sr-users May 2004

sr-users@lists.kamailio.org

154 participants
333 discussions

[Serusers] nathelper and rtpproxy for IPv4/IPv6
by Curro 28 May '04

28 May '04

Hello everybody I want to use nathelper/rtpproxy to IPv4/IPv6 interworking, but I'm having some problems. I use ser 0.8.13-dev-29 (latest from cvs) and latest rtpproxy (20040107) Ser config file is the same than nathelper example 4to6.cfg Ser starts ok, UA's registers ok. INVITE, 100 Trying, 180 Ringing are ok. Nathelper rewrites SDP bodies ok. When UA2 answer, it sends 200 ok, UA1 receives 200 OK and sends ACK. This ACK is my problem, because SER drops it, and never gets UA2. I suppose that's a problem from my config file, but I can't figure out what. Could you help me, please? Thank you very much.

3 2

[Serusers] SER 0.8.12-0 package installation
by Steve Blair 28 May '04

28 May '04

* * I am trying to install SER on a new RedHat 9 system. I downloaded the ser-mysql-0.8.12-0 package and attempted to install it. It fails with the message 'ser = 0.8.12 is needed by ser-mysql-0.8.12-0'. I installed and started SER 0.8.12 but the about package still fails to install. Should this pacjkaged installation work? If so is there something I am missing? docs/INSTALL does not mention any issues with this package. Thanks, Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-7903 fax: 215-898-9348 sip:blairs@upenn.edu

2 1

[Serusers] two PSTN gateways with SER
by Gustavo García Bernardo 28 May '04

28 May '04

Hi, I have a platform based in SER SIP proxy, and i would like to forward PSTN calls to two Cisco gateways. How could implement the logic for send the calls to the less loaded gateway? Best regards. G.

2 1

[Serusers] BYE notifications not relayed in proxy call forwarding setup
by Robert Bedell 28 May '04

28 May '04

Hello. I've been lurking for a while and I'm hoping that I have a pretty simple problem: My BYE notifications don't seem to be forwarded in a proxy'd call forwarding setup. The call connects fine, and I can talk back and forth ok, but if I hang one side the other remains connected. I tracked it down to the BYE notification not being relayed, but I'm not certain why. Please help! Here is my ser.cfg: debug=3 fork=no log_stderror=yes listen=69.20.20.192 loadmodule "/home/zebra/sys/lib/ser/modules/sl.so" loadmodule "/home/zebra/sys/lib/ser/modules/tm.so" loadmodule "/home/zebra/sys/lib/ser/modules/rr.so" loadmodule "/home/zebra/sys/lib/ser/modules/exec.so" modparam("exec", "setvars", 1) route { if(method=="REGISTER") { log("REGISTER received\n"); sl_send_reply("200","Ok"); } else if(method=="BYE") { log(1,"BYE received\n"); exec_msg("/home/zebra/bin/log_bye.php"); #sl_send_reply("200","Ok"); t_relay(); } else { # all other cases log(1,"message received"); exec_dset("/home/zebra/bin/route_call.php -l"); record_route(); t_relay(); } } =========== This is a very simple call forwarding arrangement. However, when using this SER does nothing with the BYE notifications! They just end up timing out. I've attached the output from sip_scenario.pl. The proxy is 69.20.20.192. Any advice would be greatly appreciated! This is causing me some headaches, and maybe I'm just not understanding what I'm doing with SER.

2 1

[Serusers] Need help to install mysql.
by gc 28 May '04

28 May '04

I am new to ser. After installing the ser, I tried to install mysql by following quick start steps but I got error saying "ser_mysql.sh: line 1:hen_ha1: command not found" Here is what I did to install mysql: mysh> rpm -i ser-mysql-0.8.12-0.i386.rpm mysh> /usr/local/sbin/sh ser_mysql.sh create MySql password for root: dud1234 (I put in my own password) ser_mysql.sh line1: gen_ha1:command not found HA1 calculation failed Before I install mysql, ser works fine without Authentication. I can use my ip phone to make call. Gary

2 4

[Serusers] public ip - private ip communication problem
by varala ramakanth 28 May '04

28 May '04

hello friends, sorry to disturb you people again and again iam newbie i know ser from last two weeks only as iam suffering with this problem from last week i need help of you people i hope some body is kind enough to help me out . and my first scenario in simple ascii diagram public ip (estara softphone) /|\ | | \|/ SER server (public ip) /|\ | \|/ public ip (estara softphone) my SER server is redhat linux 9.0 iam using stable version which i got through cvs and first checking with the public ip to publics ip here i could able to establish the call i.e in either side i could able to listen the voice in this iam not using any rtpproxy ------------------------------------------------------ [root@server sbin]# ser Listening on 127.0.0.1 [127.0.0.1]:5060 <public ip>[public ip]:5060 Aliases: server.pol.net.in:5060 localhost:5060 localhost.localdomain:5060 stateless - initializing [root@server sbin]# Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 9(0) INFO: fifo process starting: 17451 9(17451) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 8(17450) parse_nameaddr(): No < found 8(17450) parse_rr(): Error while parsing name-addr 8(17450) find_first_route(): Error while parsing Route HF 6(17448) parse_nameaddr(): No < found 6(17448) parse_rr(): Error while parsing name-addr 6(17448) find_first_route(): Error while parsing Route HF 6(17448) parse_nameaddr(): No < found 6(17448) parse_rr(): Error while parsing name-addr 6(17448) find_first_route(): Error while parsing Route HF 7(17449) parse_nameaddr(): No < found 7(17449) parse_rr(): Error while parsing name-addr 7(17449) find_first_route(): Error while parsing Route HF 6(17448) parse_nameaddr(): No < found 6(17448) parse_rr(): Error while parsing name-addr 6(17448) find_first_route(): Error while parsing ------------------------------------------------------- this is the second scenario public ip (msn messenger) /|\ | | \|/ SER server (public ip) /|\ | \|/ DHCP server (public ip , NAT device) /|\ | \|/ private ip (estara softphone) in the terminal of ser iam gettign this ----------------------------------------------------- [root@server sbin]# ser Listening on 127.0.0.1 [127.0.0.1]:5060 <public ip>[public ip]:5060 Aliases: server.pol.net.in:5060 localhost:5060 localhost.localdomain:5060 [root@server sbin]# stateless - initializing Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 9(0) INFO: fifo process starting: 17642 9(17642) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 5(17638) parse_nameaddr(): No < found 5(17638) parse_rr(): Error while parsing name-addr 5(17638) find_first_route(): Error while parsing Route HF 8(17641) parse_nameaddr(): No < found 8(17641) parse_rr(): Error while parsing name-addr 8(17641) find_first_route(): Error while parsing Route HF 5(17638) ERROR: extract_body: message body has lenght zero 5(17638) ERROR: force_rtp_proxy: can't extract body from the message 5(17638) ERROR: on_reply processing failed ----------------------------------------------------- here iam using the rtpproxy of version 1.4 2003/08/05 ./rtpproxy -f in the terminal of rtpproxy iam getting this ----------------------------------------------------- [root@server rtpproxy]# ./rtpproxy -f rtpproxy: new session on a port 35000 rtpproxy: lookup on a port 35000 rtpproxy: addr1 filled in: 202.65.128.24 rtpproxy: addr2 filled in: 202.65.148.252 rtpproxy: stats: 179 in from addr1, 3 in from addr2, 180 relayed rtpproxy: session on port 35000 is cleaned up ------------------------------------------------------ the result is i could able to see that both mic and speaker are working and iam listening what ever public ip softphone is speaking in private ip softphone. but in public ip softphone i have seen that only mic is working not the speakers i.e i could not able to listen what ever private ip softphone is speaking i observed that one malformed sip packet is genrating thorugh the ser in tethereal so my ethereal report is **************************************************** Frame 3 (327 bytes on wire, 327 bytes captured) Arrival Time: May 23, 2004 20:18:03.259293000 Time delta from previous packet: 2.401569000 seconds Time relative to first packet: 2.402778000 seconds Frame Number: 3 Packet Length: 327 bytes Capture Length: 327 bytes Ethernet II, Src: 00:e0:2b:90:1f:00, Dst: 00:e0:18:ed:04:61 Destination: 00:e0:18:ed:04:61 (Asustek__ed:04:61) Source: 00:e0:2b:90:1f:00 (Extreme__90:1f:00) Type: IP (0x0800) Internet Protocol, Src Addr: 202.*.*.252 (dhcp i.e nat server) (202.*.*.252), Dst Addr: 202. *.*.19 (202.*.*.19) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) 0000 00.. = Differentiated Services Codepoint: Default (0x00) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 313 Identification: 0xf5f8 Flags: 0x00 .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 125 Protocol: UDP (0x11) Header checksum: 0x9d28 (correct) Source: 202.*.*.252 (202.*.*.252) Destination: 202.*.*.19 (202.*.*.19) User Datagram Protocol, Src Port: 62812 (62812), Dst Port: 5060 (5060) Source port: 62812 (62812) Destination port: 5060 (5060) Length: 293 Checksum: 0x8112 (correct) Session Initiation Protocol Request line: REGISTER sip:202.*.*.19 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.18:5060 From: sip:203@202.*.*.19 To: sip:203@202.*.*.19 Contact: sip:203@192.168.1.18:5060 Call-ID: e166bf60-acf2-11d8-aafe-00e01846e257(a)192.168.1.18 CSeq: 32320301 REGISTER Content-Length: 0 Expires: 3600 Frame 4 (629 bytes on wire, 629 bytes captured) Arrival Time: May 23, 2004 20:18:03.261431000 Time delta from previous packet: 0.002138000 seconds Time relative to first packet: 2.404916000 seconds Frame Number: 4 Packet Length: 629 bytes Capture Length: 629 bytes Ethernet II, Src: 00:e0:18:ed:04:61, Dst: 00:e0:2b:90:1f:00 Destination: 00:e0:2b:90:1f:00 (Extreme__90:1f:00) Source: 00:e0:18:ed:04:61 (Asustek__ed:04:61) Type: IP (0x0800) Internet Protocol, Src Addr: 202.*.*.19 (202.*.*.19), Dst Addr: 202.*.*.252 (202.*.*.252) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00) 0001 00.. = Differentiated Services Codepoint: Unknown (0x04) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 615 Identification: 0x0000 Flags: 0x04 .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x8ee3 (correct) Source: 202.*.*.19 (202.*.*.19) Destination: 202.*.*.252 (202.*.*.252) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62812 (62812) Source port: 5060 (5060) Destination port: 62812 (62812) Length: 595 Checksum: 0x0d6c (correct) Session Initiation Protocol Status line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.18:5060;rport=62812;received=202.*.*.252 From: sip:203@202.*.*.19 To: sip:203@202.65.128.19;tag=b27e1a1d33761e85846fc98f5f3a7e58.d036 Call-ID: e166bf60-acf2-11d8-aafe-00e01846e257(a)192.168.1.18 CSeq: 32320301 REGISTER Status line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.1.18:5060;rport=62812;received=202.*.*.252 From: sip:203@202.*.*.19 To: sip:203@202.*.*.19;tag=b27e1a1d33761e85846fc98f5f3a7e58.d036 Call-ID: e166bf60-acf2-11d8-aafe-00e01846e257(a)192.168.1.18 CSeq: 32320301 REGISTER Contact: <sip:203@202.*.*.252:62812>;q=0.00;expires=3600 Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 202.*.*.19:5060 "Noisy feedback tells: pid=17639 req_src_ip=202.*.*.252 req_src_p ort=62812 in_uri=sip:202.*.*.19 out_uri=sip:202.65.128.19 via_cnt==1" Frame 5 (46 bytes on wire, 46 bytes captured) Arrival Time: May 23, 2004 20:18:18.476274000 Time delta from previous packet: 15.214843000 seconds Time relative to first packet: 17.619759000 seconds Frame Number: 5 Packet Length: 46 bytes Capture Length: 46 bytes Ethernet II, Src: 00:e0:18:ed:04:61, Dst: 00:e0:2b:90:1f:00 Destination: 00:e0:2b:90:1f:00 (Extreme__90:1f:00) Source: 00:e0:18:ed:04:61 (Asustek__ed:04:61) Type: IP (0x0800) Internet Protocol, Src Addr: 202.*.*.19 (202.*.*.19), Dst Addr: 202.*.*.252 (202.*.*.252) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00) 0001 00.. = Differentiated Services Codepoint: Unknown (0x04) .... ..0. = ECN-Capable Transport (ECT): 0 Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00) 0001 00.. = Differentiated Services Codepoint: Unknown (0x04) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 32 Identification: 0x0000 Flags: 0x04 .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x912a (correct) Source: 202.*.*.19 (202.*.*.19) Destination: 202.*.*.252 (202.*.*.252) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 62812 (62812) Source port: 5060 (5060) Destination port: 62812 (62812) Length: 12 Checksum: 0x4d22 (correct) [Malformed Packet: SIP] ********************************************************* i kept * inorder to hide the public ip please pardon me for that ser.cfg is of klaus which he kept in the we last febaraury with some light modifications confined to my problem my ser.cfg is ****************************************************** # # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) # Uncomment these lines to enter debugging mode /* 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=5060 children=4 fifo="/tmp/ser_fifo" alias=server.pol.net.in #------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # load the voicemail module #loadmodule "/usr/local/lib/ser/modules/vm.so" # load the enum module #loadmodule "/usr/local/lib/ser/modules/enum.so" # load the group module, to verify if a user forwards to voicemail #loadmodule "/usr/local/lib/ser/modules/group.so" # load the nathelper module loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- registrar parameter # special NAT flag indicates that a registered client is behind NAT modparam("registrar", "nat_flag", 6) # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #/ #odparam("usrloc", "db_mode", 0) #modparam("usrloc", "db_url", "mysql://ser:XXXXXXX@localhost/ser") # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") #modparam("auth_db", "db_url", "mysql://ser:XXXXXXX@localhost/ser") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- voicemail params -- #modparam("voicemail", "db_url","mysql://ser:XXXXXXX@localhost/ser") # -- voicemail params -- #modparam("group", "db_url","mysql://serro:XXXXXXXXX@localhost/ser") # -- nathelper params -- modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("tm", "fr_inv_timer", 30 ) #/modparam("register","nat_flag",6) #modparam("tm", "fr_inv_timer", 8 ) # ------------------------- request routing logic ------------------- # main routing logic if (nat_uac_test("2")) { force_rport(); fix_nated_contact(); if (method=="INVITE") { fix_nated_sdp("1"); } append_hf("P-hint: fixed NAT contact for request\r\n"); setflag(5); }; 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()) { log(1, "loose_route processing\n"); t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (isflagset(5)) { setflag(6); ...skipping 29 lines if (method=="INVITE") { force_rtp_proxy(); append_hf("P-hint: request forced to rtp proxy\r\n"); setflag(7); }; }; if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (isflagset(6)) { fix_nated_contact(); append_hf("P-hint: fixed NAT contact for response\r\n"); } if ( (status=~"200" || status=~"183") ) { if ( isflagset(7) ) { force_rtp_proxy(); append_hf("P-hint: response forced to rtp proxy\r\n"); }; }; } route[4]{ # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "REFER" || method == " BYE")) { log(1, "no invite,ack,cancel,refer->return 404\n"); sl_send_reply("404", "Not Found"); break; }; # not voicemail subscriber and no echo/conference call if ( isflagset(4)) { log(1, "flag(4) active\n"); }; if (uri =~ "conference") { log(1, "conference call\n"); }; if (uri =~ "echo") { log(1, "echo call\n"); }; if ( !( isflagset(4) || (uri =~ "conference") || (uri =~ "echo") ) ) { log(1, "no voicemail subscriber->return 404"); sl_send_reply("404", "Not Found and no voicemail turned on"); break; }; if ( isflagset(5) ) { log(1, "caller is NATed->record_route\n"); record_route(); log(1, " -->setting up reply processing ->onreply_route[1]"); t_on_reply("1"); if (method=="INVITE") { log(1, " INVITE request-->force_rtp_proxy"); force_rtp_proxy(); }; }; } ****************************************************** my requirement is private to public call establishment so please kindly guide me with regards rama kanth __________________________________ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/

2 1

[Serusers] Using ser_mysql.sh failed
by Lin-Kai Cheng 28 May '04

28 May '04

I tried to use "/usr/sbin/ser_mysql.sh" to create SER database in MySQL. I typed the password of MySQL and entered the realm But the result showed that /usr/sbin/ser_mysql.sh: line 1: gen_ha1: command not found HA1 calculation failed I've tried to type config variable CMD under command line to see whether I can enter MySQL or not. Yes! The CMD works fine. Then I have no idea about the gen_ha1. Please give me some suggestions. Thanks. My config vars are as follows, DBNAME=ser DBHOST=mymachine USERNAME=ser DEFAULT_PW=heslo ROUSER=serro RO_PW=47serro11 SQL_USER="root" CMD="mysql -u$SQL_USER -h $DBHOST " DUMP_CMD="mysqldump -h $DBHOST -u$SQL_USER -c -t " BACKUP_CMD="mysqldump -h $DBHOST -u$SQL_USER -c " TABLE_TYPE="TYPE=MyISAM" # user name column USERCOL="username" GENHA1='gen_ha1'

3 2

[Serusers] FW: Looping and sending of multiple ACK request
by Shirley Toh 28 May '04

28 May '04

> Hi, > > Please help! I am trying to forward calls to PSTN gateway. But I run the > ser code below, the call went through for the PSTN gateway but I can see > multiple request of ACK. Anyone know what is happening ...thanks.!! Please > HELP. > > <<ser.txt>> <<error.txt>> > Regards, > Shirley

2 1

[Serusers] about Suse
by Chen Youcheng 28 May '04

28 May '04

Hello, i have two questions. 1. There is no packages since version ser-version-12. Why? 2. In oder to support mysql module, what mysql packets must be installed? standard or more? Please see http://dev.mysql.com/downloads/mysql/4.0.html. Regards Yongcheng __________________________________ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/

1 0

[Serusers] Call Accounting - multiple INVITE entries
by jon＠bostontech.com 28 May '04

28 May '04

I am running ser 0.8.12 (on RedHat 9) and have it set up to log all invite and bye messages, so that I can use the accounting feature in SerWeb. It's logging fine. My problem is that if there are multiple invites and only one bye during a session, I get multiple entries in SerWeb with the same sip_callid. I noticed that this happens when the hold feature is used with the XTen soft phone. Is there any way I can avoid logging these subsequent INVITE messages or stop from displaying them in the call log on SerWeb? My accounting routine: modparam("acc", "log_level",1) modparam("acc", "db_flag", 1) if ((method=="INVITE") || (method=="BYE")) { setflag (1); }; Serweb $q call to MySQL (4.0.18) db: $q="select t1.to_uri, t1.from_uri, t1.sip_callid, t1.time, ". "t1.fromtag as invft, t2.fromtag as byeft, t2.totag as byett, ". "sec_to_time(unix_timestamp(t2.time)-unix_timestamp(t1.time)) ". "as length ". "from ".$config->table_accounting." t1, ". $config->table_accounting. " t2 ". "where t1.domain='".$config->realm."' and ". "t1.sip_callid=t2.sip_callid and ". "t1.sip_method='INVITE' and t2.sip_method='BYE'". "order by t1.time desc"; Thanks -Jon B

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sr-users May 2004