremis v2 ? Thanks! Phillip

2 4

gws with same ip
by Anders 08 Dec '10

08 Dec '10

Hi, Running Kamailio 1.5: I have two different trunks with one carrier. The IP where I send the INVITE is the same, but the prefix differs. So in the gw table, I enter the two lines, - same IP, but with different digits in the 'tag' column, and different grp_id numbers. I then set up the lcr table and point to the respective grp_id's in the gw table. Problem: it ignores the grp_id when I make a call, meaning that when there are two gw's with the same IP address, it just takes the one that comes first, even though it doesn't have the grp_id I asked for in the lcr table. Is this normal behavior? If so, how do I get around this? Thanks! --Anders

3 2

whereis the serctl utility or its replacement
by rbond1 08 Dec '10

08 Dec '10

Hello. The documentation i see on the web keeps mentioning SERCTL to do management of users and other admin. Can someone provide a link to more relevant information on that utility or it's python replacement? I have compiled ser under ubuntu with mysql support, but now i need something that will help me add contacts to the database easily as the docs imply. The version of ser i pulled using the download information is ser 3.2.0-dev1 according to the sercmd output. Is the sip-router.org site also where SERCTL can be downloaded? What is the git string to aquire it if so? thx

3 3

Re: [SR-Users] Store Source IP in DB
by Bernhard Suttner 07 Dec '10

07 Dec '10

Hi, thanks for these replys. I already saw the nathelper module but I am not sure, which method will store the "received" value. Does the nat_uac_test save the received value? Best regards, Bernhard ----- Original Message ----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] To: Bernhard Suttner [mailto:bernhard.suttner@winet.ch] Cc: sr-users(a)lists.sip-router.org Sent: Tue, 07 Dec 2010 18:59:36 +0100 Subject: Re: [SR-Users] Store Source IP in DB > > > Am 07.12.2010 15:20, schrieb Bernhard Suttner: > > Hi, > > > > I am using Kamailio with usrloc, nathelper and register module (and > > some other). All the data will be stored within a MySQL database. The > > contact address will be stored within the database. Is it somehow > > possible to store the SOURCE-IP of a incoming REGISTER from a phone > > to Kamailio somewhere in the database? It's just for debugging > > purposes. > > If you are doing NAT traversal, e.g. fix_nated_register(), you will have > the source IP address of the REGISTER in the location table in the > "received" column. > > regards > Klaus >

1 0

Store Source IP in DB
by Bernhard Suttner 07 Dec '10

07 Dec '10

Hi, I am using Kamailio with usrloc, nathelper and register module (and some other). All the data will be stored within a MySQL database. The contact address will be stored within the database. Is it somehow possible to store the SOURCE-IP of a incoming REGISTER from a phone to Kamailio somewhere in the database? It's just for debugging purposes. Best regards, Bernhard

3 2

Kamailio 1.5.4 notls Core Dump
by Stagg Shelton 07 Dec '10

07 Dec '10

Kamailio 1.5.4 segmentation faulted today. I was able to determine with gdb that the error occurred at mediaproxy.c line 1104. Line 1104 is a call to memcpy Core was generated by `/sbin/kamailio -m 512'. Program terminated with signal 11, Segmentation fault. #0 0x00007f1469db6bb5 in replace_element (msg=0x77e470, old_element=<value optimized out>, new_element=0x7fffbd553210) at mediaproxy.c:1104 1104 memcpy(buf, new_element->s, new_element->len); 1.5.4 is new to our production environment, we were previously running 1.5.2 notls (also no mediaproxy) for more than a year without issue. 1.5.4 notls with mediaproxy has been running in our lab and preproduction QA environments since March of this year without any real segmentation faults. Does anyone have any ideas of what may have caused this, or suggestions to get around this. I do see that 1.5.5 is available, although there does not appear to have been too many changes to mediaproxy.c Thank you in advance Stagg

2 4

Kamailio and Asterisk Integration
by Kurt Mullen 07 Dec '10

07 Dec '10

There is a tutorial posted at <http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb> http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb I am confused. It seems that this tutorial has Asterisk AND Kamailio on the same physical server. If this is true, why would you do that. I thought the point was to move Kamailio to a separate server to handle all the SIP sessions and have Asterisk on another box to handle media, such as voicemail, IVR's and such. If I am correct, can someone give guidance on how to integrate if Asterisk is on a separate server? Thanks, Kurt

2 1

Kamailio and Asterisk Integration
by Kurt Mullen 06 Dec '10

06 Dec '10

There is a tutorial posted at http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb I am confused. It seems that this tutorial has Asterisk AND Kamailio on the same physical server. If this is true, why would you do that. I thought the point was to move Kamailio to a separate server to handle all the SIP sessions and have Asterisk on another box to handle media, such as voicemail, IVR's and such. If I am correct, can someone give guidance on how to integrate if Asterisk is on a separate server? Thanks, Kurt

1 0

Re: [SR-Users] Error opening Kamailio's FIFO
by Kurt Mullen 06 Dec '10

06 Dec '10

> This should be a UNIX style path e.g. /tmp/kamailio.fifo The directory where the fifo is must be writable by the kamailio user. I thought the line modparam("mi_fifo", "fifo_mode",0666) set the permissions on the folder/file. Also, I tried changing the path to /tmp/kamailio.fifo that did not work either Please change as below: > kamctlrc > ## path to FIFO file > OSER_FIFO="/tmp/kamailio_fifo" kamailio.cfg > # ----- mi_fifo params ----- >modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") I have also tried this. Here is what I am getting. ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. Interesting side note: After the initial installation and reboot, Kamailio came up and was running and kamctl moni displayed the correct results. However after running "service kamailio restart" I began getting the error again and have not been able to clear it. -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: Monday, December 06, 2010 3:37 AM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 67, Issue 17 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: lcr in 3.1 - no reload without Siremis? (Anders) 2. Error opening Kamailio's FIFO (Kurt Mullen) 3. Re: lcr in 3.1 - no reload without Siremis? (Juha Heinanen) 4. Re: Error opening Kamailio's FIFO (marius zbihlei) 5. Re: Error opening Kamailio's FIFO (Huy Nguyen) ---------------------------------------------------------------------- Message: 1 Date: Sun, 5 Dec 2010 14:32:20 -0500 From: Anders <vaerge(a)gmail.com> Subject: Re: [SR-Users] lcr in 3.1 - no reload without Siremis? To: Juha Heinanen <jh(a)tutpro.com> Cc: sr-users(a)lists.sip-router.org Message-ID: <AANLkTimfd8CdXENb+oX6n2YNx54ohbudz8jUraX=yVGy(a)mail.gmail.com> Content-Type: text/plain; charset=windows-1252 For anyone else who might not be able to find the old `kamctl lcr`-commands, the alternatives to some can be found here: http://www.kamailio.org/docs/modules/3.1.x/modules/lcr.html#id2960364 Being able to reload the lcr didn't solve my problem though - still not finding any gw when I run next_gw(), so I'm still open for input... On Sat, Dec 4, 2010 at 6:08 PM, Juha Heinanen <jh(a)tutpro.com> wrote: > Anders writes: > >> But I added it manually to the tables, so maybe I needed to load them >> to memory ? but ?kamctl lcr reload? has been discontinued, so no >> loading. Is Siremis necessary as the input tool for lcr and gws? > > anders, > > read what is new in 3.1 wiki page and lcr manual. ?also, ctl help > knows about lcr ctl commands. > > -- juha > ------------------------------ Message: 2 Date: Sun, 5 Dec 2010 17:40:26 -0600 From: "Kurt Mullen" <kmullen(a)practical-pc.com> Subject: [SR-Users] Error opening Kamailio's FIFO To: <users(a)lists.kamailio.org> Message-ID: <005301cb94d5$cb5a2aa0$620e7fe0$(a)practical-pc.com> Content-Type: text/plain; charset="us-ascii" I am getting this error when I try to run kamctl: ERROR: Error opening Kamailio's FIFO FIFO ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "FIFO")' in your config ERROR: and also have loaded the mi_fifo module. I am able to start Kamailio with Kamailio start I have poured over all the responses on Google for two days trying to fix this error. I have tried everything I know of. I have tried each example. kamctlrc DB Path is correct: # database path used by dbtext or db_berkeley DB_PATH="/usr/share/kamailio/dbtext" ## control engine: FIFO or UNIXSOCK ## - default FIFO CTLENGINE="FIFO" ## path to FIFO file OSER_FIFO="FIFO" (I have already tried commenting this out) kamailio.cfg loadmodule "mi_fifo.so" Path to modules is correct: mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio.fifo") modparam("mi_fifo", "fifo_mode",0666) This is installed on Ubuntu Server 10.10 64 bit Intel . Can someone please help? Kurt A. Mullen Practical PC, LLC (O) 830.542.4102 x204 (F) 210.767.3912 ------------------------------ Message: 4 Date: Mon, 6 Dec 2010 11:28:08 +0200 From: marius zbihlei <marius.zbihlei(a)1and1.ro> Subject: Re: [SR-Users] Error opening Kamailio's FIFO To: <sr-users(a)lists.sip-router.org> Message-ID: <4CFCACA8.4060701(a)1and1.ro> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" On 12/06/2010 01:40 AM, Kurt Mullen wrote: Hello, > > I am getting this error when I try to run kamctl: > > ERROR: Error opening Kamailio's FIFO FIFO > > ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", > "FIFO")' in your config > > ERROR: and also have loaded the mi_fifo module. > > I am able to start Kamailio with Kamailio start > > I have poured over all the responses on Google for two days trying to > fix this error. I have tried everything I know of. > > I have tried each example. > > kamctlrc > > DB Path is correct: > > # database path used by dbtext or db_berkeley > > DB_PATH="/usr/share/kamailio/dbtext" > > ## control engine: FIFO or UNIXSOCK > > ## - default FIFO > > CTLENGINE="FIFO" > > ## path to FIFO file > > OSER_FIFO="FIFO" (I have already tried commenting this out) > This should be a UNIX style path e.g. /tmp/kamailio.fifo The directory where the fifo is must be writable by the kamailio user. Marius > > kamailio.cfg > > loadmodule "mi_fifo.so" > > Path to modules is correct: > > mpath="modules_k:modules" > > #!else > > mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" > > #!endif > > # ----- mi_fifo params ----- > > modparam("mi_fifo", "fifo_name", "/tmp/kamailio.fifo") > > modparam("mi_fifo", "fifo_mode",0666) > > This is installed on Ubuntu Server 10.10 64 bit Intel . > > Can someone please help? > > *Kurt A. Mullen* > > Practical PC, LLC > > (O) 830.542.4102 x204 > > (F) 210.767.3912 >

1 0

presence_xml and xpath
by Andreas Granig 06 Dec '10

06 Dec '10

Hi, I'm having troubles fetching an xpath from an xml document using presence_xml, so I probably do something wrong (using kamailio 3.0.x): $xml(x=>doc) = '<?xml version="1.0" encoding="utf-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:test@mydomain.com"><tuple id="86ae65b7-42de-4399-b635-295caad13aac"><status><basic>none</basic></status></tuple></presence>'; xlog("L_INFO", "all=$xml(x=>doc)\n"); -> prints the whole document, which is ok xlog("L_INFO", "tmp1=$xml(x=>xpath:/)\n"); -> also prints the whole document (including the xml tag in the beginning, not sure if this is ok) xlog("L_INFO", "tmp2=$xml(x=>xpath:/presence)\n"); -> prints an empty string How is this actually supposed to work? What I'd like to do in the end is extracting for example the "none" from the "basic"-tag by specifying "xpath:/presence/tuple/status/basic", which also returns an empty string now, also when appended with "/text()". Thanks a lot, Andreas

2 4

Error opening Kamailio's FIFO
by Kurt Mullen 06 Dec '10

06 Dec '10

I am getting this error when I try to run kamctl: ERROR: Error opening Kamailio's FIFO FIFO ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "FIFO")' in your config ERROR: and also have loaded the mi_fifo module. I am able to start Kamailio with Kamailio start I have poured over all the responses on Google for two days trying to fix this error. I have tried everything I know of. I have tried each example. kamctlrc DB Path is correct: # database path used by dbtext or db_berkeley DB_PATH="/usr/share/kamailio/dbtext" ## control engine: FIFO or UNIXSOCK ## - default FIFO CTLENGINE="FIFO" ## path to FIFO file OSER_FIFO="FIFO" (I have already tried commenting this out) kamailio.cfg loadmodule "mi_fifo.so" Path to modules is correct: mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio.fifo") modparam("mi_fifo", "fifo_mode",0666) This is installed on Ubuntu Server 10.10 64 bit Intel . Can someone please help? Kurt A. Mullen Practical PC, LLC (O) 830.542.4102 x204 (F) 210.767.3912

3 2

compiling the SER router
by rbond1 05 Dec '10

05 Dec '10

Hello. I am new to this forum. Can someone point me in the correct direction regarding compiling the SER package from the sip-router git repository? When I try to compile the binaries, it is having troubles finding the mysql.h header file. I have looked in some of the obvious places, but it is not installed either in the distro I for mysql nor in the sip-router directory's module folders. Here is the error I see in the make_and_install_output.txt file when I try to compile. km_dbase.c:38: fatal error: mysql/mysql.h: No such file or directory compilation terminated. make[1]: *** [km_dbase.o] Error 1 make: *** [modules] Error 1 I am using the current ubuntu server distro. Is this a well known issue? I followed the download instructions from the sip-router download page from this website so I assume it automatically pulled the most current release (?). Will I need to load a package from someplace else? Mysql is currently installed onto my machine. But I cannot find mysql.h. WIll I need to get the libs from mysql (if available) and will they include the header files? thx in adv

2 1

Kamailio Authentication
by Kurt Mullen 05 Dec '10

05 Dec '10

I have Kamailio up and running and passing calls between users. However, it is not authenticating. Therefore any one can access without user/password. Kamailio.cfg #!- define WITH_MYSQL # # *** To enable authentication execute: #!- enable mysql #!- define WITH_AUTH #!- add users using 'kamctl' # # *** To enable IP authentication execute: #!- enable mysql #!- enable authentication #!- define WITH_IPAUTH #!- add IP addresses with group id '1' to 'address' table Kurt A. Mullen Practical PC, LLC (O) 830.542.4102 x204 (F) 210.767.3912

2 1

[Kamailio]: INVITE Authorization
by Spinov Evgeniy 03 Dec '10

03 Dec '10

Hello. I have Kamailio ( K in further ) and 2x Asterisk boxes ( A1 and A2 in further ) configured, so UAC registers at K and when it sends a call, it's routed to A1 or A2, balanced. The problem is, that I cannot find how to authorize INVITE requests, so unregistered UAC could not send INVITE requests. Simply cannot find anything. I'm making registration, using www_authorize() and checking all INVITES with proxy_authorize(). Just after kamailio is started - everything works fine and as planned: registered UAC can call and not registered - cannot. But after aproximately 40 seconds everything is stopped. Not calls passed and everybody receives 407 Proxy Authorization is required. So, the question: how it is correctly to verify that incoming INVITE on K is authorized? It seems to me that I'm doing that in wrong way. Thank you.

2 3

using forward() with variables
by Pan B. Christensen 03 Dec '10

03 Dec '10

Hello, We are running Kamailio 1.5. I'm trying to use the forward() function with a variable like this: forward($shv(voip1-edge1)); That fails to parse. I then tried this, which I thought would fail: forward("$shv(voip1-edge1)"); That parsed correctly, but it then failed to resolve the domain '$shv(voip1-edge1)' as expected. Why can this function not use variables? I tried to make a workaround by doing this: $du=$ru; $dd=$shv(voip1-edge1); forward(); That also failed. Why is $du writable but $dd is not? I then made a fourth attempt, which seems to work: $du="sip:" + $shv(voip1-edge1); forward(); With kind regards, Pan

2 3

Kamailio Help
by Saurabh Srivastava 03 Dec '10

03 Dec '10

Hello Kamailio (OPENSER) Team, We have successfully installed kamailio on debian flavor of LINUX as per your documentation states. After installation kamailio is running on the server and we are trying to register a SIP User Agent to kamailio SERVER. But when we try to register, the server is returning these set of messages to us. 1st Message : 401 Unauthorized 2nd Message : 200 OK 3rd Message : 401 Not Here Looks like we are missing something in the configuration file for kamailio. Can you please guide us through the process so that we can get this working. I am attaching my kamailio.cfg file with the mail as well as the ethereal captured log file in format of .jpg file with the mail. We also want kamailio to listen on multiple ports such as 5060 and 5062 and we have made according changes in the configuration file as well. Please have a look at those as well and tell us whether we are doing it right or not. Also if you feel that we need to do some additional changes in configuration file. Please suggest them. PS: Attachments (EthCap.jpg and kamailio.cfg) ******************************************************************************************************************************************************************************************** #!KAMAILIO ################################ #Added By Saurabh# #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB ################################ # # Kamailio (OpenSER) SIP Server v3.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!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_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 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 reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* 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:10.0.0.10:5060 listen=udp:XXX.XXX.XXX.XXX:5060 #XXX is ip address of server listen=udp:XXX.XXX.XXX.XXX:5062 #XXX is ip address of server /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 port=5062 #!ifdef WITH_TLS enable_tls=yes #!endif ####### 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 # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!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 "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar 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) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* 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", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } ******************************************************************************************************************************************************************************************** Thanks and Best Regards Saurabh Srivastava

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Kamailio v3.1.1 Released
by Daniel-Constantin Mierla 02 Dec '10

02 Dec '10

Hello, Kamailio SIP Server v3.1.1 is released. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.0. There is no change to database or configuration file required to upgrade to 3.1.1 from 3.1.0 version. If you run v3.1.0 from tarball or packages, it is strongly recommended to upgrade to v3.1.1. For more details about version 3.1.1, visit: http://www.kamailio.org/w/2010/12/kamailio-v3-1-1-released/ Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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When too many calls, calls does not complete, too much delay when placing calls
by Vic Jolin 02 Dec '10

02 Dec '10

Hi all, Just want to ask if anybody here has come across this problem of mine, when single call proxy is ok, but when you send more traffic to it, calls somehow get stuck, trying to make a new call it takes a lot of time and so it times out. Does anybody know what could be the problem? Thanks!

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limit on fail-over in lcr?
by Anders 02 Dec '10

02 Dec '10

Hi there, Kamailio 1.5: I cannot find an answer to this in the documentation: Is there a limit to how many fail-overs Kamailio does? I have lcr setup for a destination where it is rejected with 403 on the first and second priority, but then it goes no further, even though I have a third priority defined. Is there a limit to fail-over on two levels? Thanks!! //Anders

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Re: [SR-Users] SIP between IPv4 and IPv6 agents
by Klaus Darilion 02 Dec '10

02 Dec '10

Am 02.12.2010 13:06, schrieb Komáromi Péter: > Hi! > > So if you say it is possible to solve the problem with the only > location table, the location_inet4 and location_inet6 is not > certainly necessary... do I _have to_ use the 4to6.cfg file from the > source of kamailio, or not? When you call force_rtp_proxy() then you have to provide the i/e flags to tell rtpproxy how it should bridge the call. Therefore you need to know for every call: - is the caller IPv4 or IPv6? - is the callee IPv4 or IPv6? Finding out the protocol version of the caller is rather easy: if (af == inet) { # caller ipv4 ... } else { # caller ipv6 ... } But finding out if the callee is v4 or v6 is difficult. Actually it may even be that the callee is registered with 2 clients at the same time - one using v4 and v6. As far as I know this multi-registration scenario can not be handled currently - only single-registrations or multi-registrations which use the same protocol version are supported. Anyway - there are several approaches how to detect the IP version of the callee. - fixed mapping: for example if you know that all usernames starting with 1 are IPv4 clients, and all usernames starting with 2 are IPv6 clients, then you can detect the callee's protocol version manually, e.g: if ( $(rU{s.substr,0,1}) == "1") { # callee ipv4 ... } else { # callee ipv6 ... } - check destination URI if it contains a IPv4 or IPv6 address, e.g. use http://www.kamailio.org/dokuwiki/doku.php/transformations:devel#resubst_exp… to test if $dd contains '.' which would indicate a ipv4 address. - use dedicated location tables > (Because it uses them, for instance here: if (method == "REGISTER") > { if (af == inet) { save("location-inet4"); } else if (af == inet6) > { save("location-inet6");...) > > If I do not have to use it, I do not need to merge it with my > kamailio.cfg file, the steps I have to do are: install rtpproxy, > start it with the appropriate parameters and set the forcing in the > kamailio.cfg file?! The more I think about it I prefer the approach with separated location tables. It should be rather easy to integrate the 2 location tables into your existing config. regards Klaus PS: please Cc the mailing list

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Parallel forking and recording with rtpproxy?
by Jeremya 02 Dec '10

02 Dec '10

Hi, I'm having some issues with recording in rtpproxy when I parallel fork an incoming call. The scenario is that I want to record incoming calls to a small number of operators, and the call is converted to an 'all-call' by the local kamailio instance - parallel forking. It appears - though I could be wrong - that only the call tag is used to differentiate calls in kamailio/rtpproxy. I am guessing that when the call is forked and presented to rtpproxy that it somehow creates/uses only one data structure. When the call is answered, the other calls are cancelled and this results in the data structure being deleted. Is this correct? If so, are there any easy ways to achieve recording in this scenario? Thanks Jeremy

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nathelper force_rtp_proxy etc flag "f"
by Jeremya 02 Dec '10

02 Dec '10

Is the flag "f" of any actual use in force_rtp_proxy, rtp_proxy_offer and rtpproxy_answer? With an upstream proxy the lack of "f" causes problems. Does some problem occur when it is used without an upstream proxy? Thanks Jeremy

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Releasing Kamailio 3.1.1
by Daniel-Constantin Mierla 02 Dec '10

02 Dec '10

Hello, there are almost 2 months since release of Kamailio 3.1.0, so it is time for 3.1.1 as there were several important fixes done. I propose Thursday, next week, by that time we should sort out most of the issues reported lately. Other opinions? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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"invalid version 1 for table trusted found, expected 5" while trying to load permissions and drouting modules togheter
by Pierpaolo Culurciello 01 Dec '10

01 Dec '10

Hello to everybody, I'm facing the following problem while trying to load and use Permissions and Drouting modules togheter: here you are the detailed starting debug of my Kamailio 3.1.0: 0(2456) DEBUG: db_postgres [km_pg_con.c:72]: opening connection: postgres://xxxx:xxxx@192.168.190.150:5432/kamailio 0(2456) DEBUG: db_postgres [km_pg_con.c:80]: PQsetdbLogin(0x93ec520) 0(2456) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (-127): permissions 0(2456) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (-127): ldap 0(2456) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (-127): alias_db 0(2456) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (-127): domain 0(2456) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (-127): presence 0(2456) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (-127): presence_xml 0(2456) DEBUG: presence_xml [presence_xml.c:300]: [-127] pid [2456] 0(2456) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (-127): nathelper 0(2456) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (-127): mediaproxy 1(2457) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (1): mi_fifo 1(2457) DEBUG: <core> [sr_module.c:791]: DEBUG: init_mod_child (1): tm 1(2457) DEBUG: tm [callid.c:131]: DEBUG: callid: '65df7ef6-2457(a)192.168.190.185' 1(2457) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (1): avpops 1(2457) DEBUG: <core> [db.c:294]: connection 0x82a7358 found in pool 1(2457) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (1): usrloc 1(2457) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (1): registrar 1(2457) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (1): uri_db 1(2457) DEBUG: <core> [db.c:294]: connection 0x82a7408 found in pool 1(2457) DEBUG: <core> [sr_module.c:791]: DEBUG: init_mod_child (1): ctl 1(2457) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (1): acc 1(2457) DEBUG: <core> [db.c:294]: connection 0x82bc668 found in pool 1(2457) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (1): drouting 1(2457) DEBUG: <core> [db.c:294]: connection 0x82bc910 found in pool 1(2457) DEBUG: db_postgres [km_dbase.c:149]: 0x82bc8f8 PQsendQuery(select gwid,address,strip,pri_prefix,type,attrs from dr_gateways ) 1(2457) DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82a74e8 2(2458) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (2): mi_fifo 2(2458) DEBUG: <core> [sr_module.c:791]: DEBUG: init_mod_child (2): tm 2(2458) DEBUG: tm [callid.c:131]: DEBUG: callid: '65df7ef6-2458(a)192.168.190.185' 2(2458) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (2): avpops 2(2458) DEBUG: <core> [db.c:294]: connection 0x82a7358 found in pool 2(2458) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (2): usrloc 2(2458) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (2): registrar 2(2458) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (2): uri_db 2(2458) DEBUG: <core> [db.c:294]: connection 0x82a7408 found in pool 2(2458) DEBUG: <core> [sr_module.c:791]: DEBUG: init_mod_child (2): ctl 2(2458) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (2): acc 2(2458) DEBUG: <core> [db.c:294]: connection 0x82bc668 found in pool 2(2458) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (2): drouting 2(2458) DEBUG: <core> [db.c:294]: connection 0x82bc910 found in pool 2(2458) DEBUG: <core> [sr_module.c:807]: DEBUG: init_mod_child (2): permissions 2(2458) DEBUG: <core> [db.c:294]: connection 0x82a7500 found in pool 2(2458) DEBUG: db_postgres [km_val.c:158]: PQescapeStringConn: in: 7 chars, out: 7 chars 2(2458) DEBUG: db_postgres [km_dbase.c:149]: 0x82a74e8 PQsendQuery(select table_version from version where table_name='trusted') 2(2458) DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82a7598 LOG: statement: select gwid,address,strip,pri_prefix,type,attrs from dr_gateways 2(2458) DEBUG: db_postgres [km_dbase.c:403]: 0x82a74e8 PQresultStatus(PGRES_TUPLES_OK) PQgetResult(0x93f6010) 2(2458) DEBUG: db_postgres [km_res.c:108]: 6 columns returned from the query 2(2458) DEBUG: <core> [db_res.c:153]: allocate 24 bytes for result names at 0x82a75c0 2(2458) DEBUG: <core> [db_res.c:163]: allocate 24 bytes for result types at 0x82a75e0 2(2458) DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[0] at 0x82a7600 2(2458) DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0x82a7600)[0]=[gwid] 2(2458) DEBUG: db_postgres [km_res.c:140]: use DB1_INT result type 2(2458) DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[1] at 0x82a7610 2(2458) DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0x82a7610)[1]=[address] 2(2458) DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type 2(2458) DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[2] at 0x82a7620 2(2458) DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0x82a7620)[2]=[strip] 2(2458) DEBUG: db_postgres [km_res.c:140]: use DB1_INT result type 2(2458) DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[3] at 0x82a7630 2(2458) DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0x82a7630)[3]=[pri_prefix] 2(2458) DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type 2(2458) DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[4] at 0x82a7640 2(2458) DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0x82a7640)[4]=[type] 2(2458) DEBUG: db_postgres [km_res.c:140]: use DB1_INT result type 2(2458) DEBUG: db_postgres [km_res.c:126]: allocate 8 bytes for RES_NAMES[5] at 0x82a7650 2(2458) DEBUG: db_postgres [km_res.c:133]: RES_NAMES(0x82a7650)[5]=[attrs] 2(2458) DEBUG: db_postgres [km_res.c:166]: use DB1_STRING result type 2(2458) DEBUG: db_postgres [km_res.c:222]: allocate for 6 columns 24 bytes in row buffer at 0x82a7660 2(2458) DEBUG: <core> [db_res.c:182]: allocate 8 bytes for rows at 0x82a7680 2(2458) DEBUG: db_postgres [km_res.c:242]: PQgetvalue(0x82a74e8,0,0)=[1] 2(2458) DEBUG: db_postgres [km_res.c:251]: [0][0] Column[gwid]=[1] 2(2458) DEBUG: db_postgres [km_res.c:242]: PQgetvalue(0x82a74e8,0,1)=[192.168.190.150] 2(2458) DEBUG: db_postgres [km_res.c:251]: [0][1] Column[address]=[192.168.190.150] 2(2458) DEBUG: db_postgres [km_res.c:242]: PQgetvalue(0x82a74e8,0,2)=[2] 2(2458) DEBUG: db_postgres [km_res.c:251]: [0][2] Column[strip]=[2] 2(2458) DEBUG: db_postgres [km_res.c:242]: PQgetvalue(0x82a74e8,0,3)=[88] 2(2458) DEBUG: db_postgres [km_res.c:251]: [0][3] Column[pri_prefix]=[88] 2(2458) DEBUG: db_postgres [km_res.c:242]: PQgetvalue(0x82a74e8,0,4)=[0] 2(2458) DEBUG: db_postgres [km_res.c:251]: [0][4] Column[type]=[0] 2(2458) DEBUG: db_postgres [km_res.c:242]: PQgetvalue(0x82a74e8,0,5)=[] 2(2458) DEBUG: <core> [db_row.c:119]: allocate 120 bytes for row values at 0x82c17f8 2(2458) DEBUG: <core> [db_val.c:73]: converting INT [1] 2(2458) DEBUG: <core> [db_val.c:117]: converting STRING [192.168.190.150] 2(2458) DEBUG: <core> [db_val.c:127]: allocate 16 bytes memory for STRING at 0x82a7690 2(2458) DEBUG: <core> [db_val.c:73]: converting INT [2] 2(2458) DEBUG: <core> [db_val.c:117]: converting STRING [88] 2(2458) DEBUG: <core> [db_val.c:127]: allocate 3 bytes memory for STRING at 0x82c1878 2(2458) DEBUG: <core> [db_val.c:73]: converting INT [0] 2(2458) DEBUG: <core> [db_val.c:56]: converting NULL value 2(2458) DEBUG: db_postgres [km_res.c:266]: freeing row buffer at 0x82a7660 2(2458) DEBUG: db_postgres [km_dbase.c:302]: PQclear(0x93f6010) result set 2(2458) DEBUG: <core> [db_res.c:79]: freeing 6 columns 2(2458) DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[0] at 0x82a7600 2(2458) DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[1] at 0x82a7610 2(2458) DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[2] at 0x82a7620 2(2458) DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[3] at 0x82a7630 2(2458) DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[4] at 0x82a7640 2(2458) DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[5] at 0x82a7650 2(2458) DEBUG: <core> [db_res.c:92]: freeing result names at 0x82a75c0 2(2458) DEBUG: <core> [db_res.c:97]: freeing result types at 0x82a75e0 2(2458) DEBUG: <core> [db_res.c:52]: freeing 1 rows 2(2458) DEBUG: <core> [db_row.c:68]: free VAL_STRING[1] '192.168.190.150' at 0x82a7690 2(2458) DEBUG: <core> [db_row.c:68]: free VAL_STRING[3] '88' at 0x82c1878 2(2458) DEBUG: <core> [db_row.c:97]: freeing row values at 0x82c17f8 2(2458) DEBUG: <core> [db_res.c:60]: freeing rows at 0x82a7680 2(2458) DEBUG: <core> [db_res.c:134]: freeing result set at 0x82a7598 2(2458) ERROR: <core> [db.c:408]: invalid version 1 for table trusted found, expected 5 (check table structure and table "version") 2(2458) ERROR: permissions [trusted.c:249]: error during table version check. 2(2458) DEBUG: <core> [db_pool.c:97]: connection still kept in the pool 2(2458) ERROR: <core> [sr_module.c:811]: init_mod_child(): Error while initializing module permissions 2(2458) ERROR: <core> [pt.c:332]: ERROR: fork_process(): init_child failed for process 2, pid 2458, "" 2(2458) : <core> [main.c:1439]: main_loop: Cannot fork As you can see expecially from the extract reported below, the problem seems to occur when multiple threads run at the same time and placing queries almost at the same time and it happens that the results of those queries are returned back to the other thread. 2(2458) DEBUG: db_postgres [km_val.c:158]: PQescapeStringConn: in: 7 chars, out: 7 chars 2(2458) DEBUG: db_postgres [km_dbase.c:149]: 0x82a74e8 PQsendQuery(select table_version from version where table_name='trusted') 2(2458) DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82a7598 LOG: statement: select gwid,address,strip,pri_prefix,type,attrs from dr_gateways 2(2458) DEBUG: db_postgres [km_dbase.c:403]: 0x82a74e8 PQresultStatus(PGRES_TUPLES_OK) PQgetResult(0x93f6010) 2(2458) DEBUG: db_postgres [km_res.c:108]: 6 columns returned from the query Here thread number 2 executes the query "select table_version from version where table_name='trusted'" but then the same thread doesn't evaluate its result but the one of the previous query, "select gwid,address,strip,pri_prefix,type,attrs from dr_gateways"; as a matter of fact you can see in the LOG statement "6 columns returned from the query". The strange thing is that this problem occurs only when I try to load Permissions and Drouting modules at the same time: whenever I load anyone of that two modules alone Kamailio starts whitout complaining about anything. Obviously I have already checked the value of "table_version" in version table and it's the correct one (5). Moreover I have also tried to apply the patch suggested in the following thread for a similar problem, (http://www.mail-archive.com/sr-dev@lists.sip-router.org/msg07551.html) but without any success. Please let me know if you have some advice for my problem, I really need to use these two modules at the same time! Greetings, Pierpaolo Culurciello

2 2

CANCEL with Totag
by Iñaki Baz Castillo 01 Dec '10

01 Dec '10

Hi, Samsung PBX's generate a wrong CANCEL: - It contains To tag !!! - It contains "application/sdp" mirroring the SDP of the 183 received !!! Really ugly. Actually those CANCEL's are being rejected by my kamailio (403) as they contain Totag but no Route header. So, as a workaround, I will move the section of CANCEL before the loose-routing section. I expect it will work, but I would like to be sure that Kamailio (1.5) will not react due to the Totag of the CANCEL (it shouldn't IMHO as the Totag has no relationship with the INVITE transaction being cancelled). Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 11

Kamailio 3.1 doesn't start with siptrace module
by vonlouda 01 Dec '10

01 Dec '10

Hi, I've installed a new instance of Kamailio 3.1 without prb and it works fine. However, in order to activate siptrace, I've added the following lines in the config file, loadmodule "siptrace.so" modparam("siptrace","db_url", DBURL) modparam("siptrace","table","sip_trace") modparam("siptrace","trace_on",1) modparam("siptrace","trace_flag",13) I had the following message in kamailio logs: Dec 1 17:17:55 km36133 /usr/sbin/kamailio[5522]: INFO: usrloc [hslot.c:53]: locks array size 512 Dec 1 17:17:55 km36133 kernel: [10765022.447713] kamailio[5522]: segfault at 0 ip 0 sp 7fffffffc748 error 4 in kamailio[400000+1f7000] Dec 1 17:17:55 km36133 kamailio: ERROR: <core> [daemonize.c:269]: Main process exited before writing to pipe Is it a known issue? Do you have an idea how to fix that, or at least how to activate siptrace module without craching Kamailio. Regards.

1 0

memory debugging
by Jason Penton 01 Dec '10

01 Dec '10

Hi All, thought I would give you a small little perl program I quickly hacked up to help with some memory leaks I was getting. Turns out I was forgetting to call release_urecord after a get_urecord on the usrloc module. Because I use Kamailio in DBOnly mode - the memory was never being freed. This little program helped me to find the source in no time at all. I hope it helps you too. A little info: The program works on the log file. NB you have to have memlog debugging enabled for this program to do anything useful. Effectively the program continuously monitors the log file looking for allocs and frees. It tallies all the fragments in use and gives a report of all the fragments and where they weer allocated. It also keeps a running counter of bytes used. Here is an example of the output: ***************************************************************************************************************************************************** DUMPING ALL UNFREED MEMORY FRAGMENTS (25) ***************************************************************************************************************************************************** called from h_table.c: build_cell(251) 6 21312 called from dlg.c: new_dlg_uac(169) 1 128 called from t_msgbuilder.c: build_uac_req(921) 6 1896 called from pv_svar.c: set_var_value(104) 3 132 called from ../../ut.h: shm_str_dup(501) 2 48 called from reply.c: build_contact(127) 1 132 called from t_hooks.c: insert_tmcb(108) 6 144 ************************************************************TOTAL BYTES HELD IN FRAGMENTS: 23792*************************************************************** To use - simply untar into a directory and run ./memalyze.pl Hope it helps Cheers Jason

1 0

Kamailio Solaris Install
by Sherker, Christa 01 Dec '10

01 Dec '10

I am getting the following error when attempting the compile Kamailio on Sun 4140 (Intel) running Solaris 10 . Has anyone encountered this error before? cfg.lex: In function `yylex': cfg.lex:557: error: `yylval' undeclared (first use in this function) cfg.lex:557: error: (Each undeclared identifier is reported only once cfg.lex:557: error: for each function it appears in.) cfg.lex:557: error: `FORWARD' undeclared (first use in this function) cfg.lex:558: error: `FORWARD_TCP' undeclared (first use in this function) cfg.lex:559: error: `FORWARD_TLS' undeclared (first use in this function) cfg.lex:560: error: `FORWARD_SCTP' undeclared (first use in this function) cfg.lex:561: error: `FORWARD_UDP' undeclared (first use in this function) cfg.lex:562: error: `DROP' undeclared (first use in this function) cfg.lex:563: error: `EXIT' undeclared (first use in this function) cfg.lex:564: error: `RETURN' undeclared (first use in this function) cfg.lex:565: error: `BREAK' undeclared (first use in this function) cfg.lex:566: error: `SEND' undeclared (first use in this function) cfg.lex:567: error: `SEND_TCP' undeclared (first use in this function) cfg.lex:568: error: `LOG_TOK' undeclared (first use in this function) cfg.lex:569: error: `ERROR' undeclared (first use in this function) cfg.lex:570: error: `SETFLAG' undeclared (first use in this function) <Truncated cgf.lex errors> gmake: *** [lex.yy.o] Error 1 Thanks, -Christa ________________________________ CONFIDENTIALITY NOTICE: This e-mail may contain information that is privileged, confidential or otherwise protected from disclosure. If you are not the intended recipient of this e-mail, please notify the sender immediately by return e-mail, purge it and do not disseminate or copy it.

2 2

VIA header x-route-tag
by alex pappas 01 Dec '10

01 Dec '10

Dear All, We have some cisco gw that they send to the Kamailio in the first invite and inside the VIA header an x-route-tag="The Gateway that the call entered". I'm trying to add this to the acc module in order to have it recorded but without sucess. My config is this: if (search("x-route-tag")) { avp(s:xroutetag) = $sel(@via); xlog("----- We have a x-route-tag in the SIP msg avp: $avp(s:xroutetag) -----"); }; I'm trying to add the VIA header to an avp so I could manipulate the string but i cannot implement this. Does anyone can suggest a way to add the *x-route-tag* value to an AVP? Thank you in advance Alex sample VIA header: Via: SIP/2.0/UDP 10.0.0.15:5060;* x-route-tag="tgrp:TANDEM-KOR"*

3 9

Redirect w/ Xlite?
by Eric Hiller 01 Dec '10

01 Dec '10

Hello all, I have seen some other complaining about this issue, but the posts were over 4 years ago so I would think xlite would be fixed at this point. I wanted to ensure my formatting was correct before I started trying other phones. All I am trying to do at the moment is a simple redirect. My config is: xlog("tu= $tu || $(tU{s.select,0,-}) \n");rewritehost("pbx.newhost.com");xlog("host is now $rd; all is $ru\n");sl_send_reply("300","Redirect");exit; Insanely simple, however xlite is not redirecting the actual ip of the next invite/register/whatever. It only changes the RURI, but not the actual ip. Per the RFC, it looks like I am doing this right, the new host is in the contact field of the 300 and the client should then redirect its next packet to the new host. Any ideas what I am doing wrong here? Or is this a known redirect issue. And given that, how supported is redirect? Should any system be based upon it? Thanks for the help!-Eric

3 5

too many calls, no calls going thru
by Vic Jolin 30 Nov '10

30 Nov '10

Hi ALL, Just wanted to ask why is it that when I have too many calls, there is no call going thru at all? I mean if there are too many calls coming in, I could not make any other call. Is it my config?

1 0

has_sdp() missing
by Jeremya 30 Nov '10

30 Nov '10

Hi, I've been following some examples using has_sdp() function but it is not supplied by 3.0.x TEXTOPS I've seen some discussion on mailing lists about has_sdp() being included since 1.5. But checking the online documentation is doesn't exist in any version. I use has_body("application/sdp") as an alternative but perhaps the has_sdp() function could be included, or references and examples using it altered or removed? Thanks Jeremy

3 3

Re: [SR-Users] SIP between IPv4 and IPv6 agents
by Komáromi Péter 30 Nov '10

30 Nov '10

Hi again, My kamailio 1.5 is working well and I'm able to create the session between an IPv4 and an IPv6 UA, but there is no RTP session. That's the point where the rtpproxy comes in. I installed it, and made a bridging (I hope it is): /usr/local/bin/rtpproxy -F -l /192.168.124.146 -6 FEC0:0:0:0:0:0:0:2 and started rtpproxy -F start. But there is no RTP traffic yet. I read here ( https://www1.ethz.ch/id/people/allid_list/armin/SIP-IPv4-IPv6.pdf ) I have to make two tables named â01Clocation _inet4â01D and â01Clocation_inet6â01D instead of the only â01Clocationâ01D. If it's true, were can I add these? Because it says: "If mySQL is used, the mySQL initialization script (/usr/local/sbin/ser_mysql.sh) has to be modified to create two additional tables" But there is no /usr/local/sbin/ser_mysql.sh on my computer. Where else can I add these tables? Thanks, Peter

2 1

Tutorial: FreeSwitch as Media Server and SBC for Kamailio 3.1
by Daniel-Constantin Mierla 30 Nov '10

30 Nov '10

Hello, a short note to inform that I updated my tutorial about using FreeSwitch and Kamailio together for large VoIP platforms. Besides upgrade to use latest Kamailio major stable release, v3.1.0, there are couple of new features added in the architecture of the VoIP platform: - along with providing media services (voicemail, conferencing, a.s.o.), FreeSwitch is used now also as SBC for topology hiding and media relay (this helps for transconding needs, palying audio messages during the early session or NAT traversal) - Kamailio config has added features to detect DoS/DDoS and scanning attacks, secure SIP communication over TLS, IP authentication and a bunch of neat things that help for an easier maintenance and update of parameters such as database connectivity details, local IPs, etc. The link to tutorial is: * http://asipto.com/u/kfsb Hope it will be useful for many of you. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

rtpproxy_offer() and rtpproxy_answer() missing in 3.0.x
by Jeremya 30 Nov '10

30 Nov '10

http://kamailio.org/docs/modules/3.0.x/modules_k/nathelper.html refers to functions rtpproxy_offer() and rtpproxy_answer() But I get an error when using them in my script. Are these not available in 3.0.x? Regards Jeremy

4 6

Re: [SR-Users] Settings adjustments
by Javier Gallart 30 Nov '10

30 Nov '10

Hi thanks for the reply, I will try with those values... Javi On Mon, Nov 29, 2010 at 6:24 PM, Alex Balashov <abalashov(a)evaristesys.com>wrote: > On 11/29/2010 11:57 AM, Javier Gallart wrote: > > Could these settings cause any "collateral damage"? >> > > Not really. They will result in increased memory usage on a small scale; > they are preprocessor constants that are used to give static sizes to > character arrays inside dynamically allocated structures. But it shouldn't > be a big deal. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 1170 Peachtree Street NE > 12th Floor, Suite 1200 > Atlanta, GA 30309 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ >

1 0

Fwd: Need Clarification on sst module
by Suresh Arunachalam 30 Nov '10

30 Nov '10

Hi Alex, Thanks for your reply, I had same doubt as you, I have double checked the script for exit call. I do not have copy of the configuration to prove myself, for your information I just copied the example from sst module readme file. Thanks Suresh On Mon, Nov 29, 2010 at 9:32 PM, Alex Balashov <abalashov(a)evaristesys.com>wrote: > On 11/29/2010 06:57 AM, Suresh Arunachalam wrote: > > i have configured kamailio server with sst module and server is sending >> 422 error for Min-SE negotiation. Along with 422 response I see 404 >> response coming from the server which is causing my application to >> remove the transaction. Please let know am I missing something in the >> configuration, let me know the fix for the same. >> > > That would be difficult to determine without seeing the configuration. > > My guess is that at the point in the script where the 422 response is sent, > you do not have an 'exit' or 'return' statement, and so the script keeps > executing until it also reaches a 404 stateless reply. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 1170 Peachtree Street NE > 12th Floor, Suite 1200 > Atlanta, GA 30309 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.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 >

1 1

Failover with SRV records
by Eric Hiller 30 Nov '10

30 Nov '10

Sorry all for the second question of the night as I go through working out bugs in my setup. I want to use SRV records to loadbalance across hosts. This works great, when all the hosts listed in the SRV are up. However, I want kamailio to use the next host if the current hosts fails. I tried just setting up a new branch, but kamailio keeps using the same SRV entry over and over on that transaction, only a new transaction seems to give kamailio a chance to use a different SRV entry. Any ideas as to how I can force kamailio to try the next SRV entry if the first one fails? For example I t_relay to 2.domain.com which has a _sip._udp.2.domain.com entry for hostA and hostB. If hostB is down, but kamialio decides to send to hostB it just keeps doing so even though hostA is perfectly up it doesn't try for it. Is there a way to remove hostB from the kamailio try list? Or as an alternative to do a manual SRV lookup on 2.domain.com and then put then in a variable and go through them one by one manually? Ideas? Thanks-Eric

3 5

Settings adjustments
by Javier Gallart 29 Nov '10

29 Nov '10

Hello I have a Kamailio acting as Redirect server. In some cases I need to include more that 12 Contact headers in the 302 message. I've set these values in the config.h file. #define MAX_BRANCHES 20 /*!< maximum number of branches per transaction */ #define MAX_REDIRECTION_LEN 756 /*!< maximum length of Contact header field in redirection replies */ Could these settings cause any "collateral damage"? Thanks! Javi

2 1

SIP between IPv4 and IPv6 agents
by Komáromi Péter 29 Nov '10

29 Nov '10

Hello! I write my diplomawork, and kindly ask for your help! I would establish a videoconference with SIP protocol between two PCs, one of them using IPv4 and the other uses IPv6. I have access to an RMX 4000 Multipoint Conference Platform with a Kamailio SIP server on it. Were here on the mailing list any previous discussion about translating between the two IP versions? I read some about RTP Proxies, but did not find the proper solution so far. Could somebody help me please? Best Regards, Peter

4 12

Need Clarification on sst module
by Suresh Arunachalam 29 Nov '10

29 Nov '10

Hi All, i have configured kamailio server with sst module and server is sending 422 error for Min-SE negotiation. Along with 422 response I see 404 response coming from the server which is causing my application to remove the transaction. Please let know am I missing something in the configuration, let me know the fix for the same. Thanks Suresh

2 1

kamailio 3.1.0 - missing packages from rpms repo
by Stephen McVarnock 29 Nov '10

29 Nov '10

Hi, I have been looking for the kamailio mysql package at the following location:- http://www.kamailio.org/dokuwiki/doku.php/packages:rpms the 3.1 link is a dead link though. 3.0.x links are just fine. Can anyone tell me where I can find the 3.1 equivalent of kamailio-mysql-3.0.3-0.i386.rpm <http://www.kamailio.org/pub/kamailio/3.0.3/packages/centos-5.5/kamailio-mys…>. Regards, Steve. -- ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Stephen McVarnock, AePONA Ltd, Interpoint Building, 20-24 York Street, Belfast BT15 1AQ Tel: +44 (0)28 9026 9114 Fax: +44 (0)28 9026 9111 http://www.aepona.com ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

1 0

db_flatstore not loading
by Javier Gallart 25 Nov '10

25 Nov '10

Hi I'd like to test db_flatstore module at 3.1 to write accounting data. The module is not loading though. The config is really simple: loadmodule "db_flatstore.so" modparam("acc", "db_url", "flatstore:/var/log/kamailio/cdr") I'm getting this error: 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_bind_api> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_bind_api> in module <db_flatstore> not found 0(27360) DEBUG: <core> [db.c:210]: using export interface to bind db_flatstore 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_use_table> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_use_table> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_init> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_init> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_close> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_close> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_query> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_query> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_fetch_result> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_fetch_result> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_raw_query> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_raw_query> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_free_result> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_free_result> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_insert> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_insert> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_delete> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_delete> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_update> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_update> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_replace> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_replace> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_last_inserted_id> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_last_inserted_id> in module <db_flatstore> not found 0(27360) DEBUG: <core> [sr_module.c:631]: find_export_record: <db_insert_update> not found 0(27360) DEBUG: <core> [sr_module.c:684]: find_mod_export: <db_insert_update> in module <db_flatstore> not found 0(27360) ERROR: <core> [db.c:79]: module db_flatstore does not export db_use_table function 0(27360) ERROR: acc [acc.c:333]: bind_db failed 0(27360) ERROR: acc [acc_mod.c:459]: failed...did you load a database module? 0(27360) ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module acc (/usr/local/kamailio-3.1/lib/kamailio/modules_k/acc.so) ERROR: error while initializing modules Any idea? Thanks in advance Javier

2 2

SIP Softphone with RPID and PAI
by Uriel Rozenbaum 25 Nov '10

25 Nov '10

Hey Guys, I was wondering if anyone knows some Windows Based softphone to do some tests on my Kamailio 1.5.4 deployment that will allow insertion of custom headers or at least Remote-Party-ID and/or P-Asserted-Identity. Thanks in advance, Uriel

4 6

Kamailio crash
by michel freiha 25 Nov '10

25 Nov '10

Dear Daniel, Do you mean by Kamailio 3.1 branch the git of kamailio 3.1.0 or the future version 3.1.1 that will be published? Regards

1 0

tm.t_uac_wait asynchronous?
by Alex Balashov 24 Nov '10

24 Nov '10

Does tm.t_uac_wait RPC call block a worker process while waiting for a reply, or use some sort of asynchronous callback? 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

Re: [SR-Users] Doubts with Registration
by Daniel-Constantin Mierla 24 Nov '10

24 Nov '10

never write private emails. They will be simply discarded. ngrep gets the traffic before the firewall. Do you have a firewall? start kamailio with log_stderror=yes and debug=5. if you don't see any sip message processing, then sip traffic does not reach the application layer. Cheers, Daniel On 11/18/10 10:39 AM, Danny Dias wrote: > Thanks for the answer Daniel, > > 2010/11/17 Daniel-Constantin Mierla<miconda(a)gmail.com> >> Hello, >> >> On 11/17/10 6:58 PM, Danny Dias wrote: >> >> Hello, >> I'm having problems with my kamailio, my phones are unable to register into the Proxy. I have been checking with ngrep-sip and all i'm receiving is many REGISTER from the phones but never getting some response from the Proxy >> The Proxy has a public IP and the phones are behind NAT. The weir thing is that my logs are not showing any output...and as you can see on the script, i have a section to receive REGISTER requests.... >> if (is_method("REGISTER")) { >> if (!save("location")) { >> sl_reply_error(); >> } >> exit; >> } >> >> If everything is fine, then no log message should be printed if debug parameter is low. > But i should see in the Kamailio logs the answers, right? or i have to > configure this on the scripts? > > My debug is on "3" > >> But if you run in debug mode with a high value and nothing is printed, then the SIP traffic does not reach the>application. A reason for that might be a firewall, do you run one? > The SIP traffic is arriving the machine (at least), i have checked > this with the ngrep on the server running kamailio: > > # > U 2010/11/18 10:26:03.331925 212.230.253.164:43686 -> 212.230.19.191:5060 > REGISTER sip:212.230.19.191 SIP/2.0' > Via: SIP/2.0/UDP 10.134.16.164:43686;branch=z9hG4bK247264420;rport' > From:<sip:101@212.230.19.191>;tag=1251867918' > To:<sip:101@212.230.19.191>' > Call-ID: 2022066534-16466-1(a)BA.BDE.BG.BGE' > CSeq: 4647 REGISTER' > Contact:<sip:101@10.134.16.164:43686>;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B821FA066>"' > Max-Forwards: 70' > User-Agent: Grandstream GXV3140 1.0.1.1' > Supported: path' > Expires: 3600' > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, UPDATE, MESSAGE' > Content-Length: 0' > ' > > # > U 2010/11/18 10:26:05.339925 212.230.253.164:43686 -> 212.230.19.191:5060 > REGISTER sip:212.230.19.191 SIP/2.0' > Via: SIP/2.0/UDP 10.134.16.164:43686;branch=z9hG4bK247264420;rport' > From:<sip:101@212.230.19.191>;tag=1251867918' > To:<sip:101@212.230.19.191>' > Call-ID: 2022066534-16466-1(a)BA.BDE.BG.BGE' > CSeq: 4647 REGISTER' > Contact:<sip:101@10.134.16.164:43686>;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B821FA066>"' > Max-Forwards: 70' > User-Agent: Grandstream GXV3140 1.0.1.1' > Supported: path' > Expires: 3600' > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, UPDATE, MESSAGE' > Content-Length: 0' > > And i never gets a response from Kamailio, for example, a 200 OK, or a > 407 unauthorized... > > What can i do? > >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com > > -- > Ing. Danny Dias > www.DannTEL.net -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

7 16

kamctl get number of subscribers
by Jon Bonilla 24 Nov '10

24 Nov '10

Hi all I'm trying to get the number of registered subscribers in a kamailio 3.0.4 setup. I'm trying to use usrloc's exported function "ul_get_num_users" but I get a 500 error: kamctl fifo ul_get_num_users 500 command 'ul_get_num_users' not available Using get_statistics I get: test3:~# kamctl fifo get_statistics usrloc: usrloc:registered_users = 2 usrloc:location-users = 2 usrloc:location-contacts = 2 usrloc:location-expires = 0 Trying to query for a specific stat, seems like I do not use the correct syntax: test3:~# kamctl fifo get_statistics usrloc:registered_users 404 Statistics Not Found Anyone could help me with the correct syntax for the function and fifo? thanks

3 7

SER proxy drops incoming SIP requests
by Igor Potjevlesch 24 Nov '10

24 Nov '10

Hello, After days of debugging, I'll need some help from the SER specialists to troubleshoot this problem. The SER version in service is: 0.9.9+cvs20090925 It's running on a Red Hat EL 5.3 server. I currently have about 10.000 users registered. I encounter several seconds (1 to 5) without replies from the SIP proxy every 30 seconds. Some requests are delayed, but most of them are not replied at all. At the same time, all the 6 SER childs (forking is enabled) get about 100% used CPU (from "top") After some troubleshooting, I found that the rythm of those deny of service was exactly the time configured as the "timer_interval" parameter of "usrloc" module. When changing the parameter, problem rythm changes as well. What we did so far: - Check DNS and MySQL performances - Moved the number of childs from 6 to 9 - Changed the "timer_interval" ("usrloc" module) from 60 to 900 - Stopped the "natping" feature ("nathelper" module) as it impacted the service as well. Please let me know if the problem has already been observed and what would be the best way to go deeper in the troubleshooting. I need to solve that as soon as possible: some of our users higly suffer of random REGISTER not beeing taken into account. Sincerely, Igor

1 1

Unforcing RTPProxy for a Cancel
by Joe Hart 24 Nov '10

24 Nov '10

I'm trying to figure out if there is some way to identify a call for which an rtpproxy session has been set up when the call is cancelled. I can't check the flag I set for the call, nor can I check the route param I set (which is what I'm doing to figure out when to unforce the media stream for a BYE). If this is not possible, is there any danger in calling unforce_rtp_proxy() for all calls, regardless of whether media proxying has been set up for the call or not? -- Joe Hart Voice Systems Integrator 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 2

How to relay to ITSP or secure gateway.
by Johny Kadarisman Kwan 24 Nov '10

24 Nov '10

Hi there, Is there a way to relay call to secure gateway or ITSP. ie, invite being challange to provide authorization (username/password) Could someone please give me some advice if this is possible. Thanks, Jkwan

2 3

TCP Children and Connection Handling
by Jijo 24 Nov '10

24 Nov '10

Hi All, I'm facing an issue with TCP connection from Proxy(kamailio) to a SIP server on a load test, where kamailio is not processing the message fast enough and which leads socket buffer getting filled up. I have increased the tcp_rd_buf_size but that really didn't help. According to my understanding the each TCP connection is assigned to a TCP children, and TCP children will read until it there is no data for few seconds and return the connection to tcp main process. In this case the TCP connection is to the SIP server which always have some data due to periodic SIP ping ,registration refresh and calls etc. So i believe this connection will be hold always by one children and since that children is processing the message sequentially, it might not be possible to increase the performance on this connection. My question is, If my understanding is correct How do we increase the performance for a particular TCP connection? Does Kamailio has some mechanism to distribute the message from a single TCP connection to different child, when it is overloaded? What is the best way to solve this issue? Thanks in advance. Jijo

1 0

avp_db_query returning wrong data
by Jijo 23 Nov '10

23 Nov '10

Hi All, While doing 5 REGISTER/second, i observed that that avp_db_query is returning data from a different db entry. This happens only with TCP with more than 1 tcp_children. it works fine with UDP. It also work with TCP with tcp_children as 1. Is there any relation between the TCP connections and Postgres in kamailio. I'm using kamilio 3.1.0. How should i debug this issue?? What are the areas need to look? Thanks Jijo

2 1

SIP Routing Logic in Lua
by Daniel-Constantin Mierla 22 Nov '10

22 Nov '10

Hello, during last days I spent some time to extend the native API exported to Lua. Many more functions exported by core and modules can be called from embedded Lua scripts. Note that you get also access to psedo-variable operations and you can call all functions exported by modules that have no fixup pretty safe, via sr.modf(...), without a need to be exported natively. I made an easy-to-do tutorial where all the SIP routing logic is implemented in a Lua script (including authentication, accounting, registrar, user location). You can read it at: http://asipto.com/u/h Therefore, if the configuration file language is not offering everything you need to route your SIP traffic, look at app_lua module and its API, it is a real option right now. Lua is a small and fast embeddable language, the interpreter is linked to and loaded by Kamailio at statup, therefore the performance penalty is not significant (Lua project site is: http://www.lua.org). It is also popular among other SIP telephony projects. Any kind of feedback is appreciated! Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

6 9

Nuno Ribeiro wants to stay in touch on LinkedIn
by Nuno Ribeiro via LinkedIn 22 Nov '10

22 Nov '10

LinkedIn ------------Nuno Ribeiro requested to add you as a connection on LinkedIn: ------------------------------------------ Luis, I'd like to add you to my professional network on LinkedIn. - Nuno Ribeiro Accept invitation from Nuno Ribeiro http://www.linkedin.com/e/-y8k3ym-ggtfgz8y-47/XkppUi8mK5TI9VJsZIpnfBovln4Su… View invitation from Nuno Ribeiro http://www.linkedin.com/e/-y8k3ym-ggtfgz8y-47/XkppUi8mK5TI9VJsZIpnfBovln4Su… ------------------------------------------ Why might connecting with Nuno Ribeiro be a good idea? Nuno Ribeiro's connections could be useful to you: After accepting Nuno Ribeiro's invitation, check Nuno Ribeiro's connections to see who else you may know and who you might want an introduction to. Building these connections can create opportunities in the future. -- (c) 2010, LinkedIn Corporation

1 0

Double-quote with t_uac_dlg
by Kévin L'heveder 22 Nov '10

22 Nov '10

Hi, I have some troubles to send via t_uac_dlg a REGISTER method with an authorization field cause the values like digest, realm...are containing between double-quotes and that produce a parse error in kamailio. I didn't find how can i escape those double-quotes to prevent from this error. Here's the error logged: Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: we have a quoted value, "From: <sip:131@192.168.40.150>To: <sip:131@192.168.40.150>Contact: <sip:131@192.168.40.210:53000>Authorization: Digest username="131", realm="Asterisk/CM-YAHSAT", algorithm=MD5, uri="sip:192.168.40.202", nonce="5c87d7ed", response="fc126aadd3ba6b6ffbf38aea6be02122"Call-ID: b1ab82011346ff36(a)192.168.40.101CSeq: 10036 REGISTERExpires: 4294967295"Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: " found p is "131", realm="Asterisk/CM-YAHSAT", algorithm=MD5, uri="sip:192.168.40.202", nonce="5c87d7ed", response="fc126aadd3ba6b6ffbf38aea6be02122"Call-ID: b1ab82011346ff36(a)192.168.40.101CSeq: 10036 REGISTERExpires: 4294967295"Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: we have reached the end of attr value, p is "131", realm="Asterisk/CM-YAHSAT", algorithm=MD5, uri="sip:192.168.40.202", nonce="5c87d7ed", response="fc126aadd3ba6b6ffbf38aea6be02122"Call-ID: b1ab82011346ff36(a)192.168.40.101CSeq: 10036 REGISTERExpires: 4294967295"Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: attr value <From: <sip:131@192.168.40.150>To: <sip:131@192.168.40.150>Contact: <sip:131@192.168.40.210:53000>Authorization: Digest username=> foundNov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_parse_node: didn't find newline case1 Nov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_parse_node: parse error around 1Nov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_parse_tree: parse error!Nov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_server: failed to parse the MI tree regards,Koon

1 0

SCTP multihoming support in Kamailio
by mosbah abdelkader 19 Nov '10

19 Nov '10

Hello, Does Kamailio supports SCTP multihoming?

2 1

Re: [SR-Users] iphone
by oscar ying 19 Nov '10

19 Nov '10

Daniel, I can log into serweb.iptel.org with my username 'netlink83' and my mobile number is saved in my account. I hope that is enough to have my mobile phone registered. Maybe I should register to iptel again. What numbers can I call via iptel? How do I configure my own pstn gateway? Is there instructions on iptel.org? Thanks, Oscar Ying 650 776 9821 oscar.ying(a)yahoo.com ________________________________ From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: oscar ying <oscar.ying(a)yahoo.com> Cc: kamailio <sr-users(a)lists.sip-router.org> Sent: Thu, November 18, 2010 2:02:18 AM Subject: Re: [SR-Users] iphone On 11/18/10 4:13 AM, oscar ying wrote: >I've tried two different numbers and they both fail. I can make regular >mobile calls but not VoIP calls via my SIP softphone (media5fone). What >do you mean if my phone is registered? > A sip phone register to SIP server with a username and password - that is what I mean with registered. Note that iptel.org does not allow you to call to pstn, unless you have your own pstn gateway configured. You can do voip calls for free. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

3 2

Storing multiple SQL results in htable/mtree
by Robert McGilvray 19 Nov '10

19 Nov '10

Hello, I'm interested in using the htable and/or mtree modules to keep my aliases and did lookups in memory to minimize the amount of SQL lookups that are being performed. It's easy enough to use htable to cache a single value after it is retrieved but what I'm not clear on is how I can do it when SQL returns multiple destinations. Consider the following query and returned results: SELECT username,domain,CAST(q_value AS CHAR) FROM aliases where alias_username = '10001' AND alias_domain = 'us.sip.globeop.com' AND active = '1' ORDER by q_value; +----------+--------------------+-----------------------+ | username | domain | CAST(q_value AS CHAR) | +----------+--------------------+-----------------------+ | 8500 | us.sip.globeop.com | 1.000 | | 3874 | us.sip.globeop.com | 1.000 | | 7050 | us.sip.globeop.com | 1.000 | | 2100 | us.sip.globeop.com | 1.000 | In my existing script I just append the results as branches and all works well but I end up with a lot of unnecessary SQL traffic. I see the htable supports array but I'm not sure how to go about assigning multiple values to a single key to use later. How can I take the above results and store them in memory for use by other transactions? Thanks, Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

2 1

SIP Scanning Attacks Experiences
by Daniel-Constantin Mierla 19 Nov '10

19 Nov '10

Hello, during the testing period of Kamailio 3.1.0, while running it at voipuser.org, I had the chance to watch live and analyze a SIP scanning attack. Yesterday I noticed another one by looking at Siremis 2.0 charts, therefore I wrote an article with some hints about what you can use to protect your SIP services within Kamailio configuration file. You can read it at: * http://asipto.com/u/i Hope is going to be useful for many of you! Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

6 8

iphone
by oscar ying 18 Nov '10

18 Nov '10

Will my iptel SIP work with iPhone 3G and iPhone app Media5fone? Thanks, Oscar Ying 650 776 9821 oscar.ying(a)yahoo.com

2 5

Doubts with Registration
by Danny Dias 17 Nov '10

17 Nov '10

Hello, I'm having problems with my kamailio, my phones are unable to register into the Proxy. I have been checking with ngrep-sip and all i'm receiving is many REGISTER from the phones but never getting some response from the Proxy The Proxy has a public IP and the phones are behind NAT. The weir thing is that my logs are not showing any output...and as you can see on the script, i have a section to receive REGISTER requests.... if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } Thanks in advance for your help

2 1

Kamailio 1.5.4 - IP addresses WhiteList
by Miguel Baptista 17 Nov '10

17 Nov '10

Hi all, I want to enable pike module based on IP addresses. I mean ... I don't want to check if requests from some specific IP addresses trigged the Pike's threshold. but for all other IP addresses I want to check it. I was thinking about using some kind of IP addresses' whitelist. The userblacklist module has some whitelist but it's based on username/domain not IP addresses. Can someone please give me some advice how to implement it properly. Is there any module that will allow me to do it properly? Best Regards, Miguel Baptista

3 3

Dispatcher is using gateways in Probing mode
by alex pappas 17 Nov '10

17 Nov '10

Dear all, Does anyone can explain how Dispatcher module is sending calls to the gateways in the different states? I mean if I have 3 gateways and the one of them is in Probing mode why Dispatcher is continuing sending there calls? Why this gateway is not changing from Probing to Inactive since is not replaying to OPTIONS messages send by Kamailio? Configuration: modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshhold", 2) modparam("dispatcher", "dst_avp", "$avp(s:dst_avp)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(i:202)") modparam("dispatcher", "ds_append_branch", 0) using db for gw: setid 123,destination sip:xxx.xxxx.xxx, flag 0 ... in ROUTE: if(ds_select_domain("$avp(s:disp_dstgrp)", "4")) in Failure Route if(!ds_next_domain()) I want Dispatcher when it sees a node gateway that is dead , to Not send any other calls. Thank you in advance Alex

3 7

Creating multiple Accounts on kamailio with script.
by Siddhardha Garige 16 Nov '10

16 Nov '10

Hello all, I am conducting some load testing and need to create 50,000 accounts on kamailio. I want to write a shell script to add 50,000 users. Kamctl add [user] [password] prompts for a password and I checked kamctl -help and there is no option for providing password along with the command. Is is possible to disable password and run my script? Thanks Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

3 2

Regarding SER Listening on Custom Port
by Saurabh Srivastava 16 Nov '10

16 Nov '10

Hello SER Team, We have made SER listen on two ports via adding this command (listen=eth0:5060 eth0:5062) in ser.cfg file. Now we are able to get the server listening on two ports as 5060 & 5062 and it is sending 200 OK for the user agents which try to communicate on both the ports. SER is able to register two user agents, one on 5060 and one on 5062. But when we are trying to make a communication between the two registered user agents (i.e. one on 5060 and other on 5062) then no communication is possible (i.e. we cannot make calls and chats). Though when the two user agents are registered on same port (i.e. both UAs on 5060 or both UAs on 5062) then communication is possible between the two clients (i.e. we can make calls and chats). Can you people please guide me through that how can I make the 2 UAs registered to SER on different ports (i.e. one on 5060 and other on 5062) communicate with each other (i.e. they can make calls and chats). I have attached a log as well which shows the same problem. This is a wireshark captured output on the machine which has SER installed. Thanks and Best Regards Saurabh Srivastava

3 2

Handle '486 busy here' from upstream carrier
by JR Richardson 15 Nov '10

15 Nov '10

Hi All, Asterisk><kamailio 3.0.2><sip carrier I see normal tansactions until the carrier send back a '486 Busy Here' then kamailio sends an 'INVITE SDP', then I get a '100 trying' from the carrier then a '482 Loop Detected" from the carrier. Shouldn't kamailio relay the 486 to the original invite server and not send SDP? I'm guessing I'm not handling 486 propperly in my config: http://pastebin.com/crfMe81D Here is a pastebin of the call graph: http://pastebin.com/rnQZDyFU I was thinking about including this in my failure route: if (t_check_status("486")) { append_branch(); t_relay(); } Would that do any good? Thanks. JR -- JR Richardson Engineering for the Masses

4 10

Problem with Kamailio not routing ACK to a 200 OK
by Morten Isaksen 15 Nov '10

15 Nov '10

Hi, I have a strange problem with Kamailio 3.0.2. When one of our end users makes a call Kamailio does not route the ACK (in response to the 200 OK). For all other end users it works fine. For me it looks the the has_totag() checks for some reason fails and then t_check_trans() thinks it is a ACK to a local transactions and then terminates the script. Otherwise there should be more lines in the log file. The conf is pretty standard. route{ xlog("L_WARN", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_WARN", "ua=$ua"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } if (has_totag()) { xlog("L_WARN", "has_totag start - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_WARN", "loose_route - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests #route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("L_WARN", "ACK t_check_trans - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { xlog("Ignoring ACK - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } xlog("L_WARN", "has_totag end - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } setflag(4); t_check_trans(); ... The log files show: Oct 28 10:51:13 sip-core-1 /usr/local/sbin/kamailio[10503]: WARNING: <script>: New request - M=ACK RURI=sip:1105@178.21.248.56:5060 F=sip:87776688@sip.uni-tel.dk T=sip:869 87106(a)sip.uni-tel.dk IP=178.21.248.20 ID=AI231CA9BD0A4A1C53(a)10.0.0.150 Oct 28 10:51:13 sip-core-1 /usr/local/sbin/kamailio[10503]: WARNING: <script>: ua=Aastra Intelligate The trace U 2010/10/28 10:51:02.616337 178.21.248.7:5060 -> 178.21.248.56:5060 INVITE sip:86987106@178.21.248.56 SIP/2.0. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>. Record-Route: <sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. To: <sip:86987106@sip.uni-tel.dk>. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Max-Forwards: 68. Contact: <sip:87776688@87.104.233.108:5060;line=AI7EFC34995E724DD7>. Accept: application/sdp. Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER. P-Preferred-Identity: <sip:87776688@sip.uni-tel.dk>. Privacy: none. User-Agent: Aastra Intelligate. Content-Type: application/sdp. Content-Length: 280. X-trunktype: IC. . v=0. o=intelligate 1194032777 1194032777 IN IP4 87.104.233.106. s=call. c=IN IP4 178.21.248.22. t=0 0. m=audio 60984 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. # U 2010/10/28 10:51:02.636854 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Length: 0. . ##### U 2010/10/28 10:51:04.134445 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:04.134748 178.21.248.7:5060 -> 178.21.248.20:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:04.136586 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:04.136837 178.21.248.7:5060 -> 178.21.248.20:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ############################## U 2010/10/28 10:51:12.881179 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:12.882388 178.21.248.7:5060 -> 178.21.248.20:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ##### <<<<<<<<<<<<<<<<<<<<<< This is the problem packet >>>>>>>>>>>>>>>>>>>>> U 2010/10/28 10:51:13.267863 178.21.248.20:5060 -> 178.21.248.7:5060 ACK sip:1105@178.21.248.56:5060 SIP/2.0. Record-Route: <sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.2. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK07103fe69c8af048b9b8216eb2f7233f. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 ACK. Max-Forwards: 69. Route: <sip:178.21.248.7;lr=on;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>. User-Agent: Aastra Intelligate. Content-Length: 0. -- Morten Isaksen

2 8

append_branch
by Rizwan Qureshi 15 Nov '10

15 Nov '10

Hi All, I just want to know how I can use append_branch() function for register requests. What I want to do is register all users on all asterisk servers by forwarding the register requests (to all asterisk servers) and then if the sip proxy throws call to any of the server the asterisk server, the call should be successful. If its possible then please let me know. Cheers -- Best Regards Rizwan Qureshi

2 1

Re: [SR-Users] advertised_address and RR
by Ovidiu Sas 15 Nov '10

15 Nov '10

On Mon, Nov 15, 2010 at 8:04 AM, Torbjörn Abrahamsson <torbjorn.abrahamsson(a)gmail.com> wrote: > Hello. > > > We are trying to use Kamailio 3.0.3 behind a NAT. After reading the cookbook > we added the advertised_address parameter set to the public IP in our cfg. > This works for the via-header but the record-route-header still uses the > internal IP. Shouldn’t this be changed as well? The documentation says that > advertised_address should apply to RR. advertised_address parameter works for Via headers. > We have also tried to use record_route_preset, but still ends up with the > internal IP in the record-route-header. record_route_preset and record_route are exclusive - use one or the other but not both for the same message Regards, Ovidiu Sas

2 1

advertised_address and RR
by Torbjörn Abrahamsson 15 Nov '10

15 Nov '10

Hello. We are trying to use Kamailio 3.0.3 behind a NAT. After reading the cookbook we added the advertised_address parameter set to the public IP in our cfg. This works for the via-header but the record-route-header still uses the internal IP. Shouldnt this be changed as well? The documentation says that advertised_address should apply to RR. We have also tried to use record_route_preset, but still ends up with the internal IP in the record-route-header. We found some discussions about this when googleing, dating back to 2005. We also found an open-ser-bugreport regarding this. Neither of these resources provided an insight in what we have missed. BR, Torbjörn Abrahamsson

1 0

siremis 2.0.0 CDRS link issues !
by X- x -X 15 Nov '10

15 Nov '10

Hi all, I was wondering if Im the only one getting this error (in siremis 2.0.0 with kamailio 3.1). i.e. I have activated the accounting modules & all but still when I click the "CDR list" in "Accounting Services" it errors on me saying that CDRS table is missing, which ofcourse is missing from the "openser" mysql DB. Why is CDRS table missing ? Is it okay (advisable) to create a CDRS table mentioned in siremis 1.0 docs (http://siremis.asipto.com/install-accounting/ ) Cheers --- X [2010-11-12 14:01:08 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.cdr_id, T0.src_username, T0.src_domain, T0.dst_username, T0.dst_domain, T0.call_start_time, T0.duration, T0.sip_call_id, T0.sip_from_tag, T0.sip_to_tag, T0.src_ip, T0.cost, T0.created FROM `cdrs` T0 LIMIT 10. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'openser.cdrs' doesn't exist Script name and line number of error: /var/www/siremis-2.0.0/openbiz/bin/data/BizDataObj_Lite.php:447 function: _run_search ( Array(2) ) @ /var/www/siremis-2.0.0/openbiz/bin/data/BizDataObj_Lite.php 243 function: fetch ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyForm.php 690 function: fetchDataSet ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/FormRenderer.php 69 function: renderSmarty ( Object(EasyForm), "/var/www/siremis-2.0.0/siremis/modules/ser/template/grid.tpl" ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/FormRenderer.php 45 function: render ( Object(EasyForm) ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyForm.php 1648 function: renderHTML ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyForm.php 1538 function: render ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/ViewRenderer.php 83 function: renderSmarty ( Object(EasyView), "/var/www/siremis-2.0.0/siremis/modules/ser/template/view.tpl" ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/ViewRenderer.php 40 function: render ( Object(EasyView) ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyView.php 274 function: _render ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyView.php 237 function: render ( ) @ /var/www/siremis-2.0.0/openbiz/bin/BizController.php 221 function: renderView ( "ser.view.CdrsListView", "", "", Null, "" ) @ /var/www/siremis-2.0.0/openbiz/bin/BizController.php 107 function: dispatchRequest ( ) @ /var/www/siremis-2.0.0/openbiz/bin/BizController.php 32 function: include_once ( "/var/www/siremis-2.0.0/openbiz/bin/BizController.php" ) @ /var/www/siremis-2.0.0/siremis/bin/controller.php 6 function: include ( "/var/www/siremis-2.0.0/siremis/bin/controller.php" ) @ /var/www/siremis-2.0.0/siremis/bin/_forward.php 90 function: include ( "/var/www/siremis-2.0.0/siremis/bin/_forward.php" ) @ /var/www/siremis-2.0.0/siremis/index.php 3 Please ask system administrator for help...

2 1

Siremis 2.0: error in LCR table editing??
by Klaus Feichtinger 15 Nov '10

15 Nov '10

Hello list, has anybody else tried using SIREMIS 2.0 for administrating LCR tables in kamailio 3.1.0? I am not happy with the form / input boxes that are presented in the web browser, because some fields are not as expected..... Following points are - from my point of view - incorrect: 1) table "LCR_Gateway_List": the fields 'URI-scheme', 'Transport' and 'Strip' are represented by checkboxes. However, in reality these fields are defined with smallint(3) types and e.g. transport is representing values from NULL...4 - not boolean! 2) table "LCR_Target_List": the field priority is represented by a checkbox, too. However, also this field is not boolean. Priorities are represented by numbers, starting with '0' = highest priority a.s.o. Can anybody check if this "problem" is a real one, or only present on my PC? I have tested with IE and Firefox - the result is still the same. regards, Klaus

5 6

Call from web
by Sergey Babichev 12 Nov '10

12 Nov '10

2 2

Re: [SR-Users] Question about the Q-value.
by Kosilov Fedor 12 Nov '10

12 Nov '10

OK, and how do you usually deal with this? I was thinking of putting the Q-value into the "Display Name" field, so when the device registers, I could get it using $fn (reference to display name of 'From' header) and put it directly to the location table in the database...

4 6

help in kamailio and tunnel server
by hala alramli 12 Nov '10

12 Nov '10

i want to instal tunnel server in kamailio . i want to ask if there is any opensource voip tunnel server . and if there is an tutorial to configure kamailio with tunnel server thanks for help hala

2 1

Re: [SR-Users] noack and t_relay erros errors
by JR Richardson 11 Nov '10

11 Nov '10

> Hi All, > > I'm still getting these errors and I'm struggling to resolve the > problem. I think I'm missing an append_branch or something simple in > my config, a little guiedance will be appriciated. > > The error: > > Nov 11 10:23:26 sip-router1 /usr/local/sbin/kamailio[23739]: ERROR: tm > [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding > Nov 11 10:23:26 sip-router1 /usr/local/sbin/kamailio[23739]: ERROR: tm > [tm.c:1300]: ERROR: w_t_relay_to: t_relay_to failed > > The sip graph causing the error: > > http://pastebin.com/VHhQ1sJY > > My config: > > http://pastebin.com/BYn4g5ur > > Do I need to have a mechanism to deal with a 486 form the carrier? > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses > > > > ------------------------------ > > Message: 4 > Date: Thu, 11 Nov 2010 11:52:21 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] noack and t_relay erros errors > To: sr-users(a)lists.sip-router.org > Message-ID: <4CDC1F45.1000704(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > 1. One problem may be that you are calling route[RELAY] from your > failure route and using sl_send_reply(). The route[RELAY] is still > being invoked in a failure route execution context, even if you have > managed to contextually invoke another route, or subroutine if you will. > Stateless replies (sl_reply_error()) cannot be sent from within a > failure route. > > 2. Yes. throw an append_branch() into your failure route. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 1170 Peachtree Street NE > 12th Floor, Suite 1200 > Atlanta, GA 30309 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > Well, that seemed to help greatly, so far no more errors after changing my failure route to: failure_route[FAIL_ONE] { if (t_is_canceled()) { exit; } ds_mark_dst("P"); ds_next_dst(); append_branch(); t_relay(); } Thanks Alex, right on target again. JR -- JR Richardson Engineering for the Masses

2 2

Re: [SR-Users] Variable substitution with Dispatcher
by Lee Archer 11 Nov '10

11 Nov '10

Hi, can anyone answer this? Regards Lee From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Lee Archer Sent: 10 November 2010 10:25 To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Variable substitution with Dispatcher Hi, I am trying to compare 3.0.4 and 3.1 and the following works in 3.1 but not 3.0.4. Should this work in 3.0 or was it only fixed for 3.1? ds_select_dst($tU,"4"); Regards Lee

2 1

noack and t_relay erros errors
by JR Richardson 11 Nov '10

11 Nov '10

Hi All, I'm still getting these errors and I'm struggling to resolve the problem. I think I'm missing an append_branch or something simple in my config, a little guiedance will be appriciated. The error: Nov 11 10:23:26 sip-router1 /usr/local/sbin/kamailio[23739]: ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding Nov 11 10:23:26 sip-router1 /usr/local/sbin/kamailio[23739]: ERROR: tm [tm.c:1300]: ERROR: w_t_relay_to: t_relay_to failed The sip graph causing the error: http://pastebin.com/VHhQ1sJY My config: http://pastebin.com/BYn4g5ur Do I need to have a mechanism to deal with a 486 form the carrier? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Extension state
by Lucas Alvarez 10 Nov '10

10 Nov '10

Hi, is there a way "database driven" to know who is using the phone? Thanks in advance. Lucas

2 1

multidomain: running multiple kamailio on one host
by MÉSZÁROS Mihály 10 Nov '10

10 Nov '10

Hello all! Is there any known problem to run multiple kamailio/sip-router instance in one host. I need this to handle multidomain situation. The multidomain what is built in is not satisfactory, because the UA-s what are from varying vendors. And some has strange or a very limited SIP implementation. Some can only register ip address, some can use RFC3263, but some can use only DNS A record. :( Any answer, suggestion appreciated very well! Thanks, Misi

7 8

Asterisk 1.6 and Kamailio 3.1 realtime integration tutorial
by Daniel-Constantin Mierla 10 Nov '10

10 Nov '10

Hello, I got the time to upgrade my tutorial about Asterisk and Kamailio realtime integration to latest stable release of Kamailio, v3.1.0: You can find the document at: * http://asipto.com/u/c Hope it is useful for some people within this community. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

Re: [SR-Users] "append_rpid_hf()" and "append_hf()" issues
by Dominguez Jover, Ricardo 10 Nov '10

10 Nov '10

Hi Alex, I think I'm doing what you say, look at the configuration my config file: append_rpid_hf("$fU", ";party=calling;id-type=subscriber;privacy=off;screen=yes"); consume_credentials(); (the following lines are commented) ####### I also tested the following # append_hf("Remote-Party-ID:<sip:$avp(rpid)@$avp(domain);user=phone>;privacy=$avp(privacy);party=calling\r\n"); # append_hf("P-Asserted-Identity:<sip:$avp(rpid)@$avp(domain)>\r\n", "Call-ID"); # consume_credentials(); Is that right or you mean something like this: append_rpid_hf("$avp(rpid)", ";party=calling;id-type=subscriber;privacy=off;screen=yes"); consume_credentials(); ? Anyway I've just tested this last configuration and I can't see "Remote Party ID" headers in SIP message. Kind regards, Ricardo -----Mensaje original----- De: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] En nombre de Alex Balashov Enviado el: miércoles, 10 de noviembre de 2010 11:24 Para: sr-users(a)lists.sip-router.org Asunto: Re: [SR-Users] "append_rpid_hf()" and "append_hf()" issues Ricardo, The 'rpid' value is loaded from the 'subscriber' table as part of the authentication functions; if you consume_credentials() before it, it will be gone. You should append_rpid_hf() the RPID header first, and consume_credentials() afterward. Cheers, -- Alex

1 0

Variable substitution with Dispatcher
by Lee Archer 10 Nov '10

10 Nov '10

Hi, I am trying to compare 3.0.4 and 3.1 and the following works in 3.1 but not 3.0.4. Should this work in 3.0 or was it only fixed for 3.1? ds_select_dst($tU,"4"); Regards Lee

1 0

"append_rpid_hf()" and "append_hf()" issues
by Dominguez Jover, Ricardo 10 Nov '10

10 Nov '10

Hi everybody, I've been testing and setting up kamalio 3.0.3 with MySQL since a few weeks ago. All the basic functionalities are working properly, but now I have an issue which I'm not been able to solve by myself. I've read documentation and posts, but no idea about what is happening. This is the issue. I'm trying to append Remote Party ID field to the SIP message header. I've been trying with both "append_rpid_hf()" and "append_hf()" and I can't see any Remote Party Header at the SIP messages sniffed with wireshark. Here is the configuration I've tried: #para cargar los Remote party ID de la tabla suscriber modparam("auth_db", "load_credentials", "$avp(s:rpid)=rpid") modparam("auth", "rpid_avp", "$avp(s:rpid)") modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", ";user=phone>;party=calling;screen=yes;id-type=subscriber;privacy=off") #Modificacion para no pedir autenticacion si viene desde un trusted if (!allow_trusted("$si", "$(proto{s.toupper})")) { if (!proxy_authorize("umh.es", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } append_rpid_hf("$fU", ";party=calling;id-type=subscriber;privacy=off;screen=yes"); consume_credentials(); ####### I also tested the following # append_hf("Remote-Party-ID: <sip:$avp(rpid)@$avp(domain);user=phone>;privacy=$avp(privacy);party=cal ling\r\n"); # append_hf("P-Asserted-Identity: <sip:$avp(rpid)@$avp(domain)>\r\n", "Call-ID"); # consume_credentials(); } } In a similar post I've read that the user had a problem because of the domain table. I have both the domain name and the IP of the host defined, and I also get response with "$avp(s:rpid) , when RPID field is defined to a subscriber. As I said I can't find the mistake, so any help would be appreciated. Thanks in advance and best regards, Ricardo Dominguez

2 1

Re: [SR-Users] LCR database issues
by Lee Archer 10 Nov '10

10 Nov '10

Anyone any idea why these are reporting command disabled? Regards Lee From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Lee Archer Sent: 08 November 2010 17:27 To: sr-users(a)lists.sip-router.org Subject: [SR-Users] LCR database issues Hi, I have kamailio configured to get it's LCR info from a mysql database but when I try to get this info from the command line I am told the command is disabled. Why is this? [root@kam-3-1-s1 kamailio]# kamctl lcr show_gws ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr dump_gws ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr show_routes ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr dump_routes ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr show_gws ERROR: command disabled lcr show_gws....... show database gateways lcr show_routes.... show database routes lcr dump_gws....... show in memory gateways lcr dump_routes.... show in memory routes lcr reload ........ reload lcr gateways and routes lcr eval_weights .. evaluates probability for given GW's weights [root@kam-3-1-s1 kamailio]# kamctl lcr reload ERROR: command disabled Regards Lee

2 4

kamctl duplication of output
by Lee Archer 10 Nov '10

10 Nov '10

Is there a problem with kamctl and duplication of the existing commands? [root@kam-3-1-s1 ~]# kamctl /usr/local/sbin/kamctl 3.1.0 Existing commands: -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart <uri> ................ restart phone configured for <uri> -- command 'online' - dump online users from memory online ............................. display online users -- command 'monitor' - show internal status monitor ............................ show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping <uri> ......................... ping <uri> with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [<username>]................ show in-RAM online users ul show --brief..................... show in-RAM online users in short format ul rm <username> [<contact URI>].... delete user's usrloc entries ul add <username> <uri> ............ introduce a permanent usrloc entry ul add <username> <uri> <expires> .. introduce a temporary usrloc entry -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart <uri> ................ restart phone configured for <uri> -- command 'online' - dump online users from memory online ............................. display online users -- command 'monitor' - show internal status monitor ............................ show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping <uri> ......................... ping <uri> with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [<username>]................ show in-RAM online users ul show --brief..................... show in-RAM online users in short format ul rm <username> [<contact URI>].... delete user's usrloc entries ul add <username> <uri> ............ introduce a permanent usrloc entry ul add <username> <uri> <expires> .. introduce a temporary usrloc entry Regards Lee

2 1

parsing issue 3.1.0
by Jeremya 10 Nov '10

10 Nov '10

These text lines parse if ($sht(ipban=>$si)!=$null) $sht(ipban=>$si)=1; These don't parse if($sht (ipban=>$si)!=$null) $sht (ipban=>$si)=1; In both cases the gap between $sht and the opening '(' is the problem. Is this expected behaviour? Jeremy

2 1

Dispatcher reload in Kamailio 3.1
by Lee Archer 09 Nov '10

09 Nov '10

Hi, after adding another dispatcher destination I tried to reload my dispatcher but got the following: [root@kam-3-1-s1 kamailio]# kamctl dispatcher reload 500 No reload support for call load dispatching Other kamctl dispatcher commands work so am I missing something? Regards Lee

2 3

LCR database issues
by Lee Archer 09 Nov '10

09 Nov '10

Hi, I have kamailio configured to get it's LCR info from a mysql database but when I try to get this info from the command line I am told the command is disabled. Why is this? [root@kam-3-1-s1 kamailio]# kamctl lcr show_gws ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr dump_gws ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr show_routes ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr dump_routes ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr show_gws ERROR: command disabled lcr show_gws....... show database gateways lcr show_routes.... show database routes lcr dump_gws....... show in memory gateways lcr dump_routes.... show in memory routes lcr reload ........ reload lcr gateways and routes lcr eval_weights .. evaluates probability for given GW's weights [root@kam-3-1-s1 kamailio]# kamctl lcr reload ERROR: command disabled Regards Lee

2 1

no audio
by Rizwan Qureshi 08 Nov '10

08 Nov '10

Hi List, I am using dispatcher module to route everything from registerations to calls toa asterisk pbx. thats all what openser is doing at the moment. outgoing calls are fine but I get CHANUNAVAILABLE status for incoming calls on Asterisk. I guess the reverse sip dialogue are not working properly through openser. Following is my open ser config route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } #record_route(); ds_select_dst("1","2"); forward(); } I have tried with and without record_route() but nothing working for incoming calls. Any ideas?

1 0

query dispatcher status from config
by Klaus Darilion 08 Nov '10

08 Nov '10

Hi! Is it somehow possible to query the status (active/inactive/probing) of a dispatcher target from within kamailio.cfg? Thanks Klaus

1 0

Kamailio 3.1 + LCR + CARRIERROUTE howto ?
by Sébastien Cramatte 08 Nov '10

08 Nov '10

Hello, I'm new in kamailio/sr. I've installed 3.1 packages on my debian squeeze. I would like to configure LCR and CARRIERROUTE modules but I don't find suficient doc to do it properly. Does someone can give me resources where I can found : - required modparam for LCR and CARRIERROUTE - SQL examples datas for both modules - example of route logic using both modules Thank you

3 3

checking if calls are still active
by Vic Jolin 08 Nov '10

08 Nov '10

Hi, Just want to ask what is the best way to check if calls are still active especially when we are not handling media? Thanks

4 6

avp_load_radius unknown command
by Vitaly Pougatchev 07 Nov '10

07 Nov '10

Please, help me.. kamailio-3.1.0# ser -c loading modules under /usr/local/lib/ser/modules:/usr/local/lib/ser/modules_k:/usr/local/lib/ser/modules_s 0(12801) : <core> [cfg.y:3412]: parse error in config file /usr/local/etc/ser/ser.cfg, line 242, column 36: unknown command, missing loadmodule? 0(12801) : <core> [cfg.y:3412]: parse error in config file /usr/local/etc/ser/ser.cfg, line 248, column 42: unknown command, missing loadmodule? ---line 242 --- kamailio-3.1.0#cat /usr/local/etc/ser/ser.cfg | head -242 | tail -1 if(avp_load_radius("callee")) ---line 248 --- kamailio-3.1.0#cat /usr/local/etc/ser/ser.cfg | head -248 | tail -1 avp_printf("i:100", "$Tf");

1 0

Error on creating user or edit user
by Joseph Ong 07 Nov '10

07 Nov '10

siremis 2.0 new setup not sure if I made a mistake in setup of siremis. add new user in Application Menu->Users->User Management Add [Sun Nov 07 02:30:30 2010] [error] [client 1.2.3.4] PHP Parse error: syntax error, unexpected ',' in /var/www/siremis/openbiz/bin/Expression.php(330) : eval()'d code on line 1, referer: http://localhost/siremis/system/user_new ----- *function:* eval_syntax ( "$ret = joseph, 3, 10;" ) @ /var/www/siremis/openbiz/bin/Expression.php 327 *function:* evaluateExpression ( "{[fld_username], 3, 10}", Object(UserForm) ) @ /var/www/siremis/openbiz/bin/Expression.php 190 *function:* replaceVarExpr ( "@validate:betweenLength([fld_username], 3, 10)", Object(UserForm) ) @ /var/www/siremis/openbiz/bin/Expression.php 311 *function:* evaluateExpression ( "{@validate:betweenLength([fld_username], 3, 10)}", Object(UserForm) ) @ /var/www/siremis/openbiz/bin/easy/element/Element.php 472 *function:* validate ( ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 1370 *function:* validateForm ( ) @ /var/www/siremis/siremis/modules/system/form/UserForm.php 146 *function:* validateForm ( ) @ /var/www/siremis/siremis/modules/system/form/UserForm.php 51 *function:* CreateUser ( ) @ /var/www/siremis/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /var/www/siremis/openbiz/bin/BizController.php 110 *function:* dispatchRequest ( ) @ /var/www/siremis/openbiz/bin/BizController.php 32 *function:* include_once ( "/var/www/siremis/openbiz/bin/BizController.php" ) @ /var/www/siremis/siremis/bin/controller.php 6

1 0

Re: [SR-Users] Terminated call processing Status: 487
by JR Richardson 05 Nov '10

05 Nov '10

>> I'm seeing random errors in my Kamailio 3.0.2 log: >> >> ERROR: tm [tm.c:1300]: ERROR: w_t_relay_to: t_relay_to failed >> ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding > > Are you sure those are the only errors in the log? > > I've gotten that before when getting bogus or missing 'Via' headers > from bandwidth.com in the 487s. > Here is a pastebin of the conversation, what is interesting is line 29 through 32. http://pastebin.com/QcaAeqjN For some reason Kamailio is sending 2 additional 487's without a response. I believe this is why I am getting the log errors. I played around with "modparam("tm", "cancel_b_method", 2)" but have the same results. Here is a pastebin of my kamailio.cfg: http://pastebin.com/Tn3humr0 Any suggestions on where to put some logging statements and what to look for would be helpful. Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Re: [SR-Users] installing dispatcher module
by Daniel-Constantin Mierla 05 Nov '10

05 Nov '10

do not send private emails. Keep the mailing list cc-ed. Thanks, Daniel On 11/5/10 1:41 PM, Rizwan Qureshi wrote: > Thanks. Problem solved anyway. > > On Thu, Nov 4, 2010 at 10:04 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > run kamailio with -c parameter to get hints about the line where > the error happens. > > Daniel > > > On 11/4/10 4:17 PM, Rizwan Qureshi wrote: >> Hi List, >> This is my first email to this list and I am a kamailio newbie. I >> have worked on asterisk a lot but this is the first time im going >> to use kamailio. >> >> I have successfully installed kamailio on my centos system. I can >> see all the modules located in the >> /usr/local/lib/kamailio/modules_k/ AND >> /usr/local/lib/kamailio/modules/ directories. But the problem is >> when i try to load the dispatcher module from kamailio.cfg it >> gives me the following error. >> >> loading modules under >> /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ >> ERROR: bad config file (1 errors) >> >> Has anyone any clue why this is happening? I cant find any >> details about thsi error otherwise i would have posted it. >> >> Cheers >> >> Best Regards >> Rizwan Qureshi >> >> >> _______________________________________________ >> 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

Terminated call processing Status: 487
by JR Richardson 04 Nov '10

04 Nov '10

Hi All, I'm seeing random errors in my Kamailio 3.0.2 log: ERROR: tm [tm.c:1300]: ERROR: w_t_relay_to: t_relay_to failed ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding After going blind staring at wireshark, I can consistently reproduce this error by initiating a call, then after about a second ro two, quickly hanging up. So here is my setup: SIP Phone><Asterisk><Kamailio><SIP Service Provider I see all transaction messages flowing properly through all nodes and I don't get any hung channels, all call legs release properly. I'm guessing I need a special [route] in Kamailio to handle 487 request so they look normal and not issue an error? Kamailio is forwarding the 487 messages ok, so why do I get a t_relay error? What info would be helpful to diagnose the errors? Any insight would be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

3 2

installing dispatcher module
by Rizwan Qureshi 04 Nov '10

04 Nov '10

Hi List, This is my first email to this list and I am a kamailio newbie. I have worked on asterisk a lot but this is the first time im going to use kamailio. I have successfully installed kamailio on my centos system. I can see all the modules located in the /usr/local/lib/kamailio/modules_k/ AND /usr/local/lib/kamailio/modules/ directories. But the problem is when i try to load the dispatcher module from kamailio.cfg it gives me the following error. loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ ERROR: bad config file (1 errors) Has anyone any clue why this is happening? I cant find any details about thsi error otherwise i would have posted it. Cheers Best Regards Rizwan Qureshi

2 1

Try to build the SER 2.0 RC1 on Solaris 10 but failed.
by Ivan Yiu 04 Nov '10

04 Nov '10

I try to build the SER using gmake, after resolving some dependency issue, the script can run but finally got a error as follows. yacc -d -b cfg cfg.y conflicts: 1 shift/reduce gmake: *** [cfg.tab.c] Broken Pipe What is the problem?

2 1

How can I remove all RPID headers from SIP message
by Mino Haluz 04 Nov '10

04 Nov '10

Hi, I tried to add RPID headers at the beggining of SIP options on the client side, and I was able to change the caller number simply this way. How can I avoid this ? I'm searching for some function in Kamailio which would delete all RPID headers so that Kamailio could add only one (and correct) RPID header (I'm using append_rpid_hf) Thanks.

2 1

udp_send "Invalid argument"
by Juha Heinanen 04 Nov '10

04 Nov '10

in some of my tests i get the following udp send related errors, when request is t_relayed: Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: INFO: Routing initial INVITE <sip:+35892223333@test.fi> to <sip:+35892223333@193.166.3.2;transport=udp> Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: ERROR: <core> [udp_server.c:566]: ERROR: udp_send: sendto(sock,0xb2d1f1e8,1190,0,193.166.3.2:5060,16): Invalid argument(22) Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: : <core> [udp_server.c:571]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: ERROR: tm [../../forward.h:148]: msg_send: ERROR: udp_send failed Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: ERROR: tm [t_fwd.c:1244]: ERROR: t_send_branch: sending request on branch 0 failed $du is set to sip:+35892223333@193.166.3.2;transport=udp and the proxy is listening on many sockets, both 127.0.0.1 and external ip address. any idea what might cause the errors? -- juha

5 8

LCR Question
by Trevor Francis 03 Nov '10

03 Nov '10

I have managed to work my way through the LCR module documentation and added the loadmodule and module parameters. However, I dont know where to add the routing logic that will allow me to add LCR rules and gateways in Siremis and to have Kamailio recognize them. Below is my Kamailio config: Advice? root@u15396663:/var/log# cat /usr/local/etc/kamailio//kamailio.cfg #!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v3.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!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_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://root:f1n4lh0p@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 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 reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* 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:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif ####### 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 # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!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 "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "mi_datagram.so" loadmodule "rtimer.so" loadmodule "sqlops.so" loadmodule "cfgutils.so" # ----- LCR ----- loadmodule "lcr.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar 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) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* 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", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@finalhop.net<sip%3Apinger(a)finalhop.net> ") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif # ----- Mi_Datagram ----- modparam("mi_datagram", "socket_name", "udp:127.0.0.1:8033") # ----- Charts ----- modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=STATS") modparam("sqlops","sqlcon", "ca=>mysql://root:f1n4lh0p@localhost/openser") # ----- LCR ----- modparam("lcr","db_url", "mysql://root:f1n4lh0p@localhost/openser") modparam("lcr","lcr_gw_table","lcr_gw") modparam("lcr","id_column", "id") modparam("lcr","lcr_id_column", "lcr_id") modparam("lcr","tag_column","tag") modparam("lcr","weight_column","gw_weight") modparam("lcr","flags_column","flags") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","params_column", "params") modparam("lcr","hostname_column","hostname") modparam("lcr","uri_scheme_column","uri_scheme") modparam("lcr","strip_column","strip") modparam("lcr","transport_column","transport") modparam("lcr","tag_column", "tag") modparam("lcr","defunct_column", "defunct") modparam("lcr","lcr_rule_table","lcr_rule") modparam("lcr","prefix_column","prefix") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","stopper_column", "stopper") modparam("lcr","enabled_column", "enabled") modparam("lcr","lcr_rule_target_table", "lcr_rule_target") modparam("lcr", "rule_id_column", "rule_id") modparam("lcr", "gw_id_column", "gw_id") modparam("lcr", "priority_column", "priority") modparam("lcr","weight_column", "weight") modparam("lcr", "lcr_count", 1) modparam("lcr","gw_uri_avp", "$avp(i:709)") modparam("lcr","ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "defunct_capability", 0) modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") modparam("lcr", "defunct_gw_avp", "$avp(s:defunct_gw_avp)") modparam("lcr", "lcr_rule_hash_size", 128) modparam("lcr", "lcr_gw_count", 128) modparam("lcr", "fetch_rows", 1024) ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Siremis Charts route[STATS] { sql_query("ca", "insert into statistics (time_stamp,shm_used_size,shm_real_used_size," "shm_max_used_size,shm_free_used_size,ul_users,ul_contacts) values ($Ts," "$stat(used_size),$stat(real_used_size),$stat(max_used_size)," "$stat(free_size),$stat(location-users),$stat(location-contacts))", "ra"); } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } root@u15396663:/var/log#

1 0

Connection tests
by Olle E. Johansson 03 Nov '10

03 Nov '10

Hello friends, Is there any way from the outside (like from MI) I can check the existing TCP/TLS connections? Can I drop them somehow? /O

3 2

buffer overflow
by Maw Zin Tun - Real Telekom 03 Nov '10

03 Nov '10

Hi List, Any idea how to correct this error ? Nov 3 15:01:18 s15391037 /usr/local/kamailio-3.0/sbin/kamailio[27812]: ERROR: <core> [pvapi.c:967]: no more space for text [84] Nov 3 15:01:18 s15391037 /usr/local/kamailio-3.0/sbin/kamailio[27812]: ERROR: <core> [pvapi.c:995]: buffer overflow -- increase the buffer size... Nov 3 15:01:18 s15391037 /usr/local/kamailio-3.0/sbin/kamailio[27812]: ERROR: sqlops [sql_api.c:204]: cannot print the sql query Thanks, Maw Tun Real Telekom

2 1

DoS on Kamailio
by Bernhard Suttner 03 Nov '10

03 Nov '10

Hi, is there a Kamailio module to prevent DoS Attacks like SIP Register Scanners? I know, that there is a way to implement that with Fail2Ban and xlog but it would be nice to have a "pure" Kamailio solution for that? Any idea is appreciated. Best regards, Bernhard Suttner

3 2

Siremis V2.0.0 Released
by Elena-Ramona Modroiu 03 Nov '10

03 Nov '10

Hi, Siremis v2.0.0 is out – the web management interface for Kamailio (Openser) v3.1.0 and SIP Express Router (SER). This is a major release, with countless improvements and new features since v1.x series, among them: * major re-factoring of web interface * better accessibility * simplified menu structure * completely new look * dashboard with the map of all available tools * developed on top of Cubi and PHPOpenBiz v2.4 frameworks * web installation wizard * added new modules: xcap, dialog, new lcr * usage of separate database for siremis itself * management of users that can login to siremis * management of menu can be done from web interface * building Apache conf and htaccess file can be done by Makefie * charts to monitor location transport layers Step by step installation tutorial, screenshots and demo are available on the web at: * Installation Tutorial - http://asipto.com/u/6 * Screenshots - http://asipto.com/u/7 * Web Demo - http://asipto.com/u/8 Siremis is used during Kamailio Advanced Training classes for management of SIP server, a good opportunity to learn about Siremis itself, next locations are: * Kamailio Advanced Training for Carrier Services, November 22-25, 2010, Berlin, Germany (http://asipto.com/u/kat) * Kamailio Advanced Training for Carrier Services, January 24-26, 2010, Irvine, CA, USA (http://asipto.com/u/katu) More details about Siremis project are available at: * http://siremis.asipto.com Regards, Ramona

3 6

Latest Siremis build problem
by Trevor Francis 03 Nov '10

03 Nov '10

Not Found The requested URL /siremis/system/general_default was not found on this server. Apache/2.2.8 (Ubuntu) PHP/5.2.4-2ubuntu5.12 with Suhosin-Patch Server at 74.208.174.183 Port 80 I have tested this on both Firefox and Safari.

2 1

using tls from modules_s instead of core
by Jijo 02 Nov '10

02 Nov '10

Hi all. What changes i have to make in the build to use tls from modules_s instead from core modules. Thanks Jijo

2 1

loop through variables
by Bernhard Suttner 02 Nov '10

02 Nov '10

Hi, I want something like that: # define some router to use later in the script (global section) Mod_param(router(1), "10.10.10.1") mod_param(router(2), "10.10.10.2") mod_param(router(3), "10.10.10.3") Mod_param(router_count, "3") within the different routes I want to loop through the routers like that: $var(i) while($(var(i) < $var(router_count)) { if ($var(router($var(i)) == src_ip") { .. do something } } Is that somehow possible? Any hint or idea is appreciated! Best regards, Bernhard Suttner

5 11

Question about the Q-value.
by Kosilov Fedor 02 Nov '10

02 Nov '10

Hello, everybody! As I understand, the Q-value must be set by a client-side device. We use mainly Linksys SPA2102, I have searched through the web interface of this device but have not found where this value should be set. Is this parameter not supported by simple SOHO devices? Or have I misunderstood something?

3 2

siremis 2.0 and Captcha issues !
by X- x -X 02 Nov '10

02 Nov '10

Hi all, Is anyone else is also having problems in seeing captcha while logging in http://myhost.mydomain/siremis/index.php/user/login<http://192.168.90.125/siremis/index.php/user/login> (during installation, I haven't been able to login since Im not able to see any image. just the image reference thing) I've tried firefox, chome & opera (all with flash plugins installed & of-course simple images also show fine) Is there a way I can disable them. Seem like these 2 files are driving the whole captch show /var/www/siremis-2.0.0/siremis/bin/antispam_image.php /var/www/siremis-2.0.0/siremis/modules/user/form/Antispam.php Are there any distro version specific libraries required. Im trying the whole thing on debian lenny. Cheers -- X

2 2

SMS module not found parameter.
by peter_green lion 02 Nov '10

02 Nov '10

hi all, i am a new bie in kamailio. i use SMS module to send message to GSM net work. my kamailio.cfg is : modparam("sms", "modems", "wavecom[d=/dev/ttyS;c=+84900000022;b=57600; m=new;l=20]") modparam("sms", "networks", "D1[m=10]; d2[m=10]") modparam("sms", "links", "wavecom[D1;d2]") modparam("sms", "default_net", "localhost") modparam("sms", "domain_str", "D1.d2") modparam("sms", "max_sms_parts", 10) modparam("sms", "use_contact", 1) modparam("sms", "sms_report_type", 1) if(method == "MESSAGE"){ if(has_body("text/plain")){ sms_send_msg_to_net("D1"); } } but when i start kamailio the log show : Oct 25 06:28:26 localhost kamailio: ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <domain_str> not found in module <sms> Oct 25 06:28:26 localhost kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 173, column 42: Can't set module parameter please help to configure it correct. i use wavecom modem and some infomation: baudrate=57600 c=+84900000022 ip server:192.168.1.100 modem connect with server via com1 (/dev/ttyS0) and kamailio version 3.0.2 thanks for help me. Peter Green

2 3

Dead Charts in Siremis 2.0
by Trevor Francis 02 Nov '10

02 Nov '10

I managed to get the SHM chart working by manually adding the field "random" inside the table "statistics." However, the Charts "Load Charts" and "TM Charts" gave me "Internal System Errors" unless I add the fields tm_active, rcv_req_diff, fwd_req_diff, and 2xx_trans_diff to the statistics table in mysql. I am not sure if these statistics are being updated, because my charts dont appear to be displaying information. Can anyone speak to what is going on here? TGF

2 1

Kamalio 3.1, Siremis 2.0 Problem
by Trevor Francis 02 Nov '10

02 Nov '10

I am running into the following errors: /usr/local/sbin/kamailio[6774]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Unknown column 'random' in 'field list' /usr/local/sbin/kamailio[6774]: ERROR: <core> [db_query.c:130]: error while submitting query /usr/local/sbin/kamailio[6774]: ERROR: sqlops [sql_api.c:217]: cannot do the query I believe it is preventing my LCR from working properly.

2 1

New Gateway Configuration
by Trevor Francis 01 Nov '10

01 Nov '10

My final question/comment for the day, I promise. I am used to using Siremis 1.x or configuring gateways. We are testing a new box on Kam 3.1 and Siremis 2.0. Previously to add an external LCR gateway, I would plop in the gateway name, group id, and tag to the LCR Gateway section and create an LCR rule with the LCR ID, Group ID, Prefix, and Priority that points to the LCR gateway that I created and I could fly straight. Since there are new attributes in 3.1 and in Siremis, how does the new configuration look? I have successfully called from softphone subscriber to softphone subscriber with no issues, but cannot get the LCR to allow me to dial beyond my internal network. Ideas? TGF

1 0

Kamailio 3.1, Siremis 2.0
by Trevor Francis 01 Nov '10

01 Nov '10

I am running into the following errors: /usr/local/sbin/kamailio[6774]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Unknown column 'random' in 'field list' /usr/local/sbin/kamailio[6774]: ERROR: <core> [db_query.c:130]: error while submitting query /usr/local/sbin/kamailio[6774]: ERROR: sqlops [sql_api.c:217]: cannot do the query I believe it is preventing my LCR from working properly. -- Trevor G. Francis Managing Member trevor.francis(a)tgrahamcapital.com Ph. +1 888.616.0662 Fx. +1 214.722.1318 P.O Box 54771 Oklahoma City, OK 73154 MSN: trevor.francis(a)fiberhaus.com Personal emails should be addressed to: tfrancis(a)fas.harvard.edu --

1 0

Setting up TLS between proxy and authentication server
by Couprie Geoffroy 01 Nov '10

01 Nov '10

Hello, I am testing TLS communication with Kamailio 3.0.2, and I encounter a strange problem. My setup is like this: Client <-UDP-> Proxy server <- TLS with client certificate authentication -> Authentication server 192.168.24.1 192.168.24.128 192.168.24.129 The two servers are instance of Kamailio 3.0.2 When the client sends a REGISTER, the proxy retransmits the message to the authentication server, which sends back a 401 Unauthorized. But it seems the proxy closes the TLS connexion right after forwarding the REGISTER, and doesn't receive the 401 message. The TLS handshake is OK, and the client certificate is required (I didn't add the verification part yet). The REGISTER message goes through TLS, and is received by the authentication server. Then, the proxy sends a TLS alert (Close-notify), and after that message, the authentication server sends back the 401, and the proxy ignores that message. Could it be caused by a timeout? Is there a way to keep the TLS connection opened? Here are the relevant files (I don't like to send mails of more than 100 lines): authentication server configuration: http://pastebin.com/QBmnNc4e authentication server log: http://pastebin.com/uYdHDG5G proxy server configuration: http://pastebin.com/8WPPJBtM proxy server log: http://pastebin.com/JTwJSKtk I am just testing TLS, so I have tried to remove most of the irrelevant parts. Best regards, Geoffroy ________________________________ Ce message et les pi?ces jointes sont confidentiels et r?serv?s ? l'usage exclusif de ses destinataires. Il peut ?galement ?tre prot?g? par le secret professionnel. Si vous recevez ce message par erreur, merci d'en avertir imm?diatement l'exp?diteur et de le d?truire. L'int?grit? du message ne pouvant ?tre assur?e sur Internet, la responsabilit? du groupe Atos Origin ne pourra ?tre recherch?e quant au contenu de ce message. Bien que les meilleurs efforts soient faits pour maintenir cette transmission exempte de tout virus, l'exp?diteur ne donne aucune garantie ? cet ?gard et sa responsabilit? ne saurait ?tre recherch?e pour tout dommage r?sultant d'un virus transmis. This e-mail and the documents attached are confidential and intended solely for the addressee; it may also be privileged. If you receive this e-mail in error, please notify the sender immediately and destroy it. As its integrity cannot be secured on the Internet, the Atos Origin group liability cannot be triggered for the message content. Although the sender endeavours to maintain a computer virus-free network, the sender does not warrant that this transmission is virus-free and will not be liable for any damages resulting from any virus transmitted.

3 5

drop in branch route
by Juha Heinanen 31 Oct '10

31 Oct '10

based on core cookbokk, it is not clear to me what happens when drop is executed in branch route: drop Stop the execution of the configuration script and alter the implicit action which is done afterwards. If the function is called in a 'branch_route' then the branch is discarded (implicit action for 'branch_route' is to forward the request). i understand that request is not forwarded to that branch, but if that branch is the only branch, does t_relay() then return -1 or is script execution terminated? -- juha

1 0

setdebug() and 3.X
by Sergey Okhapkin 30 Oct '10

30 Oct '10

setdebug() function is no longer provided by core. How to change debug level dynamically in the script to trace execution of a few statements only?

3 11

How to log local SIP replies?
by Sergey Okhapkin 30 Oct '10

30 Oct '10

Is there a way to log local replies sent by kamailio? For example, I want to log a reply sent by kamailio in response to REGISTER request.

2 1

Check out Elias's photos on Facebook
by Elias Gitonga 30 Oct '10

30 Oct '10

Hi Serusers, I set up a Facebook profile where I can post my pictures, videos and events and I want to add you as a friend so you can see it. First, you need to join Facebook! Once you join, you can also create your own profile. Thanks, Elias To sign up for Facebook, follow the link below: http://www.facebook.com/p.php?i=100001050585836&k=Z6E3Y6W5U2VBWABJPB63QUTTV… Already have an account? Add this email address to your account: http://www.facebook.com/n/?merge_accounts.php&e=serusers%40iptel.org&c=2814… ======================================= serusers(a)iptel.org was invited to join Facebook by Elias Gitonga. If you do not wish to receive this type of email from Facebook in the future, please follow the link below to unsubscribe. http://www.facebook.com/o.php?k=679489&u=100001575534555&mid=335e9d2G5af36e… Facebook, Inc. P.O. Box 10005, Palo Alto, CA 94303

1 0

Re: [SR-Users] setdebug() and 3.X
by Daniel-Constantin Mierla 30 Oct '10

30 Oct '10

On 10/29/10 5:40 PM, Sergey Okhapkin wrote: > I can't test now - high volume traffic is running. But I looked at the diff, it > seems to me OK, I see no reason why it could not work:-) Because of the law, the Murphy law ;-) Probably I will test sometime next week if you don't do it before and then backport. Cheers, Daniel > On Friday 29 October 2010, you wrote: >> On 10/29/10 4:33 PM, Daniel-Constantin Mierla wrote: >>> On 10/29/10 4:09 PM, Sergey Okhapkin wrote: >>>> I just realized that 3.1 executes main routing block when broken SIP >>>> message >>>> is received! But sanity_check() function returns 0 on failed checks >>>> and script >>>> execution stops, my favorite xlog() is not executed. >>> OK, that can be fixed very easy. Iirc, in some cases you don't get the >>> sip message in the config if it is really broken, but I am not sure if >>> error_route was executed for that case anyhow. >> I committed on git master, id: >> >> 39a8108c62a2beafba18417613b9660a8abef86a >> >> >> Can you take the patch and test it on 3.1? If all is fine, then I can >> backport - even the default cfg expects non-drop behavior since it >> attempts to print a log message there. >> >> Cheers, >> Daniel >> >>> Cheers, >>> Daniel >>> >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [msg_translator.c:1946]: ERROR: build_res_buf_from_sip_req: alas, parse >>>> _headers failed >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: WARNING: sanity >>>> [sanity.c:253]: sanity_check(): check_required_headers(): failed to s >>>> end 400 via sl reply >>>> >>>> On Friday 29 October 2010, Daniel-Constantin Mierla wrote: >>>>> On 10/29/10 11:53 AM, Sergey Okhapkin wrote: >>>>>> Daniel, I think this could be implemented without re-introduction of >>>>>> error_route, >>>>> it will not be re-introduced for sure, we have now a different core >>>>> framework which care reuse event_route mechanism. >>>>> >>>>> In the past the error route was executed when the error was discovered, >>>>> so it several places of the code, and I am not sure that the core can >>>>> continue up to config execution of something is really broken. >>>>> >>>>> I will have it in mind when I get some spare time. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>>> what about core parameter route_on_error=yes/no (default no)? If >>>>>> route on error is enabled, then sanity module will do the trick: >>>>>> >>>>>> route{ >>>>>> if(!sanity_check("1159")) { >>>>>> xlog("L_INFO","Bad message from >>>>>> $proto:$si:$sp\n$mb\n"); >>>>>> break; >>>>>> } >>>>>> .... >>>>>> >>>>>> It will be up to script writer what to do in main routing block if >>>>>> message parsing fails. >>>>>> >>>>>> On Friday 29 October 2010, Sergey Okhapkin wrote: >>>>>>> It would be nice to get error_route back. The route was trivial in my >>>>>>> cfg file: >>>>>>> >>>>>>> error_route { >>>>>>> xlog("L_INFO","Bad message from $proto:$si:$sp\n$mb\n"); >>>>>>> drop(); >>>>>>> } >>>>>>> >>>>>>> On Friday 29 October 2010, Daniel-Constantin Mierla wrote: >>>>>>>> On 10/28/10 1:41 PM, Sergey Okhapkin wrote: >>>>>>>>> I've found one more feature missed in 3.1 - error_route is >>>>>>>>> eliminated >>>>>>>>> and sanity module added. But if received SIP message is >>>>>>>>> malformed and >>>>>>>>> can't be parsed, routing script is not executed and I have no >>>>>>>>> way to >>>>>>>>> log the message to kamailio log. I did it in error_route before. >>>>>>>>> >>>>>>>>> This logging is important to me to support customers. SIP ALG in >>>>>>>>> routers often breaks Via line :-( >>>>>>>> Since I developed error_route, I remember that it handled just >>>>>>>> sporadic >>>>>>>> cases, but I can add an event_route to pass the buffer and some >>>>>>>> hints >>>>>>>> to configuration file in case the initial parsing fails and the >>>>>>>> message >>>>>>>> is dropped before getting to config file. >>>>>>>> >>>>>>>> It is good to know that someone use such case, introduction of >>>>>>>> sanity >>>>>>>> was clearly mentioned and so far there was no complain. >>>>>>>> >>>>>>>> Cheers, >>>>>>>> Daniel >>>>>>>> >>>>>>>>> On Thursday 28 October 2010, Daniel-Constantin Mierla wrote: >>>>>>>>>> On 10/26/10 2:22 PM, Sergey Okhapkin wrote: >>>>>>>>>>> setdebug() function is no longer provided by core. How to change >>>>>>>>>>> debug level dynamically in the script to trace execution of a few >>>>>>>>>>> statements only? >>>>>>>>>> seems was lost in transition to 3.0, I will check to see what >>>>>>>>>> can be >>>>>>>>>> done with the new architecture. >>>>>>>>>> >>>>>>>>>> Thanks, >>>>>>>>>> Daniel >>>>>>>>> _______________________________________________ >>>>>>>>> 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 >>>>>>> _______________________________________________ >>>>>>> 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 >>>>>> _______________________________________________ >>>>>> 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 >>>> _______________________________________________ >>>> 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

2 1

NAT ping replies
by Sergey Okhapkin 29 Oct '10

29 Oct '10

Kamailio 3.1 log is filled with lines like Oct 29 03:32:46 west /usr/local/sbin/kamailio[632]: INFO: <script>: incoming reply from udp:188.62.4.186:65333 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 204.74.213.5:5060;branch=0 From: <sip:pinger@sip.callwithus.com>;tag=3993f0c To: <sip:188.62.4.186:65333> Call-ID: aff32da5-89ef0136-71dc8c5(a)204.74.213.5 CSeq: 1 OPTIONS Content-Length: 0 Oct 29 03:32:46 west /usr/local/sbin/kamailio[632]: INFO: <core> [forward.c:774]: broken reply to forward - no 2nd via Shouldn't replies to NAT ping messages be silently dropped like in 1.4?

2 2

Billing party and radius accounting
by Efelin Novak 29 Oct '10

29 Oct '10

Hi folks, I have a problem with setting billing party in my radius messages. My accounting is done by CDRTool via radius messages to radius server. The thing I don`t know is how to set billing party variable (by default $avp(s:billing_party) with reliable value. I can try adding Diversion header and fill the variable with it, but can I trust the Diversion header? When UAC sets redirect(302) and Invite comes, he replies 302 and Kamailio passes this message to origin. However is it possible to change Diversion header by UAC? This could be way how to make free calls and bill it to someone else. I was thinking of uac_redirect module. If I get it right this module saves original number in memory and does not use message itself to store it. In documentation there is written that accounting can by done by setting modparam("uac_redirect", "acc_function", "acc_rad_request"). Even when I use exported function set_accept_filter() the accounting itself isn't done. I`d like to get original number from module uac_redirect and set it to $avp(s:billing_party). This would solve the problem of billing forwarded calls. Am I getting it right? Is there anything else what can be passed to billing party variable to make correct bililng? I'm using kamailio 3.1.x and CDRTool v 8.0.6 Thanks, Efelin

4 12

tm.t_uac_start examples from CLI
by Alex Balashov 29 Oct '10

29 Oct '10

Can someone post an example of usage of tm.t_uac_start from the command line using sercmd over a UNIX domain socket (e.g. /tmp/ser_ctl)? I am quite unclear on how to pass minimally necessary header values to it in this form and cannot seem to get the syntax right. 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/

1 0

problem getting siriemis to login
by Gerry Kernan 29 Oct '10

29 Oct '10

Hi I am trying to get siremis working with kamailio, butwhen i open the login page and enter the username and password nothing happens. I have followed the instruction on the siremis home page. Created mysql db called openser and inserted username admin& password admin. mysql> use openser; Reading table information for completion of table and columnnames You can turn off this feature to get a quicker startup with-A Database changed mysql> select * from ob_users; +-------+--------+----------+ | SYSID | USERID | PASSWORD | +-------+--------+----------+ | 1 | admin |admin | +-------+--------+----------+ 1 row in set (0.00 sec) mysql> edited /var/www/html/siremis-1.0.1/siremis/metadata/Config.xml <?xml version="1.0"standalone="no"?> <Config> <DataSource> <DatabaseName="Default" Driver="Pdo_Mysql"Server="localhost" DBName="openser" User="admim"Password="admin" /> <DatabaseName="Seremis" Driver="Pdo_Mysql"Server="localhost" DBName="openser" User="admin"Password="admin" /> </DataSource> </Config> Ifsomeone could point out to me where i have gone wrong i would really appreciateit. Regards, Gerry Kernan InfinityIT Suite 17 The Mall, Beacon court, Sandyford, Dublin 18. p:+353-1-2930090 f:+353-1-2930137

1 0

Kamailio Business Directory
by Daniel-Constantin Mierla 28 Oct '10

28 Oct '10

Hello, I am please to announce the launch of Kamailio Business Directory. This is a web page hosted by the project site that will list companies and individuals offering products, services or solutions based on Kamailio or SER. It is an open directory that tries to enable more visibility to the business market around the project and help finding quickly the best service for specific market demands. We invite everyone doing such business to apply to be listed, please follow the instructions at: http://www.kamailio.org/w/business-directory-rules/ We have already several companies listed, couple of them to still in review process, therefore check it again soon, the directory is available at: http://www.kamailio.org/w/business-directory/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

About "reply_code" in 'carrierroute' module
by Iñaki Baz Castillo 28 Oct '10

28 Oct '10

Hi, how is the syntax of the "reply_code" column in 'carrierroute' module? In the doc I some examples: 408 404 50. Obviously "50." means any code starting by "50", but is it a typical regular expression? For example, does it allow something like "500|503" or "50[03]"? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 4

Asterisk, Kamailio, Openfire and Social Media Integration
by Daniel-Constantin Mierla 28 Oct '10

28 Oct '10

Hello, courtesy of Kelvin Chua, I received the slides he presented yesterday at Astricon 2010 to the session "Asterisk, Kamailio, Openfire and Social Media Integration". More details at: http://www.kamailio.org/w/2010/10/asterisk-kamailio-openfire-and-social-med… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

PSTN call
by michel freiha 28 Oct '10

28 Oct '10

Dear All, Can someone help me to connect my kamilio in order to make a PSTN call by rewriting host tp PSTN gateway? I changed my config in a manner to do that...The line is ringing but as soon as I open the line on other side the call will hangup...Please find the piece of code # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rewritehost("XX.XX.XX.XX"); force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } Regards

2 1

SIP application type
by Mino Haluz 28 Oct '10

28 Oct '10

Hi, I'm using radius server to do an accounting, but it does not store an application type (audo/video). Audio/video calls has to be rated with cdrtool, that is why I need this information. It is not even sent through radius protocol, so the radius server cannot get it. My extra-params are: 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)") Suggestions? Thanks.

4 4

HELP
by michel freiha 28 Oct '10

28 Oct '10

Hello All, I have my kamailio server running and working properly as much as it's listening to an IP address..>As soon as i add an alias as a domain mae, I'm not be able to make call anymore and i'm getting Proxy Authentication required on my X-lite registered on the server...Please find below SIP packets: U +0.045706 Kamailio_IP:5060 -> X-Lite_Public_IP:49088 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport=49088;received=X-Lite_Public_IP. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 INVITE. Proxy-Authenticate: Digest realm="Aliasdomain.com", nonce="TMgYGEzIFuyzAuVUGSbPwq3ZJaBE8k4a". Server: kamailio (3.1.0 (i386/linux)). Content-Length: 0. . U +0.253584 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 INVITE sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:michofr@X-Lite_Public_IP:49088>. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com>. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="michofr",realm="asterisk",nonce="2fe0dd55",uri="sip:00XXXXXX@Aliasdomain.com ",response="a2f6f58360e4a3059a82964f9e8ddbe2",algorithm=MD5. Proxy-Authorization: Digest username="michofr",realm="Aliasdomain.com",nonce="TMgYF0zIFutYZomwP0F6LJLcvg4y/gwt",uri="sip:00XXXXXX@Aliasdomain.com ",response="143021428e9b74db816f76a97b522a60",algorithm=MD5. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 283. . v=0. o=- 6 2 IN IP4 192.168.0.164. s=CounterPath eyeBeam 1.5. c=IN IP4 192.168.0.164. t=0 0. m=audio 17956 RTP/AVP 18 101. a=alt:1 1 : 0V1r/5Dg TGVW9xcA 192.168.0.164 17956. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. U +0.057365 Kamailio_IP:5060 -> X-Lite_Public_IP:49088 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport=49088;received=X-Lite_Public_IP. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 INVITE. Proxy-Authenticate: Digest realm="Aliasdomain.com", nonce="TMgYGEzIFuyzAuVUGSbPwq3ZJaBE8k4a". Server: kamailio (3.1.0 (i386/linux)). Content-Length: 0. . U +0.255609 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 ACK sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-4312233712260f32-1---d8754z-;rport. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c9fd. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 3 ACK. Content-Length: 0. . U +0.015090 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 ACK sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 ACK. Content-Length: 0. . U +0.136372 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 ACK sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 ACK. Content-Length: 0.

2 1

Re: [SR-Users] libosip2 parse body error
by Daniel-Constantin Mierla 27 Oct '10

27 Oct '10

Never ever send private emails - they will be discarded, keep the mailing list cc-ed. Iptel org is running SIP Express Router 3.1. Take a ngrep of the traffic and send it here, the hexa codes you send are not for humans to read, like we are here on mailing lists. Otherwise make a tar.gz of pcap file. SER changes content length only if it is an update to sdp (for nat traversal), otherwise is the one set by the other phone. Daniel On 10/27/10 12:21 PM, t wrote: > Thanks Daniel, > > fyi, the Content-Length doesn't match the actual length in the invite > ok returned from iptel.org <http://iptel.org> (213.192.59.75), and > the same situation also occurs from ekiga.net <http://ekiga.net>. > > libosip2 interprets the rfc as saying that this 'should' be correct. > (and I buy this!). > > it looks like an openser (opensips?) quirk. > > anyway, I have put in code locally to fixup the Content-Length. > > Thanks, > Tony. > > 73 75 62 0d 0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e sub..Content-Len > 67 74 68 3a 20 20 20 32 30 31 0d 0a 0d 0a 76 3d gth: 201....v= > 30 0d 0a 6f 3d 74 77 69 6e 6b 6c 65 20 31 33 33 0..o=twinkle 133 > 33 37 30 37 32 38 39 20 38 30 39 37 35 37 36 37 3707289 80975767 > 33 20 49 4e 20 49 50 34 20 31 39 32 2e 31 36 38 3 IN IP4 192.168 > 2e 32 35 34 2e 31 0d 0a 73 3d 2d 0d 0a 63 3d 49 .254.1..s=-..c=I > 4e 20 49 50 34 20 31 39 32 2e 31 36 38 2e 32 35 N IP4 192.168.25 > 34 2e 31 0d 0a 74 3d 30 20 30 0d 0a 6d 3d 61 75 4.1..t=0 0..m=au > 64 69 6f 20 35 30 32 32 20 52 54 50 2f 41 56 50 dio 5022 RTP/AVP > 20 39 38 20 31 30 31 0d 0a 61 3d 72 74 70 6d 61 98 101..a=rtpma > 70 3a 39 38 20 73 70 65 65 78 2f 31 36 30 30 30 p:98 speex/16000 > 0d 0a 61 3d 72 74 70 6d 61 70 3a 31 30 31 20 74 ..a=rtpmap:101 t > 65 6c 65 70 68 6f 6e 65 2d 65 76 65 6e 74 2f 38 elephone-event/8 > 30 30 30 0d 0a 61 3d 66 6d 74 70 3a 31 30 31 20 000..a=fmtp:101 > 30 2d 31 35 0d 0a 0-15.. > > On Wed, Oct 27, 2010 at 5:40 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > you claim that contact header is malformed or libosip2 has issues > in parsing a good content header? > > In case of first option, can you tell what it believes is wrong. > Your message is hard to follow since is not clear text, next time > send only the text part. > > For the second option, you have to approach libosip2 support forums. > > Cheers, > Daniel > > > On 10/25/10 10:44 PM, t wrote: >> Hi, >> >> Not sure how to provide this info (or even if anybody cares), but >> libosip2 parser objects to incorrect (short) Content-Length from >> iptel.org <http://iptel.org>. >> >> Regards, >> Tony. >> >> >> _______________________________________________ >> 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

5 7

libosip2 parse body error
by t 27 Oct '10

27 Oct '10

Hi, Not sure how to provide this info (or even if anybody cares), but libosip2 parser objects to incorrect (short) Content-Length from iptel.org. Regards, Tony.

2 1

SMS module big error ????????????
by peter_green lion 27 Oct '10

27 Oct '10

hi all, i have configure SMS module and have some error. please help to fix it. my configure in kamailio : modparam("sms", "modems", "nokia[d=/dev/ttyUSB0;p=0000;m=old;s=1;l=10;r=20]") modparam("sms", "networks", "MOBIFONE[m=10]") modparam("sms", "links", "nokia[MOBIFONE]") if(method == "MESSAGE"){ if(has_body("text/plain")){ sms_send_msg_to_net("MOBIFONE"); } } but i start kamailio, log show that: Oct 26 05:32:35 localhost /usr/local/sbin/kamailio[13594]: ERROR: sms [libsms_modem.c:282]: Modem is not registered to the network Oct 26 05:32:35 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_getsms.c:256]: something happend with the modem -> was reinit -> let's retry Oct 26 05:32:36 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:356]: modem wants the PIN again! Oct 26 05:32:36 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:370]: re -init the modem!! Oct 26 05:32:36 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:221]: init modem nokia on /dev/ttyUSB0. Oct 26 05:32:37 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:250]: Checking if Modem is registered to the network Oct 26 05:32:38 localhost /usr/local/sbin/kamailio[13594]: NOTICE: sms [libsms_modem.c:275]: Waiting 2 sec. before retrying Oct 26 05:33:09 localhost last message repeated 10 times Oct 26 05:33:38 localhost last message repeated 9 times Oct 26 05:33:40 localhost /usr/local/sbin/kamailio[13594]: ERROR: sms [libsms_modem.c:282]: Modem is not registered to the network Oct 26 05:33:40 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_getsms.c:256]: something happend with the modem -> was reinit -> let's retry Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:356]: modem wants the PIN again! Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:370]: re -init the modem!! Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:221]: init modem nokia on /dev/ttyUSB0. Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:250]: Checking if Modem is registered to the network Oct 26 05:33:42 localhost /usr/local/sbin/kamailio[13594]: NOTICE: sms [libsms_modem.c:275]: Waiting 2 sec. before retrying so i cannot send ok receive sms. with contact :+84...@sipdomain my nokia phone :6020 i can send sms via nokia by gnokii. please help or suggest to get it wok. thanks alot.

2 1

Dialplan module sample
by Uriel Rozenbaum 27 Oct '10

27 Oct '10

Hey guys, Does someone have an example for rules used with this module? I understand how to use the module, but not how to fill the database. I intend to use some prefix matching and regex (if available). Thanks in advance! Uriel

2 1

Working with Variables.
by dotnetdub 27 Oct '10

27 Oct '10

Hi List, Where is the best place to lookup documentation about working with variables. I come from the if len(x$)=y then xxx xxx xxx End If school! Looking for the best place to lookup documentation for checking characters, lengths of strings and removing or prepending stuff based on the results. Thanks Brian

2 2

Sanity module documentation error?
by Sergey Okhapkin 26 Oct '10

26 Oct '10

The routing script has if(sanity_check("1159", "6")) { xlog("L_INFO","Bad message from $proto:$si:$sp\n$mb\n"); break; } Seems like sanity_check function returns true is sanity check passes, but not failed. Am I right? 0(14085) DEBUG: <core> [receive.c:145]: After parse_msg... 0(14085) DEBUG: <core> [receive.c:186]: preparing to run routing scripts... 0(14085) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=10 0(14085) DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [35]; uri=[sip:167147511@204.74.213.5:5090] 0(14085) DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: to body [<sip:167147511@204.74.213.5:5090> ] 0(14085) DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <73> <REGISTER> 0(14085) DEBUG: <core> [parser/msg_parser.c:199]: DEBUG: get_hdr_body : content_length=0 0(14085) DEBUG: <core> [parser/msg_parser.c:101]: found end of header 0(14085) DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=jzbyj 0(14085) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 0(14085) DEBUG: sanity [mod_sanity.c:217]: all sanity checks passed 0(14085) INFO: <script>: Bad message from udp:71.59.87.156:5066

3 7

Psuedo Variable.
by David J. 26 Oct '10

26 Oct '10

Hello List; I was following along the Kamailio Asterisk Realtime Integration; I modified the script to get it working on Kamailio v 3.1. One area I had a question about was here... We have the following block; # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $ad + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } If I change the line to read : $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $ad from : $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) I get the following error; loading modules under /usr/local/kamailio-3.1-proxy/lib/kamailio/modules_k/:/usr/local/kamailio-3.1-proxy/lib/kamailio/modules/ $ad 0(13715) ERROR: <core> [pvapi.c:445]: bad parameters 0(13715) : <core> [cfg.y:3409]: parse error in config file /usr/local/kamailio-3.1-proxy/etc/kamailio/kamailio.cfg, line 717, column 48-47: unknown script pseudo variable ERROR: bad config file (1 errors) I would imagine that if the "$au" pseudo variable is available the "$ad" variable would be available as well. Please help point out my mistake. Thanks. David.

6 10

Re: [SR-Users] Drouting usage
by Javier Gallart 26 Oct '10

26 Oct '10

Thanks Marius, that was it. Regards Javier On Mon, Oct 25, 2010 at 10:08 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Drouting usage (Javier Gallart) > 2. PSTN call (michel freiha) > 3. Re: Drouting usage (marius zbihlei) > 4. bflag under failure_route (kamailio 1.5) (I?aki Baz Castillo) > 5. Re: bflag under failure_route (kamailio 1.5) > (Daniel-Constantin Mierla) > 6. Re: bflag under failure_route (kamailio 1.5) (I?aki Baz Castillo) > 7. Re: bflag under failure_route (kamailio 1.5) > (Daniel-Constantin Mierla) > 8. Re: bflag under failure_route (kamailio 1.5) (I?aki Baz Castillo) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 25 Oct 2010 15:16:56 +0200 > From: Javier Gallart <jgallartm(a)gmail.com> > Subject: [SR-Users] Drouting usage > To: sr-users(a)lists.sip-router.org > Message-ID: > <AANLkTik8pWtu9uT1=mPb_zSnztej5kHzPrEVPubgzeY7(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello > > sorry if this sounds too "newbie". I'm interested in the drouting module. > The first INVITE is correctly forwarded to the gw foudn in the drouting > table: > if (is_method("INVITE") && !has_totag()) { > do_routing("0"); > record_route(); > } > However for subsequent messages (ACK, BYE) in the dialog I don't know how > to > keep track of the ip resolved by the drouting logic so the messages are > correctly forwarded to the same gw. I've made some tests using the dialog > module with no success. Any hint? > > Thanks in advance > > Javier >

1 0

ERROR
by michel freiha 26 Oct '10

26 Oct '10

Guys I'm getting the below error...Can anybody help please? Oct 22 12:27:17 localhost /usr/local/sbin/kamailio[28566]: ERROR: <script>: time_dbg:method=INVITE;time=1287750437;callid=Y2FjNDAyZDY4YTk5ZWQ1Yjc4MDk4ZGZiZjk5YjIyYmE. - in NAT out route Regards

2 1

Outbound load balancing
by Lee Archer 26 Oct '10

26 Oct '10

Hi, does Kamailio support any sort of outbound load balancing where you can set the resources available at a gateway and the system keeps a list of what's in use? Opensips has something similar but does kamailio? Thanks Lee

4 4

Re: [SR-Users] Redirecting 302 problem
by Iñaki Baz Castillo 25 Oct '10

25 Oct '10

(I forward and reply this mail to the maillist). 2010/10/26 Mino Haluz <mino.haluz(a)gmail.com>: > Maybe I have not explained it very clearly, but the issue was hidden > in using fix_nated_contact(). I commented it out and everything works > perfect :) Ok, that's a different problem. If you call fix_nated_contact() for a 302 then obviously Kamailio will rewrite the URI host part of the Contact header with the response public source address. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

Redirecting 302 problem
by Mino Haluz 25 Oct '10

25 Oct '10

Hi, my kamailio receives 302 from ip 1.1.1.1 with a correct Contact-header format 123@kamailiohost , but when it forwards to my cisco as, it suddenly changes the host part to 1.1.1.1 .. It is very strange because, this IP does not have anything common with the callee number. Do you know where could be a problem? Mino

3 2

bflag under failure_route (kamailio 1.5)
by Iñaki Baz Castillo 25 Oct '10

25 Oct '10

Hi, I know that handling bflags outside of branch_route is not a good idea. Anyhow I have a doubt: 1) In a normal route I set "setbflag(i:1)", so the bflag will be set for all the branches that could be generated for this incoming transaction. This also involves creating new branches manually under failure_route, am I right? Note: I do know that it would be much better just to use flag instead of bflags. 2) The outgoing transaction(s) fail so I enter into failure_route. If I inspect here bflag(i:1) will it be set or not? I remember that it will be set. 3) Then I call append_branch in failure_route, or perhaps call loockup("location") and I expect that all the generated branches will have the bflag set, am I right? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 8

Drouting usage
by Javier Gallart 25 Oct '10

25 Oct '10

Hello sorry if this sounds too "newbie". I'm interested in the drouting module. The first INVITE is correctly forwarded to the gw foudn in the drouting table: if (is_method("INVITE") && !has_totag()) { do_routing("0"); record_route(); } However for subsequent messages (ACK, BYE) in the dialog I don't know how to keep track of the ip resolved by the drouting logic so the messages are correctly forwarded to the same gw. I've made some tests using the dialog module with no success. Any hint? Thanks in advance Javier

2 1

Re: [SR-Users] Debug xmpp module with kamailio 3.0.4
by Huy Nguyen 25 Oct '10

25 Oct '10

Hi Robert and list, @Robert: thank you so much. After change some things as the guide of Robert, i can see more LM_DBG from other modules as: tm, core (msg_parser.c, t_lookup.c...... )...but what i need is the log from xmpp module (e.g xmpp.c, xmpp_component.c......). As the guide on xmpp page, i just can debug with xmpp_send_message command, and i can't check whether kamailio was registered successful as the component of local jabberd server. How can check this ? Thanks, Huy Nguyen www.htk-inc.com Date: Mon, 25 Oct 2010 13:25:14 +0200 From: robert(a)exa-omicron.nl To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Debug xmpp module with kamailio 3.0.4 On 10/25/2010 12:31 PM, Huy Nguyen wrote: log_facility=LOG_LOCAL0 How can I check all LM_DBG logs in some files as xmpp.c, xmpp_component.c .... to view the registration processing of kamailio to local jabberd s erver ? Huy, In the default syslog.conf of centos you have a line: *.info;mail.none;authpriv.none;cron.none /var/log/messages Which only sends log level info and lower to /var/log/messages. If you add a line: local0.* -/var/log/kamailio.log You will find all any debugging output of kamailio in /var/log/kamailio.log Don't forget to restart syslog Regards, Robert Verspuy -- Exa-Omicron Patroonsweg 10 3892 DB Zeewolde Tel.: 088-OMICRON (66 427 66) http://www.exa-omicron.nl

1 0

Debug xmpp module with kamailio 3.0.4
by Huy Nguyen 25 Oct '10

25 Oct '10

Hi all, About xmpp module on kamailio 3.0.4 (Centos server). I enabled DEBUG mode with level 7, but when check on /var/log/message I don't see any log from xmpp module (LM_DBG): #!ifdef WITH_DEBUG debug=7 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 If I change LM_DBG to LM_ERR, I just see 3 line as below: Oct 25 17:03:48 civilpol /usr/local/sbin/kamailio[31029]: ERROR: xmpp [xmpp.c:345]: cmd_send_message Oct 25 17:03:48 civilpol /usr/local/sbin/kamailio[31029]: ERROR: xmpp [xmpp.c:379]: message from <sip:101@sip.htk.com> Oct 25 17:03:48 civilpol /usr/local/sbin/kamailio[31029]: ERROR: xmpp [xmpp.c:388]: using R-URI as destination How can I check all LM_DBG logs in some files as xmpp.c, xmpp_component.c .... to view the registration processing of kamailio to local jabberd server ? Thanks, Huy Nguyen www.htk-inc.com

2 1

Re: [SR-Users] dictionary attacks
by Daniel-Constantin Mierla 24 Oct '10

24 Oct '10

probably omitted by mistake, but please keep the mailing list cc-ed. On 10/24/10 3:38 PM, Sergey Okhapkin wrote: > Note that I check return code of www_authorize to be -1 (invalid user) and > block IP in this case only. Other error codes should not block the IP address. This one remembered me that in 3.1 we merged the auth modules and we used the one coming from ser because it has better nonce protection and other enhancements than kamailio version. That means the return codes have changed, the new ones are listed now at: http://kamailio.org/docs/modules/stable/modules_k/auth_db.html#id2753068 Added also note in migration wiki page: http://www.kamailio.org/dokuwiki/doku.php/install:3.0.x-to-3.1.0#modules_k_… Cheers, Daniel > On Sunday 24 October 2010, you wrote: >> I watched live an attack on voipuser.org while running 3.1 before >> release. It lasted 18 hours. I didn't want to ban it because was useful >> for testing and see if it reveals any weak. In most of the cases it hit >> pike module. I got some data and plan to make an article about it soon. >> >> Anyhow, as a result of that, default config for kamailio has a section >> for detecting and banning such "bad" IPs, using pike to detect floods >> and htable to keep it blocked. Search WITH_ANTIFLOOD directive. It can >> be enhanced like you pointed here, so if the authorize fails, add the IP >> in the banned list stored in htable. >> >> Using fail2ban together with IP tables has the advantage of dropping the >> packets before getting to application and eating cpu, although in the >> case of voipuser.org the cpu was not affected much - the rate was >> 170-200 requests per second. >> >> Cheers, >> Daniel >> >> On 10/24/10 3:06 PM, Sergey Okhapkin wrote: >>> I'm second for fail2ban. I block IP addresses with failed registration >>> attempts for 1 hour. Here is my setup: >>> >>> kamailio.cfg: >>> >>> if (is_method("REGISTER")) { >>> if(www_authorize("", "subscriber")< 0) { >>> if($rc == -1) { >>> xlog("L_INFO","Invalid username from >>> $proto:$si:$sp\n"); sl_send_reply("200","OK"); >>> } else >>> www_challenge("", "0"); >>> exit; >>> } >>> .... >>> >>> /etc/fail2ban/filter.d/openser.conf: >>> >>> [Definition] >>> #_daemon = kamailio >>> failregex = Invalid username from ...:<HOST>: >>> >>> /etc/fail2ban/jail.conf: >>> >>> findtime = 600 >>> >>> [openser-iptables] >>> enabled = true >>> filter = openser >>> action = iptables-allports[name=OPENSER, protocol=all] >>> logpath = /var/log/openser/openser # Replace with your sr log location >>> maxretry = 10 >>> bantime = 3600 >>> >>> On Sunday 24 October 2010, Uriel Rozenbaum wrote: >>>> Juha, >>>> >>>> I think we should be specially careful about black-lists. We receive >>>> many of these attacks in a per-day basis and a lot of them are from >>>> residential addresses or university, so I'm guessing some kind of worm >>>> or trojan performing the attack from various IPs. >>>> >>>> If you have the time, try fail2ban deamon. It can relate some >>>> brute-force events and act accordingly blocking an IP on iptables, >>>> executing a script. You send to "jail" those addresses for a period of >>>> time, then you can get them out again; and of course you can manually >>>> revert. >>>> >>>> Last, as a description of the attacks I saw, first it runs an NMAP >>>> like scan checking which IPs answer from 5060, then it starts sending >>>> registers (usually asterisk answers 404 if the user does not exist), >>>> then when the proxy challenges, it interprets the user is found and >>>> starts making dictionary attacks on the password (1234, admin, and so >>>> on). Keep safe complicated passwords, make kamailio challenge >>>> everything and you'll be safe. and again, fail2ban is a pretty good >>>> solution for brute force. >>>> >>>> This might help you finding a solution for your attacks. >>>> >>>> Cheers, >>>> Uriel >>>> >>>> On Sun, Oct 24, 2010 at 8:54 AM, Juha Heinanen<jh(a)tutpro.com> wrote: >>>>> while doing some tests, i noticed that one of my proxies started to >>>>> receive lots of register requests with different user names starting >>>>> from a letter. there was also invite attempts in the logs. they came >>>>> from ip 202.82.16.99 which according to traceroute is somewhere in >>>>> china. >>>>> >>>>> should we start publishing a black list of these attack ip addresses? >>>>> >>>>> -- juha >>>>> >>>>> _______________________________________________ >>>>> 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 >>>> _______________________________________________ >>>> 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 >>> _______________________________________________ >>> 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

3 7

dictionary attacks
by Juha Heinanen 24 Oct '10

24 Oct '10

while doing some tests, i noticed that one of my proxies started to receive lots of register requests with different user names starting from a letter. there was also invite attempts in the logs. they came from ip 202.82.16.99 which according to traceroute is somewhere in china. should we start publishing a black list of these attack ip addresses? -- juha

6 10

Reverse Filterlists idea
by "Nicolas Rüger" 23 Oct '10

23 Oct '10

Hello, Marius Zbihlei has patched the userblacklist module, so that it can handle characters as well (NOT in main release 3.1 yet). Thanks Marius. Therefore it might now be possible to filter general SIP-URIs!?? My idea is simple and described here. Please give some Feedback!!! As the "prefix"-functionality up to now was referring to SIP-URIs consisting of digits (=real telephon numbers) like 004930123456(a)1and1.de the general URI might be something like user_123(a)server.domain.de So far the "prefix"-functionality works because usernames that are real telephone numbers, that always start with country-code, followed by town- or regional-code and end with a unique number of the user (in this example 0049=Germany, (0)30=Berlin, 123456=user's unique number). To use "prefix"-functionality with general (including non-digits) URIs it must be evaluated reverse (from back to front) as the domain ends with the country- or top-level-domain and becomes more detailed the reverse way. So the IDEA is: Insert the domain as prefix in userblacklist-table in reverse, to use the functionality. I would use a perl-script to reverse the SIP-URI of the calling party in routing logic and then check it against the already reverted domain in the userblacklist-table. Do you think this is a good/well-working idea??? Any concerns or suggestions are appreciated... Regards, Nicolas -- GMX DSL Doppel-Flat ab 19,99 €/mtl.! Jetzt auch mit gratis Notebook-Flat! http://portal.gmx.net/de/go/dsl

4 8

Re: [SR-Users] Kamailio unable to forward Register request to asterisk
by Ovidiu Sas 22 Oct '10

22 Oct '10

You need to have no 'secret' field in the asterisk sipusers view. Please keep the list in cc. Private e-mail will go unanswered. Regards, Ovidiu Sas On Fri, Oct 22, 2010 at 1:53 PM, Sundeep Kumar Mishra <sundeep123kumariiit(a)gmail.com> wrote: > Hi, > Thanks for you quick reply. > We understand where we had been wrong. Since it was not quite clear in > the tutorial could u tell us how to disable authentication in asterisk? > Regards, > Sundeep Kumar Mishra > > > On Fri, Oct 22, 2010 at 7:29 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: >> >> If you use that particular config, you need to disable authentication >> on the asterisk side. >> Just read careful the tutorial. >> >> Regards, >> Ovidiu Sas >> >> On Fri, Oct 22, 2010 at 9:15 AM, Sundeep Kumar Mishra >> <sundeep123kumariiit(a)gmail.com> wrote: >> > Hi all, >> > I have installed an asterisk server at 10.3.3.143:5080 and a >> > kamailio at 10.3.3.85:5060. I am using sip communicator as softphone SIP >> > Client. It is able to register at kamailio but the kamailio is not able >> > to >> > forward register request to asterisk. >> > >> > We have gone through the tutorial >> > >> > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb >> > and used the configuration file after changing the asterisk ip and >> > kamailio >> > ip. >> > >> > We found out that when a normal softphone authenticates directly to >> > asterisk >> > it sends and recieve >> > >> > |--softphone -- REGISTER >> > |--asterisk -- 401 Unauthorised . CHallenge >> > |--softphone -- REGISTER - Response to Challenge >> > |--asterisk -- 200 OK (If response was correct) >> > >> > But I saw that when kamailio forwards register to asterisk, it sends >> > first >> > REGISTER packet but doesn't second the second one. It sends SUBSCRIBE >> > instead of REGISTER with response to the challenge. >> > >> > >> > Regards, >> > Sundeep Kumar Mishra >> > >> > _______________________________________________ >> > 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 >> > >> > > >

2 1

Kamailio unable to forward Register request to asterisk
by Sundeep Kumar Mishra 22 Oct '10

22 Oct '10

Hi all, I have installed an asterisk server at 10.3.3.143:5080 and a kamailio at 10.3.3.85:5060. I am using sip communicator as softphone SIP Client. It is able to register at kamailio but the kamailio is not able to forward register request to asterisk. We have gone through the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdba… used the configuration file after changing the asterisk ip and kamailio ip. We found out that when a normal softphone authenticates directly to asterisk it sends and recieve |--softphone -- REGISTER |--asterisk -- 401 Unauthorised . CHallenge |--softphone -- REGISTER - Response to Challenge |--asterisk -- 200 OK (If response was correct) But I saw that when kamailio forwards register to asterisk, it sends first REGISTER packet but doesn't second the second one. It sends SUBSCRIBE instead of REGISTER with response to the challenge. Regards, Sundeep Kumar Mishra

3 2

Error in syslog: BUG tm [t_msgbuilder.c:351]: unhandled reason cause -18344
by Mikko Lehto 22 Oct '10

22 Oct '10

Hi I am testing certain UAC behavior on proxy initiated 408 response with SIP Router. Call canceling is working fine - UAC gets 408 and GW gets CANCEL. However this error message is seen in syslog after fr_inv_timeout fires: BUG: tm [t_msgbuilder.c:351]: unhandled reason cause -18344 In script I set t_set_fr before t_relay, otherwise it is default: route[RELAY] { if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_TEST"); } t_set_fr(4000); if (!t_relay()) { sl_reply_error(); } exit; } SIP Router is self compiled from Git: sip-router 3.1.0 (i386/linux) 34d2f6 Here are TM parameters: #modparam("tm", "local_cancel_reason", "0") modparam("tm", "noisy_ctimer", 1) modparam("tm", "auto_inv_100_reason", "tryink") modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) The above error message is found in t_msgbuilder.c on #ifdef CANCEL_REASON_SUPPORT -block. Looks like I do not have compiled in support for that new feature as I get error when uncommenting local_cancel_reason tm parameter: ERROR: <core> [modparam.c:150]: set_mod_param_regex: parameter <local_cancel_reason> not found in module <tm> Is this bug as log message suggests? -- Mikko Lehto Setera

2 4

Siremis user guide
by Shrouk Khan 22 Oct '10

22 Oct '10

hi has anyone ever found any user guide for siremis ? it seems to take information directly from database and showing it . there is no tool tip or any clue as to what is what -- 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

3 4

[OT] Fwd: [dispatch] proposed SIXPAC charter
by Victor Pascual Avila 22 Oct '10

22 Oct '10

---------- Forwarded message ---------- From: Elwell, John <john.elwell(a)siemens-enterprise.com> Date: Wed, Oct 20, 2010 at 9:45 AM Subject: Re: [dispatch] proposed SIXPAC charter To: Peter Saint-Andre <stpeter(a)stpeter.im>, "dispatch(a)ietf.org" <dispatch(a)ietf.org> I find this a worthwhile topic to pursue. I had been wondering whether this activity would turn out to be more of a profiling exercise, and whether the IETF might not be the best choice of venue for such work. >From the current draft charter it looks like there will be at least some protocol extension work, for which I believe the IETF is the correct venue. On the other hand, the Unified Communications Interoperability Forum (UCIF) is seeking to advance the state of play on XMPP interoperability, and if we were just talking about a profile or BCP, that might have been a better venue. Perhaps the IETF should focus on requirements and protocol extensions, and consider whether the BCP work would be better done elsewhere. Or at least, there should be some coordination with other activities relating to XMPP interoperability. John > -----Original Message----- > From: dispatch-bounces(a)ietf.org > [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre > Sent: 19 October 2010 21:17 > To: dispatch(a)ietf.org > Subject: [dispatch] proposed SIXPAC charter > > Earlier this year, some folks in the RAI area proposed an > initiative to > define a few small extensions to both SIP and XMPP that would make it > easier to develop and deploy dual-stack SIP+XMPP endpoints. Based on > feedback provided on the DISPATCH list and received from the RAI ADs, > I've taken the liberty of rewriting the proposed charter, in the hopes > that fresh text will spur a more conclusive discussion. I'm > mostly just > trying to help the proponents put their best foot forward, so if folks > here have more feedback I'd expect that people like Simo Veikkolainen > and Emil Ivov will be able to engage in further discussion. > > /psa > > ### > > SIXPAC (SIP Integration with XMPP in Presence Aware Clients) > ============================================================== > =========== > > Problem Statement > > Both the Session Initiation Protocol (SIP) and the Extensible > Messaging > and Presence Protocol (XMPP) are widely deployed technologies for > real-time communication over the Internet. In order to offer > a complete > suite of features as well as communication across multiple networks, > several user-oriented software applications support both SIP and XMPP, > and more software developers have expressed interest in building such > "dual-stack" solutions. Unfortunately, it is difficult to provide a > good end-user experience in such applications because SIP and XMPP are > not aware of each other. For example: > > - XMPP presence does not include availability states related to a SIP > voice call or video call (e.g., "on the phone"), thus preventing an > a dual-stack endpoint from showing presence-based > communication hints > > - There is no correlation between an XMPP IM session and a SIP voice > or video session, thus preventing a dual-stack endpoint > from providing > integrated user interfaces and communications history > > - SIP accounts and XMPP accounts are not directly correlated > in contact > lists or vCards (and not all deployed services support > storage of such > information), thus preventing a dual-stack endpoint from > knowing that > a contact has both SIP and XMPP capabilities > > Although some proprietary solutions exist to the foregoing > problems, it > would be preferable to define standardized solutions for the sake of > improved interoperability. > > Objectives > > Because both SIP and XMPP are easily extended through new SIP headers > and XMPP elements, it should be possible to provide tighter > integration > within dual-stack SIP/XMPP user agents to improve the user experience. > > Any such extensions should meet the following criteria: > > - Be completely optional and backwards-compatible for all endpoints > > - Work without changes to deployed infrastructure such as existing > SIP and XMPP servers, B2BUAs, firewalls, etc. > > The SIXPAC WG will define a small number of SIP and XMPP extensions to > solve the following use cases in dual-stack endpoints: > > - Including SIP-based availability states in XMPP presence (limited to > basic presence and availability states only, not the full range of > PIDF extensions) > > - Correlating an XMPP IM session with a SIP voice/video session, and > vice-versa > > - Advertising a SIP account address over XMPP and an XMPP account > address over SIP > > Additional use cases are out of scope, including anything > related to or > requiring server integration, multiparty communication, SIP-based IM > and presence, XMPP-based voice and video, file transfer, generalized > service discovery and capabilities exchange, full protocol translation > in communication gateways, shared credentials across both SIP and XMPP > accounts, rich presence extensions for features such as geolocation, > etc. Although such topics are important and interesting, they are out > of scope for this group. > > However, in addition to the protocol extensions explicitly mentioned > above, the group may also define best practices related to the > implementation and deployment of dual-stack SIP/XMPP endpoints, > including topics such as user agent configuration. > > Deliverables > > - Use cases and protocol requirements > > - XMPP presence extension for SIP-based availability states > > - Media session correlation extensions for SIP and XMPP > > - Contact address advertisement extensions for SIP and XMPP > > - Best practices for implementation and deployment of dual-stack > endpoints > > Milestones > > Feb 2011 Submit use cases and protocol requirements document to > IESG (Informational) > > Oct 2011 Submit XMPP presence extension document to IESG > (Standards Track) > > Oct 2011 Submit media session correlation extensions document to IESG > (Standards Track) > > Oct 2011 Submit contact address advertisement extension document to > IESG (Standards Track) > > Oct 2011 Submit best practices document to IESG (Informational) > > ### > > > _______________________________________________ dispatch mailing list dispatch(a)ietf.org https://www.ietf.org/mailman/listinfo/dispatch -- Victor Pascual Ávila

5 40

Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy
by Daniel-Constantin Mierla 22 Oct '10

22 Oct '10

Hello, apart of the email with logs, or private data that was required, please continue the discussion on public mailing list. I need some time to check the logs, the first inconvenience is the rar archive, a zip or tgz is much better -- not a fan of installing binaries from unknown companies on mac os x, going to get unrar from macports. Then solaris is not an OS i have at hand nor use frequently, so need to read about. It is also why using mailing list keeps you in touch with others that can help. Cheers, Daniel On 8/18/10 7:04 PM, KevinJin wrote: > Hi Daniel, > > Does the logs have any hint for what's cause of the issue? I didn't > include the user list in the previous email since logs have the actual > IP info. > > Thanks in advance! > > Best Regard > Kevin > > ------------------------------------------------------------------------ > From: kevin.jzh(a)hotmail.com > To: miconda(a)gmail.com > Subject: RE: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > Date: Tue, 17 Aug 2010 18:51:02 +0800 > > > Hi Daniel, > > Attached are the two kamailio logs, > unix_socket_log --- run rtp proxy wth -s unix:/tmp/rtpproxy.sock > udp_rtpproxy_log --- run rtp proxy with -s udp:*:7722 > > Please help to check what's wrong with it. > > Thanks, > Kevin > ------------------------------------------------------------------------ > Date: Tue, 17 Aug 2010 11:29:46 +0200 > From: miconda(a)gmail.com > To: kevin.jzh(a)hotmail.com > CC: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > please send full log at startup, your snippets include just few lines > per process, being mixed from different processes. > > Cheers, > Daniel > > > On 8/14/10 3:54 AM, KevinJin wrote: > > Hello, > > ------------------------------------------------------------------------ > Date: Thu, 12 Aug 2010 22:16:18 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: kevin.jzh(a)hotmail.com <mailto:kevin.jzh@hotmail.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > On 8/12/10 8:34 PM, KevinJin wrote: > > Hello, > > ------------------------------------------------------------------------ > Date: Thu, 12 Aug 2010 18:46:19 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: kevin.jzh(a)hotmail.com <mailto:kevin.jzh@hotmail.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > On 8/12/10 4:47 PM, KevinJin wrote: > > Hi Daniel, > > What does the log below means? Does it mean nathelper has > issue to send the request to RTP proxy first or nathelper > doesn't receive a response after sending a request to the > rtp proxy? > 0(27429) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy > > this error is printed when write to socket fails. Do you have > any firewall running on the system? Is the user under which > kamailio runs allowed to write to sockets? > > > There's no firewall on the system, and I run the kamailio > as root, > root 26310 1 0 02:24:19 ? 0:00 > /usr/local/kamailio-3.0.2/sbin/kamailio -f > /usr/local/kamailio-3.0.2/etc/kamail > > You can edit module_k/nathelper/nathelper.c and replace > the line 2457 with: > > LM_ERR("can't send command to a RTP proxy (%s/%d)\n", > strerror(errno), errno); > > Recompile and reinstall. Hopefully will get more hints > about what happens. > > > Here is the error message after the change: > 2(26312) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy(Invalid argument/22) > 2(26312) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > 2(26312) ERROR: nathelper [nathelper.c:3144]: no available proxies > what could be the cause? > > hmm, invalid argurment ... try with this line: > > LM_ERR("can't send command to a RTP proxy (%s/%d) [sock %d (%d), > vcnt %d]\n", > strerror(errno), errno, rtpp_socks[node->idx], node->idx, vcnt); > > maybe will give some hints about which value is invalid. > > Here's the log after the change: > 4(14415) ERROR: nathelper [nathelper.c:2457]: can't send command > to a RTP proxy (Invalid argument/22) [sock 7 (0), vcnt 18] > 4(14415) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.x.y:7722> does not respond, disable it > 4(14415) ERROR: nathelper [nathelper.c:3144]: no available proxies > > > Can you try as well with an unix file socket: > > modparam("nathelper", "rtpproxy_sock", "unix:/tmp/rtpproxy.sock") > > then start rtpproxy with -s unix:/tmp/rtpproxy.sock > > > 4(17530) INFO: nathelper [nathelper.c:2369]: rtp proxy > <unix:/tmp/rtpproxy.sock> found, support for it re-enabled > 3(17529) ERROR: nathelper [nathelper.c:2429]: can't send command > to a RTP proxy > 3(17529) ERROR: nathelper [nathelper.c:2492]: proxy > <unix:/tmp/rtpproxy.sock> does not respond, disable it > 3(17529) ERROR: nathelper [nathelper.c:3144]: no available proxies > > Thanks, > Kevin > > I have no solaris (sparc) to try myself... > > Cheers, > Daniel > > > > Test env: > UA1 (Behind NAT) --------> Kamailio & RTPproxy (Public IP) > --------->UA2 (Public IP) > > Thanks, > Kevin > Cheers, > Daniel > > > > 0(27429) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > > There's no problem for the resource(CPU, mem etc.) on the > server, the load is very low. > > Thanks in advance! > ---------- > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) ERROR: nathelper [nathelper.c:3144]: no available > proxies > 0(27429) ERROR: nathelper [nathelper.c:2627]: no available > proxies > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) INFO: nathelper [nathelper.c:2369]: rtp proxy > <udp:210.13.124.15:7722> found, support for it re-enabled > 0(27429) DEBUG: nathelper [nathelper.c:3196]: proxy reply: > 42040 210.13.124.14 > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy > 0(27429) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > 0(27429) ERROR: nathelper [nathelper.c:3144]: no available > proxies > 0(27429) ERROR: nathelper [nathelper.c:2627]: no available > proxies > > Thanks, > Kevin > ------------------------------------------------------------------------ > > > _______________________________________________ > 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/ > > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ -- Daniel-Constantin Mierla http://www.asipto.com/

2 5

Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 22 Oct '10

22 Oct '10

Hi all, Sorry for restore this topic, but i till confused about the guide on xmpp module page. I'm integrating xmpp module to kamailio 3.0.x at my local network, with server mode. The kamailio server have domain: kamailio1.com (192.168.1.26), the xmpp server have domain: xmpp1.com (192.168.1.20). SIP user: 101(a)kamailio1.com XMPP user: 109(a)xmpp1.com Below is the part mention in xmpp doc I confused about, so i'm pasting that: In the destination address we need an extra character as a domain separator, apart from @. The address of the XMPP(jabber) client looks like this: "sip:username<domain_separator>jabber_server@gateway_domain". The address of a SIP client has the following pattern : "sip_username<domain_separator>openser_domain@xmpp_domain"; A common used character in XMPP transports is "%". It mean, from sip client 101(a)kamailio1.com, if I want to chat with xmpp client 109(a)xmpp1.com, I'll add 109 to the contact list of 101 as: sip:109%xmpp1.com@kamailio1.com. It right ? As the similar, i'll add to the buddy list of 109 as: 101%kamailio1.com(a)xmpp1.com ? Can anyone suggest how to test xmpp module ? Thanks, Huy Nguyen www.htk-inc.com

2 6

Register a different URI than the received in the To header
by Iñaki Baz Castillo 21 Oct '10

21 Oct '10

Hi, not sure if this question has already taken place in this maillist: For some exotic reason sometimes the registrar receives a REGISTER with "To: sip:alice@domain.com" but I want to save it in the location table as "sip:bob@domain.com". AFAIK there is no way to do it, am I wrong? -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 3

Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician
by JR Richardson 21 Oct '10

21 Oct '10

We have a couple of positions open, please respond to the posting if qualified and interested. http://www.ntegrated.net/resources/job-opportunities/field-service-install-… http://www.ntegrated.net/resources/job-opportunities/network-engineering-vo… These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses

1 0

Kamalio 3.1 based turn-key distribution released !
by X- x -X 21 Oct '10

21 Oct '10

Hey all, Just wanted to share this link. A distribution based on debian (lenny) & kamailio 3.1 pre-configured with presence server/RTP thing has been released. size: 245 MB. http://ciitix.ciit.net.pk/index.php/home/6-announcements-/14-ciitix-voip-10… Download link http://ciitix.ciit.net.pk/index.php/downloads Cheers --- X

2 1

siremis 1.0.1 with kamalio 3.1 ?
by X- x -X 21 Oct '10

21 Oct '10

Hi all, Not sure if this should be put to siremis list or dev list or whatever but bear with me if you could. Since siremis v 2.0 is not out. Is siremis 1.0.1 known to work fine with kamalio 3.1 ? (ref: siremis 1.0.1 install http://siremis.asipto.com/install/ ) Is it ready for production deployment etc.. ? Will greatly appreciate any response from anyone since Ramona seems to be on her sabbatical . Thanks. --

2 1

siremis 2.0 ?
by X- x -X 21 Oct '10

21 Oct '10

Hey all, Congratulation on 3.1 release ! Any news on when siremis v 2.0 is hitting the stores ? In the mean time what about alpha/beta version of it ? Its nowhere on the repos/sites (asipto etc..) Is version 1.0.1 know to work smoothly with kamailio 3.1 ? Cheers --

2 1

HTable 'size' parameter
by Iñaki Baz Castillo 21 Oct '10

21 Oct '10

Hi, in the doc of HTable module all the examples show "a=>size=4;". As the doc says: size - number specifying the size of hash table. The number of entries in the table is 2^size This is, the examples just allow 16 concurrent entries, which is not enough in most of the cases. Did I misunderstand something? or do the module examples use a non very pertinent value? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

5 8

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

1 0

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

1 0

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/

4 7

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

2 2

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

4 6

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

2 2

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

2 2

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.

3 6

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

2 2

#!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

3 2

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

7 21

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

4 5

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

4 7

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

2 2

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

4 25

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

3 7

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