sr-users April 2010

sr-users@lists.kamailio.org

65 participants
104 discussions

[Serusers] Error while running icscf.sh
by Abdul Khadar 28 Jul '10

28 Jul '10

Hi, Can you please help me in resolving this issue while setting OpenIMSCore setup. Error : [root@hss OpenIMSCore]# sh icscf.sh * 0(18935) ERROR: load_module: could not open module </opt/OpenIMSCore/ser_ims/modules/mysql/mysql.so>: /opt/OpenIMSCore/ser_ims/modules/mysql/mysql.so: undefined symbol: log * 0(18935) parse error (30,13-14): failed to load module ERROR: bad config file (1 errors) 0(18935) INFO:cdp:cdp_exit(): CDiameterPeer stoping ... icscf.sh: line 5: 18935 Segmentation fault /opt/OpenIMSCore/ser_ims/ser -f /opt/OpenIMSCore/icscf.cfg -D -D please help me to resolve this problem. -- thanks & regards, Abdul Khadar

4 4

t_on_failure to named failure route
by Steven C. Blair 18 Jun '10

18 Jun '10

I'm attempting to run an existing SER config file under a recent build of sip-router. I get a syntax error on the t_on_failure("noroute") call that exists in my config file. The specific error is "bad expression: type mismatch (str instead of int)". I presume this means sip-router does not support named failure route labels. Is that correct? Thanks,Steve --- ISC Networking & Telecommunications 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 215-573-8396 215-898-9348 (fax)

6 10

Wrong handling CANCEL message
by pars3c 08 May '10

08 May '10

Hi, i have a problem about the handling of the "cancel" message. The call flow is this: A -----------------> (Invite) Proxy (P) B (100 Tryng) <--------------------------- ----------------->(Invite) B (100 Tryng) <--------------------------- (183 Progress) <--------------------------- (183 Progress) <--------------------------- (200 OK) <--------------------------- (200 OK) <--------------------------- --------------------------------->(CANCEL) (200 canceling) <--------------------------- (200 OK) <--------------------------- (200 OK) <--------------------------- ----------------->(ACK) ----------------->(ACK) (BYE) <--------------------------- (BYE) <--------------------------- ----------------->(481 Call Leg/Transaction Does Not Exist) ----------------->(481 Call Leg/Transaction Does Not Exist) The B side answer with OK, after a while , a send a CANCEL. I don't know why Kamailio don't forward this message to the B side. B retry to send the OK message, then A send the ACK. At the end , B send BYE , but A don't have the transactin. In this situation, kamailio should deliver the "CANCEL" to the B side ? (even if, before B send the OK ) Or the proxy should not consider the CANCEL because B has answered with "OK" ? Where is the error on my configuration ? Thanks I have Kamailio 1.5.3 and my configuration is this: modparam("siptrace", "trace_on", 0) modparam("carrierroute", "config_source", "db") modparam("auth_db", "password_column", "ha1") modparam("auth_db", "use_domain", 1) modparam("usrloc", "db_mode", 2) #modparam("usrloc","nat_bflag", 8) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "db_mode", 1) modparam("permissions", "peer_tag_avp", "$avp(i:707)") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("avpops","avp_table","usr_preferences") modparam("avpops","attribute_column","attribute") modparam("avpops","value_column","value") modparam("uac","from_restore_mode","auto") modparam("tm", "fr_timer", 12) modparam("tm", "wt_timer", 32) modparam("rr", "append_fromtag", 1) #modparam("rr", "enable_full_lr", 1) modparam("userblacklist", "use_domain", 0) #modparam("dialog", "dlg_flag", 2) modparam("registrar|nathelper", "received_avp", "$avp(i:42)") #modparam("registrar", "min_expires", 17000) modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "detect_direction", 1) route{ # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Preprocessing # ----------------------------------------------------------------- if (!method=="REGISTER") record_route(); # ----------------------------------------------------------------- # NAT Detection # ----------------------------------------------------------------- force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(7); } # ----------------------------------------------------------------- # # ----------------------------------------------------------------- sip_trace(); ##Loose_route packets if (has_totag()) { if (loose_route()) { if(method=="BYE") { #Account BYE transactions setflag(2); }; route(2); } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } # ----------------------------------------------------------------- # CANCEL processing # ----------------------------------------------------------------- if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; }; t_check_trans(); # ----------------------------------------------------------------- # # ----------------------------------------------------------------- if (method =="OPTIONS" || method=="SUBSCRIBE") {exit;}; if (method =="REGISTER") {route(3);}; route(1); } route[2] { t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; exit; } .. failure_route[2] { if (t_was_cancelled()) { # xlog("INFO:cancelled transaction--\n"); exit; } revert_uri(); .. t_on_failure("2"); append_branch(); if (!t_relay()) { exit; }; }

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Error with 'Contact' field replacing
by Konstantin Shpinev 05 May '10

05 May '10

Hi! I have a problem with Kamailio. Sometimes (can not catch the moment and conditions of) situations arise when, calling from AAA to uri XXX call comes to uri YYY. The call comes from PSTN to the gray subnet of user-agents registered in kamailio. As the PSTN we using Audiocodes mediant. Kamailio ver. 1.5.3 Mediant host: 172.19.32.2, Kamailio host: 172.19.32.3 XXX host: 172.19.32.33, YYY host: 192.168.1.203 I put a line in the log section route_branch: > Apr 28 12:59:20 sipproxy /ks/sbin/kamailio[18294]: [INFO] new branch at sip:XXX@172.19.32.33:5060;user=phone Then in the log module acc wrote: > Apr 28 12:59:21 sipproxy /ks/sbin/kamailio[18302]: ACC: transaction answered: timestamp=1272473961;method=BYE;from_tag=1e76ae41;to_tag=1c1795565414;call_id=17 955646162422000234627(a)172.19.32.2;code=481;reason=Call/Transaction Does Not Exist;from_uri=sip:XXX@172.19.32.3 <sip%3AXXX(a)172.19.32.3>;from_username=XXX;from_name=;from_domain= 172.19.32.3;to_uri=sip:AAA@172.19.32.2 <sip%3AAAA(a)172.19.32.2>;to_username=AAA;to_name=;to_domain=172.19.32.2;request_uri=sip:AAA@172.19.32.2 <sip%3AAAA(a)172.19.32.2>;request_username=AAA;route_id=;route_name=;route_type_id=;destination=;calllist_id= Despite the error, the call passes, but to the another uri. Here's the log of the mediant: 1: > Apr 28 12:52:31 172.19.32.2 ( lgr_flow)(44244101 ) ---- Outgoing SIP Message to 172.19.32.3:5060 ---- Apr 28 12:52:31 172.19.32.2 ACK sip:XXX@172.19.32.33:5060;user=phone SIP/2.0^M Via: SIP/2.0/UDP 172.19.32.2;branch=z9hG4bKac1897823584^M Max-Forwards: 70^M From: <sip:AAA@172.19.32.2 <sip%3AAAA(a)172.19.32.2>>;tag=1c1795565414^M To: <sip:XXX@172.19.32.3 <sip%3AXXX(a)172.19.32.3>>;tag=1e76ae41^M Call-ID: 17955646162422000234627(a)172.19.32.2^M CSeq: 1 ACK Contact: <sip:AAA@172.19.32.2 <sip%3AAAA(a)172.19.32.2>>^M Route: <sip:172.19.32.3;lr;ftag=1c1795565414;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA->^M Supported: em,timer,replaces,path,early-session,resource-priority^M Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE^M *User-Agent: Audiocodes-Sip-Gateway-Mediant 2000/v.5.00A.045.003*^M Content-Length: 0 As you can see, in 1. all is well. But then: 2: > ..... ---- Incoming SIP Message from 172.19.32.3:5060 ---- ....From: <sip:AAA@172.19.32.2 <sip%3AAAA(a)172.19.32.2>>;tag=1c1795565414^M > Call-ID: 17955646162422000234627(a)172.19.32.2^M CSeq: 1 INVITE^M Allow: > INVITE, ACK, CANCEL,OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO^M > Content-Type: application/sdp^M *User-Agent: Zoiper for Windows rev.1105*^M > Content-Length: 232^M ^M v=0^M *o=Zoiper_user 0 0 IN IP4 192.168.1.203^M > s=Zoiper_session^M c=IN IP4 192.168.1.203^*M t=0 0^M m=audio 8000 RTP/AVP > 8 0 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 > telephone-event/8000^M a=fmtp:101 0-15^M a=sendrecv And that is where the substitution occurred at the data fields 'Contact' an entirely different user-agent (uri YYY), which is also registered on Kamailio. That it eventually comes to call. *Repeat: In second SIP message from Kamailio 'Contact' field is replaced by 'Contact' of another (!) user-agent (YYY).* There is a suspicion that an error occurs in a block: > if (!lookup("location")) { switch ($retcode) { case -3: ... case -1: ... case -2: ... } } } I myself anywhere 'Contact' field are not affected, so all suspected to *locate() *from module Registrar.

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kamailio and presence_xml parse error
by Francesco 05 May '10

05 May '10

Hi to all. I'm new here. I'm getting crazy with presence_xml error. I'm using kamailio 1.5.4 notls with xlite 3.0 (winXP client) softphone and postgresql 8.4 as db backend. Seems that presence works, but i have error in parsing xml: those are the relevant log: Apr 29 16:42:54 ocs /usr/sbin/kamailio[2462]: INFO:presence:send_notify_request: NOTIFY sip:user1@ocs.ita.domain.net<sip%3Auser1(a)ocs.ita.domain.net>via sip:user1@10.44.17.145:24258 on behalf of sip:user2@ocs.ita.domain.net<sip%3Auser2(a)ocs.ita.domain.net>for event presence Apr 29 16:42:54 ocs /usr/sbin/kamailio[2463]: ERROR:presence_xml:agregate_xmls: while parsing xml body message Apr 29 16:42:54 ocs /usr/sbin/kamailio[2463]: ERROR:presence_xml:pres_agg_nbody: while aggregating body When i login on kamailio, the users are not immediatly notified to my presence...but, when there's a change of status, the client are notified and on softphone the status change. Other that, if the client close softphone, i can watch this on the log: Apr 29 16:42:41 ocs /usr/sbin/kamailio[2461]: ERROR:presence_xml:offline_nbody: while parsing xml memory Apr 29 16:42:41 ocs /usr/sbin/kamailio[2461]: ERROR:presence_xml:pres_agg_nbody: while constructing offline body Apr 29 16:42:41 ocs /usr/sbin/kamailio[2461]: INFO:presence:send_notify_request: NOTIFY sip:user1@ocs.ita.domain.net<sip%3Auser1(a)ocs.ita.domain.net>via sip:user1@10.44.17.145:24258 on behalf of sip:user3@ocs.ita.domain.net<sip%3Auser3(a)ocs.ita.domain.net>for event presence And the client, on the other softphone, results on line, rather then offline. I'm thinking about encoding database, i've tryied with SQL ASCII and UTF-8 too. Same behaviour. I try different releases (1.5.3 no tls) on different distribution (debian, lucid, alpine linux). Same behaviour. Obiouvsly, the softphone are right configured (presence agent is enabled). Any hints? Thanks for you kind attention. :: Francesco Colista :: Email : francesco.colista(a)gmail.com :: Jabber: francesco(a)jabber.org

3 5

RTPproxy in bridge mode question
by Uriel Rozenbaum 04 May '10

04 May '10

Guys, I'm successfully using a Kamailio + RTPproxy setup in bridge mode with most of my Gateways. My setup includes two different interfaces one with a public IP and teh other with the private IP. Now I'm facing some slight issue. Some providers won't accept my calls (or calls will have some strange behavior) if the Contact header has an IP out of immediate range. I tried to use fix_nated_contact() function but as per my topology, this function will not change the contact header because the IP is already the one on the interface. Example: U 192.168.200.X:5060 -> 192.168.200.Y:5060 INVITE sip:111160911097@192.168.200.Y SIP/2.0. Via: SIP/2.0/UDP 192.168.200.X:5060;branch=z9hG4bK096baacc;rport. From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e. To: <sip:111160911097@192.168.200.Y>. Contact: <sip:60911100@*192.168.200.X*>. U 200.A.A.A:5060 -> 200.B.B.B:5060 INVITE sip:898960911097@200.B.B.B SIP/2.0. Record-Route: <sip:200.A.A.A;r2=on;lr=on;ftag=as32794d5e>. Record-Route: <sip:192.168.200.Y;r2=on;lr=on;ftag=as32794d5e>. Via: SIP/2.0/UDP 200.A.A.A;branch=z9hG4bK5222.14fbf4f7.0. Via: SIP/2.0/UDP 192.168.200.X:5060;received=192.168.200.X;branch=z9hG4bK096baacc;rport=5060. From: "Uriel Rozenbaum" <sip:60911100@192.168.200.X>;tag=as32794d5e. To: <sip:111160911097@192.168.200.Y>. Contact: <sip:60911100@*192.168.200.X*>. Is there any way to let know Kamailio the outgoing IP I'll be using and fix the contact accordingly? I can trigger this change after I know the destination IP. Thanks! Uriel

4 7

[Kamailio-Users] re : openserctl statistics..
by Jignesh Gandhi 03 May '10

03 May '10

Hello, What does the following stat mean when you run the " openserctl moni " sl: received_ACKs = xxxx I did some digging and I got the following explanation... But am not sure if this is an error in operser or an error on a message received by openser or what? " sl: received_ACKs - number of received ACKs due sending negative replies. " any help is appreciated. thanks, --Jignesh

6 11

Add header on failure Route
by Alex Rendour 03 May '10

03 May '10

Hi, We need to add header on the failure route. Exemple: On failure 404 Not found, we need add header Reason: Q.850; cause=1 We tried this in failure_route: if(t_check_status("404")){ xlog("L_ERR", "404 Not found\n"); append_to_reply("Reason: Q.850; cause=1\r\n"); xlog("L_ERR", "404 Not found apres\n"); exit; } and if(t_check_status("404")){ xlog("L_ERR", "404 Not found\n"); append_hf("Reason: Q.850; cause=1\r\n"); xlog("L_ERR", "404 Not found apres\n"); exit; } The twice doesn't work. Someone can help me? Thanks, -- Alexandre Rendour Acropolis Telecom <http://www.acropolistelecom.net> <mailto:rendour@acropolistelecom.net> Adresse : 161-163 avenue Gallieni Paris - Porte de Bagnolet 93170 Bagnolet

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sr-users = ser + kamailio users mailing lists
by Daniel-Constantin Mierla 30 Apr '10

30 Apr '10

Hello, a short note that the merging of SIP Express Routers (SER) users mailing list into sr-users is completed from subscriber point of view: - posts to serusers(a)lists.iptel.org are directed to sr-users(a)lists.sip-router.org - posts to sr-users(a)lists.sip-router.org are received by subscribers of serusers(a)lists.iptel.org Just a bit of time ago, kamailio (openser) users' mailing list was merged as well. Since the release 3.0.x is practically the same source code, no mater the binary is compiled as ser or kamailio, and the development is in the same place (code in same repository), a combined mailing lists environment is the right for providing the best answers in shortest time for both communities. Therefore, now there are two mailing lists: - sr-users(a)lists.sip-router.org - to be used for discussions about stable versions and generic project/community topics - sr-dev(a)lists.sip-router.org - to be used for discussions about devel version and future development topics If you were subscribed to both with same email address, you won't get duplicates. If you had some settings that are no longer in place, please contact one of devels on sr-dev lists or write to me. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

Bug with avp's or feature ?
by Tristan Mahé 30 Apr '10

30 Apr '10

Hi guys, I was debugging an install on kamailio 3.0, and was asked to add per user custom outbound proxies. To do so, I used load_credentials from auth_db to store the custom route in an avp, as you can see in the relevant part below. I noticed a strange thing, POINT 1 show "OK", but POINT 2 shows "KO". Is it a normal behaviour that avp got deleted after the Dialog PART ? Regards, Gled --------------------------------------------------------------------------------- modparam("auth_db", "load_credentials", "$avp(s:proxy)=outbound_override;$avp(s:rights)=rights;$avp(s:max_out)=max_out") route[INVITE_FROM_UAC] { if (!www_authorize("", "my_user_view")) { www_challenge("", "1"); } if ( !($tU =~ $avp(s:rights_ereg)) ) { sl_send_reply("403","Forbidden"); t_release(); exit(); } /* POINT 1 */ if ( is_avp_set("$avp(s:proxy)") ) { xlog("OK"); } else { xlog("KO"); } get_profile_size("outbound","$fU","$avp(calls_out)"); if ( $avp(s:max_out) != 0 && $avp(calls_out) >= $avp(s:max_out) ) { t_reply("403","Too Many calls"); t_release(); exit(); } /* Dialog PART */ if (!is_in_profile("outbound")) { dlg_manage(); set_dlg_profile("outbound","$fU"); } /* POINT 2 */ if ( is_avp_set("$avp(s:proxy)") ) { xlog("OK"); } else { xlog("KO"); } } ---------------------------------------------------------------------------------

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sr-users April 2010