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37456 discussions

perl module compilation on centos
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, while building the v3.1.5 RPMs, the perl module failed to compile on centos 5 and 6, with following error: error: `PL_unitcheckav' undeclared (first use in this function ) so its packaging was disabled. It works fine on opensuse and fedora. Someone on twitter tried with the package perl-ExtUtils-Embed (like in fedora) and worked. However, OpenSUSE Build System fails to find that package. Is anyone here with latest centos 5 or 6 that can try compilation of perl module in v3.1.5 and let me know the result? Few months ago, it didn't fail for v3.1.4. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Kamailio and FreeSWITCH realtime integration, tutorial?
by Henrik Aagaard Sørensen 22 Sep '11

22 Sep '11

But FreeSwitch don't use MySQL like Asterisk? So adapting isn't that easy or...??? On 21/09/2011, at 16.18, David <david(a)styleflare.com> wrote: You can adapt pretty easily. On 9/21/11 3:46 PM, Henrik Aagaard Sørensen wrote: Does anyone know if there somewhere exists a tutorial about Kamailio and FreeSWITCH realtime integration? I have Googled a lot and found: http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index On the same site there is a tutorial for Kamailio and Asterisk realtime integration, but not for FreeSWITCH. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://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

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Re: [SR-Users] Problem with REGISTER retransmissions
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, I tried to reproduce with a very simple config to handle register requests. So I sent a REGISTER with expires 60 (very much the same as the one in the trace you sent), waited a while, then sent two REGISTERs having the CSeq incremented. All went fine. I tested with master and 3.1 branches (so the last is actually 3.1.5). Can you upgrade to 3.1.5 first and try again? If still fails, then update config to print the return code for save() function, something like: $var(r) = save("location"); xlog("code returned by save is $var(r)\n"); if($var(r) < 0) send_reply("500", "Failure"); Let me know the value of the returned code for the register that does not have a contact in reply. Cheers, Daniel On 9/20/11 2:10 PM, Alejandro Mingo wrote: > > Hi Daniel-Constantin, > > I send you the PCAP and LOG files corresponding to one of these > situations. As you can see in the LOG file, one thread (5444) handles > the first REGISTER, other thread (5447) handles the second REGISTER > (retransmission) and a third thread (5451) indicates that the binding > has expired, six seconds after the arrival of both REGISTERs. > > Thanks again. > > Alejandro. > > > > El 20/09/2011 10:54, Daniel-Constantin Mierla escribió: >> Hello, >> >> On 9/20/11 10:13 AM, Alejandro Mingo wrote: >>> Hi, >>> >>> Since we installed last version of Kamailio (3.1.4) >> >> the last version is now 3.1.5, but this is not really the most >> important aspect. However, it is recommended to upgrade, there were >> some fixed to registration handling when register requests have >> multiple contacts. >>> we have been experiencing a big problem with REGISTER >>> retransmissions. When the server receives a retransmitted REGISTER >>> it removes the binding and the UAC remains unregistered until next >>> refreshing period. I include a PCAP capture that shows this behaviour. >>> >>> I am not 100% sure that this problem never happened with older >>> versions. >>> >>> Has anybody experienced the same problem? >> I will try to reproduce and investigate myself. Anyhow, meanwhile >> maybe you can provide more hints faster. Do you have error messages >> in the syslog? Can you run with debug=4 in config file and send the >> output in syslog for such a case? -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Kamailio 3.1.5 RPMS
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, just to let you know that the RPMs of Kamailio 3.1.5 were generated for CentOS, Fedora, OpenSUSE and RedHat. This time CentOS RPMs are packaged for both versions 5 and 6. Links are available on our wiki at: * http://www.kamailio.org/dokuwiki/doku.php/packages:rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Kamailio and FreeSWITCH realtime integration, tutorial?
by Henrik Aagaard Sørensen 22 Sep '11

22 Sep '11

Does anyone know if there somewhere exists a tutorial about Kamailio and FreeSWITCH realtime integration? I have Googled a lot and found: http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index On the same site there is a tutorial for Kamailio and Asterisk realtime integration, but not for FreeSWITCH.

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Re: [SR-Users] Dialog module showing calls that have already been terminated
by Timo Reimann 21 Sep '11

21 Sep '11

Hey Phillip, On 20.09.2011 13:48, Phillman25 Kyriacou wrote: > Thanks for your email. > > Yes dlg_manage(); has to now be called on INVITE and BYE/CANCEL messages. > Where would i have to call loose_route()? Only on INVITE? On *all* in-dialog requests, i.e., all requests which contain a To tag. This may include Re-INVITEs too (but not initial INVITEs). > My configuration did not change between 3.1.2 and 3.1.5. > > Call flow example: > ============== > > Cisco PGW ===> Kamailio 3.1.5 ===> VOIP PROVIDER or ASTERISK PABX > > > The below is my configuration. Let's take a look at it: > #!KAMAILIO > # [...] > if(is_method("BYE|CANCEL")) > > { > > dlg_manage(); > > > > # per request initial checks > route(REQINIT); > > # NAT detection > route(NAT); > > # handle requests within SIP dialogs > route(WITHINDLG); [...] > # 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()) { [...] So route[WITHINDLG] contains the logic to track in-dialog requests. However, you seem to call that route only from BYE and CANCEL requests, missing out ACKs, which is likely the reason why things go wrong. So make sure you run all in-dialog requests through that route, and you should be fine (hopefully). Cheers, --Timo

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Kamailio Development Training, San Francisco Area, October 2011
by Daniel-Constantin Mierla 21 Sep '11

21 Sep '11

Hello, since I am attending GSoC'11 Mentors Summit and come to Bay Area again, it is time to bring the developer training in USA, therefore I am planning to organize a two-day seminar focused on developing Kamailio extensions in C. Among topics to be approached: - internal architecture - SIP parser - memory manager - locking manager - database API - config file language interpreter - RPC interface - pseudo-variables and transformations framework - module interface – write your own extensions in C as modules - inter-module APIs - tm, presence, a.s.o. - documentation docbook format This kind of training follows the pattern from the past, trying to keep a low attendee fee, targeting just to cover the expenses with the organization. I am discussing with several people for a good location at a convenient price, however, I am open for new alternatives, including companies/institutes that want to host the event at no cost in exchange of sending few attendees free of charge. Contact me if you can help here. Training is planned for October 24-25, right now the estimated fee per participant is 350USD (it will not exceed 400USD). The policy to accept people is first come, first served. You can register at this time to reserve your seat(s), emailing to me or to registration(a)kamailio.org. Number of seats is usually 10 to 20, last one organized in February in Barcelona filled up quickly. Once there will be enough interested people and a location selected, I will have more precise details regarding the participation to this seminar and I will send updates. Looking forward to meeting many of you and get new US developers on board of the project in the near future. Cheers, Daniel -- http://www.kamailio.org/w/daniel-constantin-mierla/

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how to remove media description from sdp?
by Juha Heinanen 21 Sep '11

21 Sep '11

trying to hack around the possible fix_nated_sdp("1") issue that i outlined in previous message, i thought to remove video media description from sdp. in sdpops module there is sdp_with_media(type), but for some reason no sdp_remove_media(type) function. is there some other means to get rid of sdp video media description? -- juha

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Digest authentication
by Morten Isaksen 20 Sep '11

20 Sep '11

Hi, Sorry for the slightly off topic question, but I am trying to debug a buggy PBX and want to check if the digest information is correct. I have this perl script. #!/usr/bin/perl use strict; use Digest::MD5 qw(md5 md5_hex md5_base64); my $uri = 'sip:sip1.uni-tel.dk'; my $method = "SIP/2.0"; my $username = "69102208"; my $password = "password"; my $nonce = "4e78d2b0a72196d9123be7c88d093bd74c77f35e"; my $realm = "sip1.uni-tel.dk"; my $a1 = "$username:$realm:$password"; my $ha1 = md5_hex($a1); my $a2 = "$method:$uri"; my $ha2 = md5_hex($a2); print "A1 : $a1\n"; print "A2 : $a2\n"; my $response = md5_hex("$ha1:$nonce:$ha2"); print "\nRES : $response\n\n"; I registered with a softphone and entered the above values from the Authentication header, but I cannot get the response to match. Any help would be appreciated. -- Morten Isaksen

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wt_timer, delayed replies and sendto error
by Andreas Granig 20 Sep '11

20 Sep '11

Hi, Another funny thing I encountered lately is this: Client sends INVITE via kamailio (via dispatcher) to sems, gets back 100/200, sends another couple of INVITE with same CSeq (looks like retransmissions), gets back more 200, then sends ACKs for each of the 200 it got. Looks like a quite broken client to me, but anyways. The interesting thing is that after >5000ms (which is bigger then tm.wt_timer), another 200 is sent from sems to kamailio to the client. I see the 200 coming to kamailio in a network trace, but not an according log message in the kamailio logs. What I see though is this: Sep 14 13:51:26 sp2 /usr/sbin/kamailio[15950]: ERROR: <core> [udp_server.c:586]: ERROR: udp_send: sendto(sock,0x97ec10,1055,0,1.2.3.4:58620,16): Invalid argument(22) Sep 14 13:51:26 sp2 /usr/sbin/kamailio[15950]: ERROR: <core> [forward.h:149]: msg_send: ERROR: udp_send failed The 1.2.3.4:58620 is obviously taken from the Vias, which looks like this: Via: SIP/2.0/UDP 127.0.0.1:5060;rport=5060;branch=xyz Via: SIP/2.0/UDP 10.1.1.9:58620;received=1.2.3.4;rport=58620;branch=abc The top-most Via is itself, the second is the one which reflects the destination address in the error message above. So I'm wondering whether there is some issue with finding the proper socket after wt_timer is expired (note that the leg client<->kamailio uses a different socket (with public ip) than kamailio<->sems (which uses 127.0.0.1)). Or any other idea where this error could come from? It's actually an educated guess that this could be related to wt_timer, but I don't know what else it could be. Andreas

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