sr-users March 2011

sr-users@lists.kamailio.org

83 participants
123 discussions

by Alexandre Abreu

Hello Carsten. Still no lucky. I just moved the whole rtpproxy/ directory into 3.1 branch and recompiled and then changed my script to use rtpproxy_offer()/rtpproxy_answer(). Look: Mar 15 14:04:38 devel rtpproxy[29111]: INFO:handle_command: new session e2361a59b4588d51, tag ed382a18;1 requested, type strong Mar 15 14:04:38 devel rtpproxy[29111]: INFO:handle_command: new session on a port 44206 created, tag ed382a18;1 Mar 15 14:04:38 devel rtpproxy[29111]: INFO:handle_command: pre-filling caller's address with 192.168.200.114:6380 Mar 15 14:04:39 devel kamailio[29544]: NOTICE: acc [acc.c:275]: ACC: transaction answered: timestamp=1300208679;method=INVITE;from_tag=ed382a18;to_tag=19477c42;call_id =e2361a59b4588d51;code=200;reason=OK; Mar 15 14:04:39 devel rtpproxy[29111]: INFO:handle_command: lookup on ports 44206/41680, session timer restarted Mar 15 14:04:39 devel rtpproxy[29111]: INFO:handle_command: setting custom timeout handler (xmlrpc:http://localhost:8000/RPC2) Mar 15 14:04:39 devel rtpproxy[29111]: INFO:handle_command: pre-filling callee's address with 192.168.200.149:9494 Mar 15 14:04:50 devel rtpproxy[29111]: INFO:process_rtp: session timeout Mar 15 14:04:50 devel rtpproxy[29111]: INFO:remove_session: RTP stats: 963 in from callee, 10 in from caller, 973 relayed, 0 dropped Mar 15 14:04:50 devel rtpproxy[29111]: INFO:remove_session: RTCP stats: 5 in from callee, 1 in from caller, 6 relayed, 0 dropped Mar 15 14:04:50 devel rtpproxy[29111]: INFO:remove_session: session on ports 44206/41680 is cleaned up Started with: rtpproxy -T 10 -F -i -l 192.168.200.90 -s udp:127.0.0.1:7722 -n tcp:192.168.200.90:8000 -d DBUG SIP-ROUTER config: # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722") modparam("rtpproxy", "rtpproxy_tout", 3) modparam("rtpproxy", "timeout_socket", "xmlrpc:http://localhost:8000/RPC2") # The Port for incoming XML-RPC requests modparam("mi_xmlrpc", "port", 8000) modparam("mi_xmlrpc", "reply_option", 1) modparam("mi_xmlrpc", "buffer_size", 8192) modparam("mi_xmlrpc", "log_file", "/tmp/openser_xmlrpc") The session keeps active after the RTPPROXY proxy timeout. It seems dlg_terminate_dlg isn't being triggered. Am I missing something else beyond moving rtpproxy/ directory? As this thread is a user-question, I am copying SR-USERS mail list. I am also using xmlrpc-c-1.06.41. Thanks, Alexandre. -----Mensagem original----- De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em nome de Carsten Bock Enviada em: terça-feira, 15 de março de 2011 05:15 Para: Alexandre Abreu Assunto: Re: RTPPROXY timeout patch. Hi Alexandre, sorry, yes, your correct. My changes are not in the 3.1 branch, but only GIT-Master. Only the GIT-Master currently holds the correct patch and config. I will need to backport the changes from the master... (will do so this week). Carsten 2011/3/14 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: > Hello Carsten, > > Let me ask you a question: > > If "First is important to clarify: from version 3.0.0 on, Kamailio and > SER are identical in terms of source code.", then why your changes > does not reflect in kamailio 3.1 from GIT? > I did see the changes by getting the SIP-ROUTER from GIT. Kamailio 3.1 > from GIT does not have your code. > > Thanks. > Alexandre > > -----Mensagem original----- > De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em > nome de Carsten Bock Enviada em: segunda-feira, 14 de março de 2011 > 05:31 > Para: Alexandre Abreu > Cc: RTPproxy Development > Assunto: Re: RTPPROXY timeout patch. > > Hello, > > i hope, you had a nice weekend... ;-) > Are you sure, you are using the GIT-Version of the > Kamailio/sip-router.org config? > In the GIT version it is now: > > modparam("rtpproxy", "timeout_socket", > "xmlrpc:http://localhost:8000/RPC2") > > http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=mod > ules/r > tpproxy/test/kamailio.cfg;h=0d5c73ff2065f33f262981fba2c07e65496e7415;h > b=HEAD > > This should work (as tested by me). > > Carsten > > 2011/3/11 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: >> Hello Carsten, >> >> The patch was applied just fine but I can't get that feature working. >> See >> below: >> >> [root@devel log]# rtpproxy -v >> Basic version: 20040107 >> Extension 20050322: Support for multiple RTP streams and MOH >> Extension >> 20060704: Support for extra parameter in the V command Extension >> 20071116: Support for RTP re-packetization Extension 20071218: >> Support for forking (copying) RTP stream Extension 20080403: Support >> for RTP statistics querying Extension 20081102: Support for setting >> codecs in the update/lookup command Extension 20081224: Support for >> session timeout notifications Extension 20090810: Support for >> automatic bridging Extension 20100819: Support for timeout >> notifications using XML-RPC towards Kamailio/sip-router.org >> >> Using SR 3.1.2, my configuration file was exactly the same as the >> example in test/ subdirectory you commit to rtpproxy module. >> >> # RTP-Proxy >> modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:22222") >> modparam("rtpproxy", "rtpproxy_tout", 3) modparam("rtpproxy", >> "timeout_socket", "http://localhost:8000/RPC2") >> >> # The Port for incoming XML-RPC requests modparam("mi_xmlrpc", >> "port", >> 8000) modparam("mi_xmlrpc", "reply_option", 1) modparam("mi_xmlrpc", >> "buffer_size", 8192) modparam("mi_xmlrpc", "log_file", >> "/tmp/openser_xmlrpc") >> >> Still inside test/ subdir we have the following example: >> >> [root@devel test]# cat exec_rtpproxy.txt ./rtpproxy -T 10 -f -F -i -l >> <your IP> -s udp:*:22222 -d DBUG >> >> You mean that if XML-RPC is used, it has to begin with "xmlrpc:". I >> can't see this string anywhere in these examples. I guess these files >> are not up to date. Am I correct? >> I can see the session timeout in RTPPROXY, but the session does not close. >> If I understand right how this feature works, it should send a >> "dlg_terminate_dlg + call-id" to send the BYE to both sides. >> >> Any tip on what could be wrong? >> >> Thanks, >> Alexandre >> >> -----Mensagem original----- >> De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] Em >> nome de Carsten Bock Enviada em: quinta-feira, 10 de fevereiro de >> 2011 >> 20:53 >> Para: Alexandre Abreu >> Cc: RTPproxy Development >> Assunto: Re: RTPPROXY timeout patch. >> >> Hi Alexandre, >> >> i updated the patch in the sip-router-repository (and attached the >> patch to this email). The new patch and the modifications to the >> rtpproxy module need some more testing, i have only done basic > functionality testing so far. >> I have reduced the patch a little, the socket, if the >> XML-RPC-Interface is used, has to begin with "xmlrpc:". This way, the >> protocol of the RTP-Proxy stays intact. >> If you have spare-time, please test it; otherwise, i will do it asap >> (but i am travelling a lot next week). >> >> @Maxim: This new version of the patch is much better, than the old one. >> Maybe you can review this one? Thanks! >> >> Carsten >> >> >> 2011/2/10 Carsten Bock <lists(a)bock.info>: >>> Hi Alexandre, >>> >>> the current GIT-version contains some more modifications regarding >>> timeout sockets. >>> That is the reason why the patch does not work anymore. >>> I will need to review the code and adapt my patch accordingly. >>> I will do so later. >>> >>> Carsten >>> >>> 2011/2/2 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: >>>> Hi Carsten, >>>> >>>> The stable version of RTPProxy is indeed 1.2.1. In this version, >>>> there is no file 'rtpp_notify.c' so I think the patch is really >>>> for git-version. Maybe the original patch was made regarding to an >>>> earlier >> git commit. >>>> >>>> Thanks, >>>> Alexandre >>>> >>>> -----Mensagem original----- >>>> De: kaiserbock2(a)googlemail.com [mailto:kaiserbock2@googlemail.com] >>>> Em nome de Carsten Bock Enviada em: quarta-feira, 2 de fevereiro de >>>> 2011 >>>> 06:02 >>>> Para: Alexandre Abreu >>>> Cc: RTPproxy Development >>>> Assunto: Re: RTPPROXY timeout patch. >>>> >>>> Hi Alexandre, >>>> >>>> the patch was for the stable version of RTP-Proxy (1.2.1?). I will >>>> review the patch for the git-version. >>>> I am a little busy at the moment (business travel), so i can first >>>> do this on friday or next week. >>>> >>>> Kind regards, >>>> Carsten >>>> >>>> P.S.: Please always CC the list. >>>> >>>> 2011/2/1 Alexandre Abreu <alexandre.abreu(a)redt.com.br>: >>>>> Hi Carsten, >>>>> >>>>> >>>>> >>>>> I am testing the new RTPProxy timeout feature in Kamailio (GIT > version). >>>>> >>>>> >>>>> >>>>> Got the rtpproxy from GIT: >>>>> >>>>> git clone git://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy >>>>> >>>>> >>>>> >>>>> I applied the patch: >>>>> >>>>> [root@devel rtpproxy]# patch < rtpproxy.patch >>>>> >>>>> patching file Makefile.am >>>>> >>>>> Hunk #1 FAILED at 4. >>>>> >>>>> 1 out of 1 hunk FAILED -- saving rejects to file Makefile.am.rej >>>>> >>>>> patching file config.h.in >>>>> >>>>> patching file configure.ac >>>>> >>>>> patching file rtpp_command.c >>>>> >>>>> Hunk #1 succeeded at 54 (offset 2 lines). >>>>> >>>>> Hunk #3 succeeded at 272 (offset 2 lines). >>>>> >>>>> Hunk #4 succeeded at 315 with fuzz 2. >>>>> >>>>> Hunk #5 FAILED at 466. >>>>> >>>>> Hunk #6 FAILED at 961. >>>>> >>>>> 2 out of 6 hunks FAILED -- saving rejects to file >>>>> rtpp_command.c.rej >>>>> >>>>> patching file rtpp_notify.c >>>>> >>>>> Hunk #1 FAILED at 38. >>>>> >>>>> Hunk #2 succeeded at 263 (offset 78 lines). >>>>> >>>>> misordered hunks! output would be garbled >>>>> >>>>> Hunk #4 FAILED at 250. >>>>> >>>>> Hunk #5 FAILED at 337. >>>>> >>>>> 3 out of 5 hunks FAILED -- saving rejects to file >>>>> rtpp_notify.c.rej >>>>> >>>>> patching file rtpp_notify.h >>>>> >>>>> Hunk #1 succeeded at 39 with fuzz 2 (offset 8 lines). >>>>> >>>>> patching file rtpp_session.c >>>>> >>>>> patching file rtpp_session.h >>>>> >>>>> >>>>> >>>>> As you see, not so clean but the files were patched. >>>>> >>>>> >>>>> >>>>> On CentOS 5, I cant find the LIBXMLRPC through YUM/RPM, so I >>>>> compiled xmlrpc-c-1.06.41.tgz by hand: >>>>> >>>>> >>>>> >>>>> ./configure prefix=/usr && make && make install >>>>> >>>>> AND ./configure && make && make install >>>>> >>>>> With that, the headers are both on /usr/lib/include and >>>>> /usr/local/lib/include (I did that just to be sure that the config >>>>> wasnt looking at the wrong place). >>>>> >>>>> >>>>> >>>>> xmlrpc_abyss.h >>>>> >>>>> xmlrpc_cgi.h >>>>> >>>>> XmlRpcCpp.h >>>>> >>>>> xmlrpc_server.h >>>>> >>>>> xmlrpc_client.h >>>>> >>>>> xmlrpc.h >>>>> >>>>> xmlrpc_server_w32httpsys.h >>>>> >>>>> >>>>> >>>>> The configure script from patched RTPPROXY doesnt _try_ to find >>>>> the LIBXMLRPC (theres no reference in config.log) and therefore >>>>> my rtpproxy doesnt have the support for the timeout notification. >>>>> >>>>> >>>>> >>>>> Any comment on what could be wrong? >>>>> >>>>> >>>>> >>>>> Alexandre >>>> >>>> >>>> >>>> -- >>>> Carsten Bock >>>> Schomburgstr. 80 >>>> 22767 Hamburg >>>> Germany >>>> >>>> Mobile +49 179 2021244 >>>> Home +49 40 34927217 >>>> Büro (Verl) +49 5246 801427 >>>> Fax +49 40 34927218 >>>> mailto:carsten@bock.info >>>> >>>> >>> >>> >>> >>> -- >>> Carsten Bock >>> Schomburgstr. 80 >>> 22767 Hamburg >>> Germany >>> >>> Mobile +49 179 2021244 >>> Home +49 40 34927217 >>> Büro (Verl) +49 5246 801427 >>> Fax +49 40 34927218 >>> mailto:carsten@bock.info >>> >> >> >> >> -- >> Carsten Bock >> Schomburgstr. 80 >> 22767 Hamburg >> Germany >> >> Mobile +49 179 2021244 >> Home +49 40 34927217 >> Büro (Verl) +49 5246 801427 >> Fax +49 40 34927218 >> mailto:carsten@bock.info >> >> > > > > -- > Carsten Bock > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com

13 years, 1 month

Release for SPCE version 2.2

by Deon Vermeulen

Hi Guys When can we expect the release of spce version 2.2? Kind Regards Deon Vermeulen

13 years, 1 month

Kamailio - Asterisk CDR

by Lucas Alvarez

Hi, I have an integration kamailio-asterisk. I have many calls that don't pass through asterisk, kamailio route that calls to other media gateways. I would like to log those calls in asterisk CDR, is that possible? Could someone give me any direction? Thanks in advance. Lucas

13 years, 1 month

Problem on multiple interfaces and conversion between UDP and TCP

by David

Hello, I have a Kamailio 3.0 server with two interfaces. 192.168.x.x and a public interface. I send PUBLISH out on the private interface to my presence server. Everything else happens on the public address. I did not use the "listen=" option in the config, so Kamailio was automatically detecting my interfaces to bind to. Last night I had a provider sending packets in on TCP and Kamailio was sending out on UDP. Unfortunately, it kept using the private interface despite that the IP was unreachable on this interface. It looks like Kamailio uses the first interface it finds when converting between TCP and UDP, and I believe that on the server, the private interface was being listed first. I finally added listen= in kamailio with all 4 interfaces ( tcp/udp public and tcp/udp private ). I listed the public UDP interface first, restarted and the packets started going out on the public interface. Is this a feature or a bug ? How does Kamailio decide which interface to use ? What if I sent a packet to an IP that is only reachable on one interface? At the very least, I think that a note about this difficulty should be included in the default config so that future users don't have the same issue I did. Thanks, David

13 years, 1 month

getting fragmented packet when adding record-route header

by Asgaroth

Hi All, I have a scenario where I have 2 asterisk media servers that, when calling a registered sip account, will forward off the invite to a location server, who looks up the contact information, then forwards the invite off to the proxy that the user registered against (done via the path module). This all appears to work perfectly until the us hangs up the call. The BYE message does not make it all the way back to the originating asterisk media server, it makes it back to the location server that looked up the registration information. To overcome this issue, I add a "Record-Route" header specifying the address of the asterisk media server that the request originated from. However, when I add the custom header, the Invite shows up as fragmented packet my wire-shark trace at the end-user's soft-phone. The call sets up properly and when I hang up it looks like the BYE message makes it all the way back to the media server, but when it is relayed on the the caller, the sip proxy then fails the call with a message too big. If I remove the line that adds the record-route header, then all looks fine, except that I am unable to tear down calls from sip end-points. The specific line I'm using to add the header is as follows: insert_hf("Record-Route: <sip:$si:$sp;lr=on>\r\n","Record-Route"); I do remove this from the route header on in-dialog replies as loose-route does not remove the header on the original location server as it does not see the route header as local, so I manually remove it. Has anyone come across this particular issue? I'm using Kamailio 3.1.2. Am I going about this is the wrong way? I've been reading the module docs for registrar/usrloc/rr/path/textops to see if there is something I've missed but cannot see it. Any tips/suggestions would be greatly appreciated. Thanks

13 years, 1 month

ACK not sent and rr-enforced

by Dominguez Jover, Ricardo

Hi everybody. I am using Kamailio 3.1 and RTP proxy for internet calls. I'm using external test accounts to check if the calls are established and the media flow is ok. When I use a sip2sip.info or VoIP Talk accounts, then all is working fine between my internal and these external accounts. But when I use a iptel.org account and this account calls to an internal account (registered with kamailio), then callee sends the 200 OK to the SIP proxy and the SIP proxy to iptel. IPtel.org proxy sends the ACK to my proxy with this lines at the end of the packet: P-hint: rr-enforced\r\n P-hint: rr-enforced\r\n And my SIP proxy never resends the ACK to the callee, so the callee resends OK 200 periodically and after 32 seconds sends a BYE message and the call is finished. I've been reading posts about missing ACKs but I can't find the answer to my problem, that it seems like "t_check_trans" doesn´t recognize the ACK as related to a transaction. But this is only with IPTEL accounts, my proxy SIP is working with other SIP providers, so I don't know if forcing relay of every ACK packet is a good idea. Any help would be appreciated. Thanks, Ricardo

13 years, 1 month

Installation Problem: : /usr/local/sbin/kamailio: not found

by Derrick Ding

Hi, All Yesterday I install Kamailio(3.1.x) by myself under a new Ubuntu server 10.04 environment. I got it from http://www.kamailio.org/pub/kamailio/3.1.2/bin/ kamailio-3.1.2_linux_i386.tar.gz <http://www.kamailio.org/pub/kamailio/3.1.2/bin/kamailio-3.1.2_linux_i38 6.tar.gz> I put all stuff under corresponding directory: /usr/local/sbin with bin file /usr/local/etc/kamailio with .cfg file /usr/local/lib/kamailio with lib file Based on http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-gi t I create /etc/init.d/kamailio PATH=/sbin:/bin:/usr/sbin:/usr/bin DAEMON=/usr/local/sbin/kamailio NAME=kamailio DESC=kamailio HOMEDIR=/var/run/kamailio PIDFILE=$HOMEDIR/$NAME.pid DEFAULTS=/etc/default/kamailio RUN_KAMAILIO=no Also I create /etc/default/ kamailio I find a problem after when I run /etc/init.d/kamailio start, it says: root@aastra:/etc/init.d# /etc/init.d/kamailio start Not starting kamailio: invalid configuration file! -e /etc/init.d/kamailio: 168: /usr/local/sbin/kamailio: not found But the file is there: root@aastra:/etc/init.d# ls -l /usr/local/sbin/kamailio -rwxr-xr-x 1 root root 4103257 2010-07-01 12:16 /usr/local/sbin/kamailio root@aastra:/etc/init.d# /usr/local/sbin/kamailio bash: /usr/local/sbin/kamailio: No such file or directory I wonder if this /usr/local/sbin/kamailio is not for Ubuntu server 10.04 and cannot be run Thanks for help Derrick

13 years, 1 month

major bug in auth module?

by Juha Heinanen

one sip ua sent this kind of authorization header: Authorization: Digest username="foo",realm="bar",uri="xxx",response="e439fa7438452da7d50690f2268be16f",nonce="4d8060d2000041b2430e64fe7ec681d8fb5709e9c72b8a01",qop=auth,cnonce="0004ccb0",nc=000060a2. and sr auth module didn't like it but emitted error message FIXME:check_nonce: bad nc value. i went and checked the code and found this: if (nc_enabled && (pf & NF_VALID_NC_ID) && auth->digest.nc.s && auth->digest.nc.len){ if (str2int(&auth->digest.nc, &nc)!=0){ /* error, bad nc */ ERR("FIXME:check_nonce: bad nc value %.*s\n", auth->digest.nc.len, auth->digest.nc.s); return 5; /* invalid nc */ } i.e., looks like it is using str2int to convert to nc value to int. the problem is that str2int only handles decimal strings, whereas nc value is a hex string. am i dreaming or what? -- juha

13 years, 1 month

LCR Routes and memory allocations

by Graham Wooden

Hi there, I have now approximately 400,000 routes defined in my LCR table and I would like to make sure that I have enough memory allocated for it. I read somewhere (a non official Kamailio site) about PKG_MEM_POOL_SIZE and SHM_MEM_SIZE needing to be altered to handle the increased routes. System is dedicated for the Kamailio 1.5.5 install and has 4GB of RAM. Thoughts to what I need to change these values (or anything else) to accommodate? Thanks, -graham

13 years, 1 month

Error: compile kamailio with STUN support

by Derrick Ding

Hi, All I met same problem when compile kamailio with STUN support. http://www.voipuser.org/forum_topic_18741.html /usr/include/bits/string3.h:83: warning: memset used with constant zero length parameter; this could be due to transposed parameters ser_stun.o: In function `add_fingerprint': /root/kamailio-3.0.1/ser_stun.c:890: undefined reference to `SHA1' ser_stun.o: In function `validate_fingerprint': /root/kamailio-3.0.1/ser_stun.c:973: undefined reference to `SHA1' collect2: ld returned 1 exit status make: *** [kamailio] Error 1 I search online that need link libssl -lssl But I don't know where to add this compile flag. The Makefile of this project is too complicated for me to understand. Any one can tell me how to I can fix this? By the way, I have install libssl. It's /usr/lib/libssl.so.0.9.8 Sincerely, Derrick

13 years, 1 month

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sr-users March 2011