sr-users March 2004

sr-users@lists.kamailio.org

145 participants
286 discussions

by Kapil Dhawan

Hi i want to allow other ppl to use my declared function..how can this be done.... what i have done is..i have properly declared a function call like this {"load_tm", (cmd_function)load_tm, NO_SCRIPT, 0, 0}, as in tm module...but how can we include all this stuff in some other module or lets say....vm module declares one function called vm_action which can be called by ser.cfg only...but i will change it to be called from other modules as well...now in my module how can i use that function vm_action...how to define it..etc...i hope i explained my problem.... regards _________________________________________________________________ Skin is in! Bollywood is sizzling. http://server1.msn.co.in/slideshow/striptease/ Check out these hot pics!

20 years, 11 months

[Serusers] SER receving call from PSTN and forwardind to alias

by Fábio Silvestri

Hi. I want to forward to a subscriber a incoming call from Cisco VoIP Gateway/ IOS 12.x/ SIP enabled. I get this on ngrep: U 200.1.1.102:50022 -> 200.1.1.121:5060 INVITE sip:551112344321@200.1.1.121;user=phone;phone-context=unknown SIP /2.0..Via: SIP/2.0/UDP 200.1.1.102:5060..From: "899961430" <sip:8999614 30(a)200.1.1.102>..To: <sip:551112344321@200.1.1.121;user=phone;phone-c ontext=unknown>..Date: Wed, 03 Mar 2004 16:45:10 rio..Call-ID: 1D998259-6C8 211D8-9B358622-2F08A8D5@200.1.1.102..Cisco-Guid: 496559665-1820463576-26 03845154-789096661..User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled.. CSeq: 101 INVITE..Max-Forwards: 5..Timestamp: 1078343110..Contact: <sip:899 961430@200.1.1.102:5060;user=phone>..Expires: 180..Content-Type: applica tion/sdp..Content-Length: 167....v=0..o=CiscoSystemsSIP-GW-UserAgent 8136 7 228 IN IP4 200.1.1.102..s=SIP Call..c=IN IP4 200.1.1.102..t=0 0..m=au dio 20136 RTP/AVP 65535 65535 65535 4 18 65535 8.. # U 200.1.1.121:5060 -> 200.1.1.102:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 200.1 96.28.102:5060..From: "899961430" <sip:899961430@200.1.1.102>..To: <sip: 551112344321(a)200.1.1.121;user=phone;phone-context=unknown>..Call-ID: 1D9 98259-6C8211D8-9B358622-2F08A8D5@200.1.1.102..CSeq: 101 INVITE..Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))..Content-Length: 0.... # U 200.1.1.121:5060 -> 200.1.1.102:5060 SIP/2.0 500 I'm terribly sorry, server error occured (2/TM)..Via: SIP/2.0/U DP 200.1.1.102:5060..From: "899961430" <sip:899961430@200.1.1.102>.. To: <sip:551112344321@200.1.1.121;user=phone;phone-context=unknown>;tag= edce4e4a3d91650ad93d450210a72a7e-fb20..Call-ID: 1D998259-6C8211D8-9B358622- 2F08A8D5@200.1.1.102..CSeq: 101 INVITE..Server: Sip EXpress router (0.8. 12-tcp_nonb (i386/linux))..Content-Length: 0.... # I want to do when a SER receive a call from Cisco, then this call be forwarded to a subscriber, using alias table. Does this can be done? Regards. -- |o |o Experiência é o nome que nós damos aos nossos próprios erros. |o |o Fabio Silvestri |o fabio(a)informatec.com.br |o ICQ: 1667351 |o

20 years, 11 months

[Serusers] Calls cuts off

by Billy

Hello, I'm facing some very different when making calls with an ATA... it happens exactly as a msg written to the list a couple months ago, but I found no response for this question. It's weird cause it does not happen when I call from an ATA to another ATA with alias on the same proxy. Any clues?! Thanks.. Billy [Serusers] Session Header != Media ? Kannaiyan Natesan nkans at lycos.co.uk Fri Nov 21 19:57:33 CET 2003 Previous message: [Serusers] ser 0.8.11-r1 bugfix release Next message: [Serusers] Ser behind NAT - possible? Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] I tried to call using cisco ata 186 within itself. But after few seconds it automatically cuts off automatically. Below is the log from ata 186. Please help. Thanks. Kannaiyan [0:0]Start RTP Rx RTP Rx Init: 0, 0 [0]INVITE Retry 0 Response=100; Trying Session Header != Media AVT: payload=101 [1]: OOB DTMF [1:0]Start RTP Rx RTP Rx Init: 1, 1 SCC:ev=21[1:0] 0 0 SCC:ev=5[1:0] 13 0 [1:0]RINGING SCC Cmd[1:0]ALERT Response=180; Ringing SCC:ev=10[0:0] 3 0 [0]StartTone 3 [1]StopTone SCC Cmd[1:0]ANSWER [1]R2INV Retry 0 Session Header != Media AVT: payload=101 Response=200; OK [0]: OOB DTMF [0:0]Rel LBRC Res Frames per packet=2 SetG723Mode: 2 0 [0]Enable encoder 0 [0]: EC 1 RTP TX[0]:SSRC_ID = 1803d488 RTP Tx Init: 0, 0 [0]Enable encoder 0 Enable LEC adapt [0]=1 [0]Start/Resume Media-> 192.168.0.2:16386 (pt=0) SCC:ev=12[0:0] 3 0 [0]StopTone [1]DPKT 1st: 800 560, pt 13 [1]R2INV Retry 1 Session Header != Media AVT: payload=101 [1]R2INV Retry 2 Session Header != Media AVT: payload=101 [1]R2INV Retry 3 Session Header != Media AVT: payload=101 [1]R2INV Retry 4 Session Header != Media AVT: payload=101 [1]R2INV Retry 5 Session Header != Media AVT: payload=101 [1]R2INV Retry 6 Session Header != Media AVT: payload=101 [1]R2INV Retry 7 Session Header != Media AVT: payload=101 change next hop[1]: c = 2 R2INV TO; NoAltProxy Timeout waiting for ACK [1]BYE Retry 0 Got BYE/CANCEL Resp SCC:ev=13[1:0] 6 0 [1]StartTone 2 [1]StopTone SCC Cmd[1:0]RELEASE CCBS expired >>> SIP Soft Reset [1:0] <<< [1:0]Rel LBRC Res [1]MPT mode 0 SCC Cmd[0:0]DISCONNECT [0]BYE Retry 0 Got BYE/CANCEL Resp SCC:ev=13[0:0] 10 0 SCC Cmd[0:0]RELEASE

20 years, 11 months

Re: [Serusers] Strange error when using group module

by Klaus Darilion

If you want to authorize INVITEs before sending them to the gateway, you should also challenge the INVITEs. And you should use proxy_challenge for INVITEs and www_challenge for REGISTERs. you can try this: if (method=="INVITE") { if (!proxy_authorize("sip.netsquare.co.uk","subscriber")) { proxy_challenge("sip.netsquare.co.uk", "0"); break; }; if (!is_user_in("credentials","int")) { sl_send_reply("403", " no permission for external calls"); break; }; ...here forward to gateway }; btw: please always CC: to the mailing list. regards, klaus Mike Fallows wrote: > There is a proxy_athorize in the register statement as below. > > > > if (method=="REGISTER") { > > # Uncomment this if you want to use digest authentication > if (!proxy_authorize("sip.netsquare.co.uk", > "subscriber" > )) > { > proxy_challenge("sip.netsquare.co.uk", "0"); > break; > }; > > save("location"); > break; > }; > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: 03 March 2004 09:34 > To: Mike Fallows > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Strange error when using group module > > > There are no credetials (username/password) in the incoming request. > Have you requested credentials with proxy_authorize(...) ? > > klaus > > Mike Fallows wrote: > >>Anyone help me with this one. >> >>I am trying to group users so that some are allowed place SIP->SIP calls > > via > >>the proxy, and others are also allowed to access the PSTN. >> >>An extract of the ser.cfg is below: >> >>if (!is_user_in("credentials","int")) { >> sl_send_reply("403", " no permission for external calls"); >> break; >> } >> >>When this is called, I receive this in /var/log/messages: >> >>Mar 2 22:04:41 ns-nby-ser1 /usr/local/sbin/ser[6988]: is_user_in(): No >>authorized credentials found (error in scripts) >> >>And the user fails to connect. >> >>Any ideas would be greatly appreciated. >> >>Mike >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> > > > >

20 years, 11 months

[Serusers] SER talks SIP to Jabber Server

by Délio Filipe Russo Guerra

I'm having a problem putting jabber gateway to work. I'm using SER 0.8.12 and microsoft messenger 4.7 as a SIP client and PSI as a Jabber client. I had configured the jabber gateway in ser.cfg and i can talk between the to clients (at least for a while). I'm using user1 as both SIP client and jabber client, and user2 as a jabber client only. So the SIP address for the user1 is sip:user1@sipserver.sipdomain. user1 also have an jabber account user1(a)jabberserver.jabberdomain. user2 as an jabber account user2(a)jabberserver.jabberdomain. I start to talk from user1 (microsoft messenger logged in SIP server) to user2 (PSI logged in jabber server) and it works just fine. If i continue to talk in this direction i have no problems. When i start to talk from user2 (PSI logged in jabber server) to user1 (microsoft messenger logged in SIP server) i have no problems either. The problem appears when i want to talk again from user1 to user2... that is, when i receive the first response from user2 -> user1 it blocks the way user1 -> user2. I've put a sniffer to find out why this appened, and appears that after user2 responds, all the events (Ex: INFO for Keyboard Activity) or messages from user1->user2 are SIP to jabber server. It seams that SER is not using the jabber gateway. Can anyone help me please... Délio Guerra IT - Instituto de Telecomunicações Universidade de Aveiro - Portugal

20 years, 11 months

[Serusers] ops i forget to say voicemail

by Welesley Sibelson Dias

voicemail and sems works fine because when log direct em voicemail server, i receive msg and receive email correct. # # $Id: voicemail.cfg,v 1.2 2003/10/10 06:27:18 jiri Exp $ # # this script is configured for use as voicemail UAS; it processes # INVITEs and BYEs and asks SEMS to record media via "vm"; in this # script, all record-routing and other constructs known from proxy # scripts are not present -- it is a simple UAS # # ----------- global configuration parameters ------------------------ #debug= # debug level (cmd line: -dddddddddd) #fork=no #log_stderror=yes # (cmd line: -E) debug=7 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5090 children=4 fifo="/tmp/vm_ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules//mysql.so" loadmodule "/usr/local/lib/ser/modules/vm.so" # ----------------- setting module-specific parameters --------------- modparam("voicemail", "db_url","mysql://ser:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; if (!uri==myself) { sl_send_reply("404", "not reponsible for host in r-uri"); break; }; # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if (uri=~"sip:as_welcome@.*" || uri=~"sip:as_nomoney@.*") { if (!vm("/tmp/am_fifo", "announcement")) { log("couldn't contact announcement server\n"); t_reply("500", "couldn not contact announcement server"); }; } else { if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); }

20 years, 11 months

[Serusers] ser 0.8.12 is not booting successfully

by Ansari, Fahriz

Is there any body who can help to solve this problem please? Mysql is running and I am trying to run ser 0.8.12 with authorization configuration. It displays the message Ser starting: Then there is no message of pass or fail. Thanks, AQEEL USA.

20 years, 11 months

[Serusers] sip-communicator

by Eduard San Anselmo Mateu

Hello everyone. I'm testing SER with some different clients, to see which one fits our needs better. The last one we tried was sip-communicator (java softphone), but I'm experiencing some problems with it. At first, everything seems to work well, but after a few seconds, the client says it's unregistered (when that's not true). Another issue is that when SER returns the 200 OK response to the REGISTER request, it has three different Contact headers, with different port numbers. The config file is this one: listen=192.168.0.15 alias=albasoft.com alias=albasoft.com:5060 alias=192.168.0.15 #alias=192.168.20.15 #alias=192.168.20.15:* #alias=192.168.20.15 # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/nathelper.so" modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; force_rport(); save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; if (method=="INVITE" && !uri=~"sip:.*@192\.168\.0\..*") { log(1,"IEEE: la peticio va cap a fora\n"); #fix_nated_sdp("1"); #force_rtp_proxy(); #t_on_reply("1"); }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status =~ "(183)|2[0-9][0-9]") { if (!src_ip=~"192\.168\.0.\.*") { log(1,"onreply_route si que funciona\n"); #force_rtp_proxy(); }; }; } By the way, sip-communicator doesn't seem to be symmetric: the clients works on a random port, and the server, on 5060 (configurable), and those different Contact headres I've explained, are for 5060, and the three ports following the one sip-communicator has made the request from (for example, 1066, 1067, 1068). Thanks in advance. Eduard San Anselmo

20 years, 11 months

[Serusers] register failed. with always these message: Expires

by vivian huang

U 192.168.0.102:5060 -> 192.168.0.145:5060 REGISTER sip:server.itginc.jp SIP/2.0..Content-Length: 0..Contact: <si p:192.168.0.102>..Call-ID: 1D829350-79CC-42FD-93B7-0FBC361C65E2(a)192.16 8.0.102..From: <sip:vivian@server.itginc.jp>;tag=19895398..CSeq: 7 REG ISTER..To: <sip:vivian@server.itginc.jp>..Via: SIP/2.0/UDP 192.168.0.1 02:5060.... # U 192.168.0.145:5060 -> 192.168.0.102:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.102:5060..From: <sip:vivian @server.itginc.jp>;tag=19895398..To: <sip:vivian@server.itginc.jp>..Ca ll-ID: 1D829350-79CC-42FD-93B7-0FBC361C65E2@192.168.0.102..CSeq: 7 REG ISTER..Expires: 600..Content-Length: 0....

20 years, 11 months

[Serusers] Multiple IP Addresses

by Alan Litster

Hi List, Is there a way to force SER to send requests from a user defined IP address when used on a multihomed system? We have a box with two consecutive addresses 81.x.x.81 81.x.x.82 the primary address is .82 but it always uses .81, doesn't matter which way around the listen statements are. Many Thanks, Alan ------------------------------------------------------------------------------------------------------- This email, and any files transmitted with it, is copyright and may contain confidential information. The contents are intended for the use of the addressee(s) only. Unauthorized use may be unlawful. If you receive this email by mistake, please advise sender immediately. The views of the author may not necessarily constitute the views of Telco Electronics Limited. Nothing in this mail shall bind Telco Electronics Limited in any contract or obligation. Telco Electronics Limited 6-8 Oxford Court Brackley Northants NN13 7XY Tel 07000 701999 Fax 07000 701777

20 years, 11 months

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sr-users March 2004