sr-users August 2010

sr-users@lists.kamailio.org

47 participants
71 discussions

Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy

by Daniel-Constantin Mierla

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/

13 years, 6 months

Evaluate Perl Return Value in Routing Logic

by "Nicolas Rüger"

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

13 years, 6 months

Error when integrate xmpp module to Kamailio 3.0.1

by Quốc Huy Nguyễn

Hi all, I have some problem when integrate module xmpp to Kamailio 3.0.1. I have already searched on google and mailing-list but can't find the right way to fix it. I install Kamailio 3.0.1 on server 192.168.1.25 (kamailio1.com), Ejabberd on server 192.168.1.21 (ejabberd.com). Here the config of kamailio1 server: # ---- xmpp ---- modparam("xmpp", "gateway_domain", "kamailio1.com") modparam("xmpp", "xmpp_domain", "ejabberd.com") modparam("xmpp", "xmpp_host", "ejabberd.com") modparam("xmpp", "backend", "component") modparam("xmpp", "xmpp_password", "secret") modparam("xmpp", "domain_separator", "*") modparam("xmpp", "xmpp_port", 5347) ........ if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404", "Not found"); } return; } log("*** xmpp: unhandled message type\n"); sl_send_reply("503", "Service unavailable"); return; Here the config of ejabberd server: {5347, ejabberd_service, [ {access, all}, {shaper_rule, fast}, {ip, {192, 168, 1, 21}}, {hosts, ["kamailio1.com"], [{password, "secret"}]} ]}, I create 1 user on kamailio1 server: 103(a)kamailio1.com And 1 user on ejabberd.com: 102(a)ejabberd.com Test: - Login with account 103(a)kamailio1.com on sip communicator. Add 102*ejabberd.com(a)kamailio1.com on the sip contact list. - Login with account 102(a)ejabberd.com on pidgin. Add 103*kamailio1.com(a)ejabberd.com on the buddy list. - When 103(a)kamailio1.com chat to 102*ejabbed.com(a)kamailio1.com, kamailio1 can't send MESSAGE to ejabberd.com, and here the log on kamailio1 server: /usr/local/sbin/kamailio[4470]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4463]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4468]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4470]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4468]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4468]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4470]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4462]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff I think I had some wrong or miss something in the configuration of kamailio and ejabberd, please help me to check it. P/S: I also sent the sip trace on the attached file. In the sip trace file, the ip of sip client is 192.168.1.65, and the ip of pidgin client is 192.168.1.11 Thanks and Best Regards, Huy Nguyen Nguyễn Quốc Huy Post & Telecommunications Institute and Technology branch Ho Chi Minh City -------------------------------------------------- Email: huy_quocnguyen(a)live.com Mobile: +84 906755226 Y!M: huy_quocnguyen1987

13 years, 7 months

time format in db_flatstore

by SIP Projects

Using version 1.5.2. Trying to migrate acc from db_mysql to db_flatstore to reduce io wait during heavy dialing. However, the default time format changes from normal YYYY-MM-DD hh:mm:ss to unixtime in flatstore. I tried using $TF but its format is even more non-standard. Is there a way to change time format for either the default acc time field in the db_flatstore module or $TF? Thanks, Matt

13 years, 7 months

some issues in Kamailio WITH FreeSWITCH, SOS

by Abhishek [ABK] Kumar

Hi SiP-Users, I got kamailio installed and configured to work using mysql with FreeSWiTCH using this HowTo wiki freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms [Asipto - SIP and VoIP Knowledge Base Site]<http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms>: http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms This is also reference on FreeSWITCH portal and was the only decent tutorial I was able to find on integrating Kamailio with FreeSWITCH. Followed everything successfully, but I have a slight confusion/problem in its wonderful working... >From all tests I performed, I perceived following activities: a.) it handled the extensions added to it very well, and carried over the call to destination extension b.) if destination extension is also added to it, call established without intervention of FreeSWITCH c.) if extension is not added to it but FreeSWITCH, the call reached FreeSWITCH but was rejected there d.) Red5Phone (developed over RED5 Server, @Google-Code) could dial with success but didn't received any Now, *for point (a)* Thank You! *for point (b)* it should be transferred to FreeSWITCH and that should be allowed to establish call, otherwise all dial-plans,etc. goes waste *for point (c)* may be some settings in Wiki are Obsolete OR New, could you please point it out *for point (d)* it may be due to point (b) -- -- Regards, Abhishek Kumar https://sites.google.com/site/abhikumar163/ -- --------------ABK-----mail.signature-------------------- <http://www.blogger.com/profile/06276198262605731980><http://abhishekkr.deviantart.com/><http://www.facebook.com/aBionic><http://www.twitter.com/aBionic><http://sourceforge.net/users/abhishekkr><http://www.youtube.com/user/1ABK><http://in.linkedin.com/in/abionic> ----------------------------------------------------------- ~=ABK=~

13 years, 7 months

kamailio and asterisk proiblem

by chuks＠cybernergies.com

13 years, 7 months

failure_route send 500 retry later

by Alejandro Mellado G.

Hi' I'm trying to forward the call to voicemail on sems when the time of response is out. When the user isn't in location, the forward to voicemail work very fine ( $rc = -1 ). But failure_route, doesn't work and send message "500 Retry Later". I'm using kamailio 3.0.2 and I'm probing with: modparam("tm", "fr_timer", 10) modparam("tm", "fr_inv_timer", 15) ... ... failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset("6") || isflagset(5))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { revert_uri(); avp_db_load("$ruri","$avp(s:email_address)/$email_scheme"); append_hf("P-App-Name: myvoicemail\r\n"); append_hf("P-App-Param: Email-Address=$avp(s:email_address)\r\n"); rewritehostport("sems_host:5080"); append_branch(); t_relay(); } } Thanks. -- Saluda Atte, Alejandro Mauricio Mellado Gatica Escuela de Ingeniería Informática Universidad Católica de Temuco _________________________________

13 years, 7 months

help to use force_rtp_proxy([flags [, ip_address]]).

by peter_green lion

hi all, i am a new user kamailio. i have configure kamailio with RTP proxy, but i have a problem in using : force_rtp_proxy("c","192.168.1.10")because i want to change to force to this ip. my configure is : when server receive "200 OK" it change value in "c= IN IP4 <ip rtp server>" to "c = IN IP4 192.168.1.10" i configure as : kamailio.cfg : #!ifdef WITH_NAT if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy("c","192.168.1.10"); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } when i make call from sip a to sip b, sip b answer .trace as : U 115.78.129.190:54337 -> <server ip>:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP <server ip>;branch=z9hG4bK00c.daa8fe8.0. Via: SIP/2.0/UDP 192.168.1.10:50937;received=115.78.129.190;branch=z9hG4bK-d8754z-1f66a816d651d504-1---d8754z-;rport=63930. Record-Route: <sip:<server ip>;lr;nat=yes>. Contact: <sip:102@192.168.1.10:8576;rinstance=8392ffb3fe461110>. To: <sip:102@<server ip>:5060>;tag=d179a842. From: <sip:101@<server ip>:5060>;tag=1c61b708. Call-ID: ZTQzYjZkYzFjODE1MWFlNjIwNmQ2ZGU5MWUxYmM1NjQ.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: PortGo v6.0, Build 07282010. Content-Length: 237. . v=0. o=- 30452887 30452887 IN IP4 169.254.202.160. s=http://www.portsip.com. c=IN IP4 169.254.202.160. t=0 0. m=audio 21480 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U <server ip>:5060 -> 115.78.129.190:63930 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.10:50937;received=115.78.129.190;branch=z9hG4bK-d8754z-1f66a816d651d504-1---d8754z-;rport=63930. Record-Route: <sip:server ip;lr;nat=yes>. Contact: <sip:102@115.78.129.190:54337;rinstance=8392ffb3fe461110>. To: <sip:102@server ip>:5060>;tag=d179a842. From: <sip:101@<server ip>:5060>;tag=1c61b708. Call-ID: ZTQzYjZkYzFjODE1MWFlNjIwNmQ2ZGU5MWUxYmM1NjQ.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: PortGo v6.0, Build 07282010. Content-Length: 237. . v=0. o=- 30452887 30452887 IN IP4 169.254.202.160. s=http://www.portsip.com. c=IN IP4 169.254.202.160. t=0 0. m=audio 21480 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. thanks for help me. regards. beter_green

13 years, 7 months

ratelimit module migration from (k) to generic

by Ovidiu Sas

Hello all, The (k) ratelimit module is an enhanced version of the original (s) ratelimit module. Are there any objections in moving the ratelimit module from (k) to generic and purging the (s) version? Regards, Ovidiu Sas

13 years, 7 months

help to get ip from contact header in INVITE sip message.

by peter_green lion

hi all, please help me to get ip add from Contact header in INVITE message. i see that : ct =<sip:101@192.168.1.10:63027;rinstance=85f97591c218086a> have the ip or domain ("192.168.1.10"). so please suggest to get this ip. thanks so much. beter green

13 years, 7 months

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sr-users August 2010