sr-users January 2013

sr-users@lists.kamailio.org

82 participants
126 discussions

by Klaus Darilion

Hi! IMO the siremis CDR generation procedure is buggy: CREATE PROCEDURE `kamailio_cdrs`() DECLARE inv_cursor CURSOR FOR SELECT src_user, src_domain, dst_user, dst_domain, dst_ouser, time, callid,from_tag, to_tag, src_ip FROM acc where method='INVITE' and cdr_id='0'; ... REPEAT FETCH inv_cursor INTO v_src_user, v_src_domain, v_dst_user, v_dst_domain, v_dst_ouser, v_inv_time, v_callid, v_from_tag, v_to_tag, v_src_ip; IF NOT done THEN SET bye_record = 0; SELECT 1, time INTO bye_record, v_bye_time FROM acc WHERE method='BYE' AND callid=v_callid AND ((from_tag=v_from_tag AND to_tag=v_to_tag) OR (from_tag=v_to_tag AND to_tag=v_from_tag)) ORDER BY time ASC LIMIT 1; IF bye_record = 1 THEN INSERT INTO cdrs (src_username,src_domain,dst_username, dst_domain,dst_ousername,call_start_time, duration,sip_call_id, sip_from_tag,sip_to_tag,src_ip,created) VALUES (v_src_user, v_src_domain,v_dst_user, v_dst_domain,v_dst_ouser,v_inv_time, UNIX_TIMESTAMP(v_bye_time)-UNIX_TIMESTAMP(v_inv_time), v_callid,v_from_tag,v_to_tag,v_src_ip,NOW()); UPDATE acc SET cdr_id=last_insert_id() WHERE callid=v_callid AND from_tag=v_from_tag AND to_tag=v_to_tag; END IF; SET done = 0; END IF; UNTIL done END REPEAT; ... 1. The UPDATE query will not UPDATE BYE records which were sent from CALLEE to CALLER. This can be easily fixed with: UPDATE acc SET cdr_id=last_insert_id() WHERE callid=v_callid AND ( (from_tag=v_from_tag AND to_tag=v_to_tag) OR (to_tag=v_from_tag AND from_tag=v_to_tag) ); 2. (minor) It does not consider reINVITEs. I know the default Kamailio config does not account reINVITEs, but I think when reINVITEs are accounted too, this can mess up the CDR generation. regards Klaus

11 years, 8 months

Kamailio 1.5 and Asterisk 1.8 issue (Protocol Error, Unspecified)

by Rowie

Hi, We are having an issue where a phone (snom in particular) cannot make a call through Asterisk. It just hangup and does not allow the call to go through. I am including a a sip trace on this thread to show what is happening within the call. Please see below: Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 -- View this message in context: http://old.nabble.com/Kamailio-1.5-and-Asterisk-1.8-issue-%28Protocol-Error… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

11 years, 9 months

Kamailio and Ubuntu Upstart script

by Moacir Ferreira

As I posted here before, Ubuntu has moved to Upstart as the main engine to start daemons. As a result, Kamailio will fail to start on boot time because it starts before MySQL that has already been moved to the new Upstart daemon starting fashion. I am struggling to get some help from the Ubuntu community explaining how we can match the previous init script behavior when using Upstart but I got no success till now. Anyway, I have created an Upstart script for Kamailio that does most everything that the old script use to do. However, in this script I am making sure MySQL starts before Kamailio. If you have another SQL engine you must change the script to match your SQL engine by changing the “start on started” statement on the script. To use it you have to: A – Create a kamailio.conf file at “/etc/init/” with the contents of the kamailio.conf script in attach, that is the new Upstart stile init script. Please notice that I just adapted the existing script, no news… B – Create a symbolic link at /etc/init.d/ pointing to this kamailio.conf at /etc/init/. C – If you want, you can still have a /etc/default/kamailio file like it used to be and the script will behave like it used to behave when using the default file. Hopefully someone more “influent” at the Ubuntu community may get some help on how to adapt their “start on started” Upstart stanza to behave like we need for multiple possible SQL engines. Cheers! Moacir

11 years, 9 months

How to rewrite "tu" in CANCEL message ?

by Julia

Hello all, Is there any way to rewrite "tu" in CANCEL message in 3.2.1? We use remove_hf / insert_hf for INVITE's "tu" modification before t_relay. It works fine, but it doesn't work for CANCEL (INVITE re-parsing is disabled). Ths,

11 years, 9 months

pseudovariable values written from app_lua sometimes appear incorrect

by elactrum＠jamailca.com

Hi, I am using app_lua to frequently store some data in htable and avp entries, using sr.pv.sets. When I read the values using sr.pv.get immediately after writing them, they seem correct. However, when I read them later from Lua, I have noticed that sometimes the value is incorrect or nil. Am I doing something wrong? The following code uses rtimer to duplicate the problem on startup, printing the errors to syslog. Tested on version: kamailio 3.3.1 (i386/linux) 2d0f6f ############### # kamailio.cfg ############### loadmodule "pv.so" loadmodule "xlog.so" loadmodule "htable.so" loadmodule "rtimer.so" loadmodule "app_lua" modparam("htable", "htable", "status=>size=14;autoexpire=0;") modparam("rtimer", "timer", "name=ta;interval=10;mode=1") modparam("rtimer", "exec", "timer=ta;route=LUA_TEST") # the following path may need to change depending on filesystem setup modparam("app_lua", "load", "/etc/kamailio/kamailio.lua") route { route(LUA_TEST); } route[LUA_TEST] { $var(lua_test_key) = $(pp{s.toupper}) + "::"; lua_runstring("htable_test([[status=>$var(lua_test_key)]])"); lua_runstring("avp_test([[status=>$var(lua_test_key)]])"); } ---------------- -- kamailio.lua ---------------- -- Test of sr.pv operations in Lua function htable_test (namekey) local pvar_prefix = "$sht("..namekey pvar_test(pvar_prefix) end function avp_test ( name ) local pvar_prefix = "$avp("..name pvar_test(pvar_prefix) end function pvar_test ( pvar_prefix ) for value = 10000, 10050 do local pvar = pvar_prefix..value..")" --sr.log("info", "Lua attempting to set '"..pvar.."' to value '"..value.."'") sr.pv.sets(pvar, value) end for value = 10000, 10050 do -- now we check what was stored local pvar = pvar_prefix..value..")" local checkvar = sr.pv.get(pvar) if ( tostring(checkvar) ~= tostring(value) ) then sr.err(string.format("CHECK: Lua checking pvar '%s' and got value '%s' which is NOT '%s'", tostring(pvar), tostring(checkvar), tostring(value) )) end end end -- end of kamailio.lua Thanks, -Andrew

11 years, 9 months

Kamailio multiple registration issue

by Ramaseshi reddy kolli

Hi, i am using kamailio 3.3 version please help me with this issue. 1) Registered from SIP end point, kamailio added row in location table 2) Now using same SIP account details, pushed registration entry using *kamctl ul add , * instead of overwriting the previous contact by new one it inserts another new row in the location table 3) Total now 2 rows with same username are present in DB. I am using save("location", "0x04") so before inserting new row it should check whether already entry is there or not, if there should update the same row. I look forward to your reply. Thanks Ram

11 years, 9 months

Kamailio-Java Integration

by ron.kamailio＠mcleodnet.com

I am looking for a way to integrate Java business logic with Kamailio to influence routing decisions, add/remove/modify headers, record accounting-related information, etc. A while back there was a project named WeSIP, but it doesn't appear to active anymore. Is anyone doing something like this now, or any ideas or suggestions? Thanks, Ron

11 years, 9 months

Re: [SR-Users] sr-users Digest, Vol 92, Issue 50:Modify IP address in SDP (Austin Einter)

by madhumanjusha＠integramicro.com

On Tue, January 22, 2013 1:58 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. Modify IP address in SDP (Austin Einter) > 2. Re: Modify IP address in SDP (Olle E. Johansson) > 3. Problem with 4.0 nightlies and IMS - sem_post undefined > (Barry Flanagan) > 4. Patch to resolve segfault in json module (Richard Brady) > 5. Re: Modify IP address in SDP (Austin Einter) > 6. Re: Patch to resolve segfault in json module > (Daniel-Constantin Mierla) > 7. Re: [sr-dev] FOSDEM (Daniel Pocock) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Tue, 22 Jan 2013 18:37:13 +0530 > From: Austin Einter <austin.einter(a)gmail.com> > Subject: [SR-Users] Modify IP address in SDP > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CANXt1k--VtC6aYHGTBZ5GqF8n=66g5PVwG4YhOC_uefxpVYy=g(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > > Dear All > When 200 OK to INVITE passes through Kamailio proxy , I want to change the > IP address in SDP to a specific IP address. > > > Can this be done by just modifying configuration files or do I need to > modify in source code. > > In case I need to modify source code, which files I need to look at. > > > Kindly provide pointers for said problem. > > > Thanks > Austin Hello Austin, 1...I was doing similar one for my task. Files : /root/kamailio-3.3.2/parser/sdp/sdp.h sdp.h file has struct that holds ip address and port numbers of audio and video. Stucture: sdp_session_cell struct has member named str ip_addr will give you ip address in sdp. 2...Yes you can edit existing code or else you can create your own 200ok packet.I created my own 200OK and invite packets with sdp. 3...You can grab below ip address when kamailio proxy is about to forward 200OK to end client.File where you can find 200OK is (though not sure but you can try this file edit existing 200OK): /root/kamailio-3.3.2/modules/tm/t_reply.c has below function static int _reply( struct cell *trans, struct sip_msg* p_msg, unsigned int code, char * text, int lock ) { } 4...I did the following way to get ipaddr...and below piece of code you can use it in the above function's body to edit 200OK's ip address:: char my_src_rtp_addr[8]; int my_src_rtp_addr_len=0; struct sdp_session_cell *sdp_session; if(0 != parse_sdp(p_msg)) { LOG(L_INFO,"Unable to parse sdp\n"); } sdp_session=get_sdp_session(p_msg,0); if(sdp_session!=0){ my_src_rtp_addr_len=strlen(my_src_rtp_addr); my_src_rtp_addr[my_src_rtp_addr_len]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+1]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+2]='\0'; my_src_rtp_addr[my_src_rtp_addr_len+3]='\0'; strncpy(my_src_rtp_addr,sdp_session->ip_addr.s,sdp_session->ip_addr.len); } If any questions,please let me know. Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

11 years, 9 months

msilo patch

by Stoyan Mihaylov

When I used, last versions from git, I got error and crash of kamailo using m_dump. I got latest version from git, and I made one patch: diff --git a/modules/msilo/msilo.c b/modules/msilo/msilo.c index 209b309..4bda8fa 100644 --- a/modules/msilo/msilo.c +++ b/modules/msilo/msilo.c @@ -1125,8 +1125,16 @@ static int m_dump(struct sip_msg* msg, str* owner_s) memcpy(tmp_extra_hdrs.s, extra_hdrs_str.s, extra_hdrs_str.len); memcpy(tmp_extra_hdrs.s+extra_hdrs_str.len, str_vals[4].s, str_vals[4].len); } else { - tmp_extra_hdrs.len = 0; - tmp_extra_hdrs.s = ""; + tmp_extra_hdrs.len = 1; + if ((tmp_extra_hdrs.s = pkg_malloc(1)) == NULL) + { + LM_ERR("Out of pkg memory"); + if (msilo_dbf.free_result(db_con, db_res) < 0) + LM_ERR("failed to free the query result\n"); + msg_list_set_flag(ml, mid, MS_MSG_ERRO); + goto error; + } + tmp_extra_hdrs.s[0]='\x0'; } hdr_str.len = 1024; if(m_build_headers(&hdr_str, str_vals[3] /*ctype*/, The problem is: tmp_extra_hdrs.s = ""; and consequent pkg_free(tmp_extra_hdrs.s); As I see, there are couple of ways to solve problem, one of them is setting tmp_extra_hdrs.s = 0; but then we have warnings, or setting tmp_extra_hdrs.s = 0; and check before pkg_free, but pkg_free is used on couple of places. I am not sure what is the best way to report bugs and offer patches.

11 years, 9 months

Best way to 'detect' direction

by Paul Belanger

Greetings, I'm in the process of adding logic to 'detect' the direction of my message, I have to set: rtpproxy_manage("ei") or rtpproxy_manage("ie") respectively. Rather then hard-coding IP addresses in my configuration, I was curious how other people over come this? Perhaps tagging their call control in some other method? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

11 years, 9 months

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sr-users January 2013