sr-users March 2010

sr-users@lists.kamailio.org

60 participants
117 discussions

[Kamailio-Users] Kamailio - SIP Router on GSoC 2010

by Daniel-Constantin Mierla

Hello, I have the pleasure to announce that our project has an accepted application for Google Summer of Code 2010. The proposed application was done within SIP Communicator Organization and is related to extending the presence server for conference calls notifications. See more: http://www.sip-communicator.org/index.php/GSOC2010/Kamailio4575 http://www.sip-communicator.org/index.php/Development/Gsoc2010 The friends from SEMS project will be part as well, with separate application, implementing the audio mixer part. More details will be published soon. Start thinking about good candidates (they must be students (college or university)), we need them to get the project accepted for implementation. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

14 years, 8 months

[Kamailio-Users] uac_replace_from unexpected behavior

by Brandon Armstead

Hello, As always thank you ahead of time for your help and input! I am currently calling uac_replace_from("", "") in effort to "leave uri" and "toss away display name" Which does seem to work....... for the initial INVITE However upon receiving an ACK with an empty display, however "" <- quotations, it does not clear the display "" which is causing issues with one of my upstream vendors. Example / Scenario: From: "" <sip:uri@host> Expected Result upon uac_replace_from("",""): From: <sip:uri@host> Current Result: From: "" <sip:uri@host> As you can see it is not stripping the "" empty display quotes. Any thoughts / ideas / suggestions to get my desired affect? Sincerely, Brandon Armstead

14 years, 8 months

[Kamailio-Users] Rtpproxy/Kamailio modification to support high capacity encryption, transcoding

by Vikram Ragukumar

Hello, We're making initial modifications to rtpproxy to support high channel capacity transcoding and encryption. At this point we want to get some general idea of the scope of changes needed for rtpproxy and Kamailio... so we're starting with small steps. We've been studying rtpproxy source and our current thinking is to add a sub-structure to the existing rtpp_session structure (defined in rtpp_session.h). The new sub-structure would include: -encryption options (type, key length, salt size, type of key mgt protocol, etc) -encode / decode options (type, VAD/CNG, VIF size, etc) Any comments or advice on this approach appreciated. Not sure whether to start a separate thread, but also there is the issue of what changes are necessary to Kamailio to support sending updated commands to rtpproxy. Would modifying Nathelper alone be sufficient? Thanks and Regards, Vikram. PS : I'm posting on Kamailio's mailing list because it seems that both Kamailio and rtpproxy developers closely follow this list.

14 years, 8 months

[Kamailio-Users] SIP Router Project of VUC, March 19

by Daniel-Constantin Mierla

Hello, this Friday, March 19, late afternoon, the weekly VoIP User Conference is hosting a session about SIP Router project. My goals are to present the achievements so far within SIP Router projects, what is new in Kamailio 3.0 release and plans for the future. More details can be found at: http://www.kamailio.org/w/vuc-the-sip-router-project/ You can join the audio conference via sip, skype, pstn line or other several options presented on http://vuc.me site. There is a irc channel available for it: #vuc on irc.freenode.net. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 * http://www.asipto.com/index.php/sip-router-masterclass/

14 years, 8 months

[Kamailio-Users] kamailio logo artwork

by Daniel-Constantin Mierla

Hello, courtesy of Elio Rojano (http://www.sinologic.net), very nice logo artworks are available now, using openser or sip-router thematics. You can browse at: http://www.asipto.com/gallery/v/ksr-artwork/ Available for download as well at: http://www.kamailio.org/pub/ksr-artwork/ If you are good at graphic design and have new ideas, please submit. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

14 years, 8 months

[Kamailio-Users] Kamailio 3.0.0 Install Error

by Nathaniel L Keeling

I am trying to install the new release of Kamailio 3.0.0 on Solaris 10. When I perform the install, I am getting this error. Here is the command that I am entering for the install: make prefix=/usr/local/kamailio-3.0.0 INSTALL=install group_include="standard standard-dep postgres" CPU=ultrasparc install install mode make[2]: Entering directory `/usr/local/src/kamailio-3.0.0/lib/kcore' make[2]: Nothing to be done for `install-if-newer'. make[2]: Leaving directory `/usr/local/src/kamailio-3.0.0/lib/kcore' touch /usr/local/kamailio-3.0.0/lib/kamailio/modules_k/speeddial.so install -m 755 speeddial.so /usr/local/kamailio-3.0.0/lib/kamailio/modules_k make[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/modules_k/speeddial' mkdir -p /usr/local/kamailio-3.0.0/etc/kamailio/ # other configs /bin/sh: syntax error at line 1: `;' unexpected make: *** [install-cfg] Error 2 I am new to Kamailio and would appreciate any help. Thanks Nathaniel

14 years, 8 months

Re: [Kamailio-Users] Is Timestamp supported in kamailio?

by Iñaki Baz Castillo

2010/3/16 mustafa rifaee <mustafa.rifaee(a)gmail.com>: > Hi all, > I would like to asking you Hi, it's just better to keep the thread in the maillist. > about the Time-stamp header in SIP messages, and > why this header does not supported in most open SIP servers, Is it > recommended or not? > Is this fact true ?? I think The problem is that both UAS and UAC must have > the same system time to generate a useful result. So therefore both or all > SIP devices (think about an intermediate proxy) must have NTP activated. That's not true. The request can contain a header like: TimeStamp: 25 and the proxy/server replies in the 100 Trying: TimeStamp: 25 0.001 This means that it has token 0.001 seconds to the server in order to process the request and generate the 100 response. -- Iñaki Baz Castillo <ibc(a)aliax.net> -- Iñaki Baz Castillo <ibc(a)aliax.net>

14 years, 8 months

[Kamailio-Users] [OT] MSN presence flow

by Iñaki Baz Castillo

Hi, does somebody have a MSN protocol flow related to presence rules or buddies management? This is, I would like to know how MSN protocol imlements some tasks as: - Adding a buddy. - Blocking a buddy for presence. - Blocking a contact (not a buddy) for presence. Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

14 years, 8 months

Re: [Kamailio-Users] kamailio crashes

by Panagiotis Skoulikaritis

Hello Daniel I do have quite a few core files, please send me the gdb commands. Regards Panagiotis. Daniel-Constantin Mierla wrote: > Hello Panagiotis, > > On 03/16/2010 02:21 PM, Panagiotis Skoulikaritis wrote: >> Hello Daniel >> >> I will need time to recreate the problem, >> attached are the only traces I kept. >> on the amaze-4.cap the calls are originated by a softphone registered >> on the kamailio >> on the crash.cap the calls are originated from the "PSTN". >> >> we do account the $pd and the $pn >> >> I reply to you directly since i don't want to give the traces on the >> mailing list, I hope you will understand. > > I asked to be sent private, it is ok. > >> >> Also for workaround Alex took out the PDT module from the route and >> the kamailio do not crash anymore. > > Hmm, so you say pdt module is related? That is pretty small and old > module... > > Do you still have the core file? I can send you some gdb commands to > get more details out of it. > > Thanks, > Daniel > >> >> Regards >> >> Panagiotis >> >> >> Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> can you send me ngrep/pcap file with ip addresses so I can match >>> which 200ok is causing the problem (coming from B or coming from >>> Asterisk)? The backtrace shows ip while the sip trace is masked. >>> >>> Also, I would need a bit more info from the core file. Please keep >>> one around. The issue seems to be related to P-Asserted-Identity >>> header, but I couldn't find any such header in the sip trace you sent. >>> >>> Are you accounting the PAI header? >>> >>> Thanks, >>> Daniel >>> >>> On 03/10/2010 03:30 PM, Panagiotis Skoulikaritis wrote: >>>> Dear Marius >>>> >>>> The scenario is as follows: >>>> >>>> 1. A Call is placed by a sip subscriber "A" >>>> 2. kamailio forwards the call to the asterisk server >>>> 3. Asterisk plays an IVR message on the subscriber "A", creates a >>>> new call to a "virtual" number which is forwarded to the kamailio >>>> server, and plays an ivr to this leg as well when the call is >>>> answered, then it connects the two calls. >>>> 4. Kamailio translates the "virtual" number to the pstn number of >>>> subscriber B >>>> >>>> >>>> I have attached a picture of the above scenario. >>>> >>>> The modules that are loaded are: >>>> >>>> loadmodule "db_mysql.so" >>>> loadmodule "mi_fifo.so" >>>> loadmodule "mi_datagram.so" >>>> loadmodule "sl.so" >>>> loadmodule "tm.so" >>>> loadmodule "rr.so" >>>> loadmodule "pv.so" >>>> loadmodule "maxfwd.so" >>>> loadmodule "usrloc.so" >>>> loadmodule "registrar.so" >>>> loadmodule "textops.so" >>>> loadmodule "uri_db.so" >>>> loadmodule "siputils.so" >>>> loadmodule "xlog.so" >>>> loadmodule "acc.so" >>>> loadmodule "dispatcher.so" >>>> loadmodule "pdt.so" >>>> loadmodule "dialplan.so" >>>> loadmodule "siptrace.so" >>>> loadmodule "dialog.so" >>>> loadmodule "sqlops.so" >>>> loadmodule "userblacklist.so" >>>> loadmodule "htable.so" >>>> loadmodule "uac.so" >>>> >>>> >>>> The config that does all the routing is : >>>> >>>> route[10] { >>>> >>>> xlog("alx ------- This is Route 10 -------"); >>>> >>>> if($rU =~ "^.*%+") >>>> { >>>> xlog("alx ------- The number contains %23 "); >>>> $rU = $(rU{re.subst,/^(.*)%23(.*)/\1\2/}); >>>> #$rU = $(rU{s.unescape.user}); #It changes the %23 >>>> to # !! >>>> xlog("alx ------- The perl $rU ------- "); >>>> } >>>> >>>> if($rU =~ "^.*#+") >>>> { >>>> xlog("alx ------- The number contains #"); >>>> $rU = $(rU{re.subst,/^(.*)#(.*)/\1\2/}); >>>> #$rU = $(rU{s.unescape.user}); #It changes the %23 >>>> to # !! >>>> xlog("alx ------- The perl $rU ------- "); >>>> } >>>> >>>> if(prefix2domain("2", "0")) { >>>> >>>> $var(dial_grp) = $(rd{s.select,0,.}{s.int}); # >>>> Dialplan group prefix for routing >>>> $var(num_pr) = $(rd{s.select,1,.}{s.int}); # The >>>> number of digits that prefix has >>>> $var(num_translation) = >>>> $(rd{s.select,2,.}{s.int}); # Called number translation >>>> $avp(s:port_translation) = >>>> $(rd{s.select,3,.}{s.int}); # Port number translation >>>> #$var(test_var) = $(rd{s.select,4,.}{s.int}); # >>>> Future property >>>> >>>> $avp(s:cust_prefix) = $(rU{s.substr,0,$var(num_pr)}); >>>> $rU = $(rU{s.substr,$var(num_pr),0}); >>>> >>>> xlog("alx ------- The new rU is $rU and properties $rd >>>> -------"); >>>> >>>> if($var(num_translation) == 1) >>>> { >>>> if($sht(a=>$rU)!=null){ >>>> >>>> $rU = $sht(a=>$rU); >>>> xlog("alx ------- Translation Done. DST >>>> num=$rU ----------"); >>>> >>>> } else { >>>> xlog("alx ------- Translation NOT Done >>>> ----------"); >>>> } >>>> >>>> >>>> #xlog("alx ------- We have DST number >>>> translation for user fU $avp(s:frm_user_name) ----------"); >>>> #if(dp_translate("31", "$rU/$rU")) >>>> #{ >>>> # xlog("alx ------- Translation Done. >>>> DST num=$rU ----------"); >>>> #} else { >>>> # xlog("alx ------- Translation NOT >>>> Done ----------"); >>>> #} >>>> } >>>> >>>> if(dp_translate("$var(dial_grp)", "$rU/$rU")) >>>> { >>>> xlog("alx ------- The $rU and with >>>> attributes :$avp(s:dest) -------\n"); >>>> >>>> $var(i) = 0; >>>> while($(avp(s:dest){s.select,$var(i),.})!="#") >>>> { >>>> $avp(s:dstgrp) = >>>> $(avp(s:dest){s.select,$var(i),.}{s.int}); >>>> $var(i) = $var(i) + 1; >>>> xlog("alx ------- The >>>> avp(s:dstgrp)=$avp(s:dstgrp) var(i)=$var(i) -------"); >>>> } >>>> >>>> # backup the username so we can use >>>> different prefixes >>>> $avp(s:user) = $rU; >>>> >>>> # select destination from first group >>>> >>>> if(ds_select_domain("$avp(s:dstgrp)", "4")) >>>> { >>>> >>>> if($(ru{uri.param,prefix})!=null) >>>> { >>>> $ru >>>> = "sip:" + $(ru{uri.param,prefix}) + $avp(s:user) + "@" + $rd; >>>> >>>> } else { >>>> $ru >>>> = "sip:" + $avp(s:user) + "@" + $rd; >>>> } >>>> } >>>> >>>> $avp(s:dstgrp) = null; >>>> xlog("alx ------- The final >>>> RURI is $ru ------- "); >>>> if($avp(s:port_translation) >>>> == 1) >>>> { >>>> rewriteport("5061"); >>>> } >>>> t_on_failure("3"); >>>> t_relay(); >>>> exit; >>>> >>>> } >>>> >>>> >>>> >>>> >>>> } >>>> >>>> >>>> } >>>> >>>> Attached is the trace >>>> >>>> Regards. >>>> >>>> P. >>>> >>>> marius zbihlei wrote: >>>>> Panagiotis Skoulikaritis wrote: >>>>>> Hello Daniel >>>>>> >>>>>> the kamailio version is 1.5.3 >>>>>> >>>>>> Regards >>>>>> >>>>>> P. >>>>> Hello, >>>>> >>>>> Can you give us more details like the sip message that generates >>>>> the coredump (or if every sip message received generates the >>>>> core), if your config does something more out of the >>>>> ordinary(let's say exotic). Can we reproduce it ? >>>>> >>>>> It would also be helpful if you specify the list of modules you >>>>> have loaded. >>>>> >>>>> Cheers, >>>>> Marius >>>>>> >>>>>> Daniel-Constantin Mierla wrote: >>>>>>> Hello, >>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (OpenSER) - Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> -- >>> Daniel-Constantin Mierla >>> Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 >>> * http://www.asipto.com/index.php/sip-router-masterclass/ >>> > > -- > Daniel-Constantin Mierla > Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010 > * http://www.asipto.com/index.php/sip-router-masterclass/ >

14 years, 8 months

Re: [Kamailio-Users] Is Timestamp supported in kamailio?

by Iñaki Baz Castillo

2010/3/16 mustafa rifaee <mustafa.rifaee(a)gmail.com>: > Hi Castillo, > I would like to asking you about the Time-stamp header in SIP messages, and > why this header does not supported in most open SIP servers, Is it > recommended or not? > Is there any open source support the time stamp header. > > Is this fact true ?? I think The problem is that both UAS and UAC must have > the same system time to generate a useful result. So therefore both or all > SIP devices (think about an intermediate proxy) must have NTP activated. > > Please Help me; I think I *already* helped you as you *already* asked me directly. Please, keep the threads in the maillist, nobody likes to give private free support. -- Iñaki Baz Castillo <ibc(a)aliax.net>

14 years, 8 months

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sr-users March 2010