sr-users November 2005

sr-users@lists.kamailio.org

245 participants
571 discussions

[Serusers] Problem in SER+SEMS conference with NAT
by Ravic Costa 03 Nov '05

03 Nov '05

Hello, We have configured SER 0.9.4 and SEMS last cvs version in order to implement a conference funcionallity. Everything is working fine when the clients are using public IPs. When a client starts a conference, he listens a message that in on a wav file. When a client is behind NAT, when he tries to start a conference, he doen's listen to any message and in the soft phone (X-Lite) it appears a message saying "connected". We have captured the traffic with tcpdump, on the machine that ser and sems is running, and we've noticed that the RTP packets that the client behind NAT send are well received, however, there is only one RTP packet sent by the SER to the Client, and the Client does not get that message, maybe because he is behind NAT. Any ideia? In the ser.cfg file, to the conference funcionallity we have : _________________________________________________________ route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[5] { # select messages to redirect: if ( method=="ACK" || method=="INVITE" || method=="BYE" || method=="CANCEL" ){ # switch to stateful mode: if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; # prevent timeout on the other side: t_reply("100","Trying - just wait a minute !"); # actively absorb ACKs if (method == "ACK") { t_relay(); break; } if (method=="INVITE"){ if (nat_uac_test("19")) { setflag(6); } # redirect the call to the 'conference' plug-in # if the URI begin with 100 if (uri=~"sip:100.*@") { # assumes that Sems configuration parameter 'socket_name=' # has been set to /tmp/am_sock if(!t_write_req("/tmp/am_fifo","conference")) { t_reply("500","error contacting sems"); }; break; route(4); route(1); }; } else if (method=="BYE" || method=="CANCEL") { # Sems should already know which plug-in is handling that # call. "bye" is no plug-in name. It is a reserved name which # tells Sems to terminate the call. if(!t_write_req("/tmp/am_fifo","bye")) { t_reply("500","error contacting sems"); }; }; }; } _______________________________________________________ Best regards, Ravic Costa

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[Serusers] Redirect server
by harry gaillac 03 Nov '05

03 Nov '05

Hello, I need your advises Here is my problem : I setup asterisk ser : Asterisk 5050-- |--Firewall---|----- SIP User agents SER 5060--- My client use SER as outbound proxy and asterisk as sip registrar. SER is able to forward SIP register request to asterisk however . How SER can inform SIP User agents to send register requests to asterisk. May I have to configure a redirect server to advise SIP User agents of the sip register server (asterisk)? can you help me for ser.cfg if possible ? Regards ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] accounting on mysql
by Colin Jordan 03 Nov '05

03 Nov '05

Hi there Can you please help me to setup accounting on mysql. I need to log the call transactions. Best regards, Colin

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[Serusers] problem with ser registration
by rajesh volluru 03 Nov '05

03 Nov '05

hi I get Process Request Timedout ever whenever i try to register with SER. I get following messages on starting ser. 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 0(0) DBG: open_uac_fifo: opening fifo... 0(0) DEBUG: FIFO created @ /tmp/ser_fifo 0(0) DEBUG: fifo /tmp/ser_fifo opened, mode=432 0(0) init_unixsock_socket: No unix domain socket will be opened 1(0) DEBUG: init_mod_child (-1): tm 1(0) DEBUG: callid: '3fbfd771-0(a)69.229.23.57' 1(0) DEBUG: init_mod_child (-1): usrloc 1(0) get_connection(): Connection not found in the pool 2(0) INFO: fifo process starting: 10317 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '3fbfd771-0(a)69.229.23.57' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(10313) DEBUG: init_mod_child (1): tm 0(10313) DEBUG: callid: '3fbfd771-10313(a)69.229.23.57' 0(10313) DEBUG: init_mod_child (1): usrloc 0(10313) get_connection(): Connection not found in the pool 1(10316) DEBUG: init_mod_child (-1): auth_db 1(10316) get_connection(): Connection not found in the pool 0(10313) DEBUG: init_mod_child (1): auth_db 0(10313) get_connection(): Connection not found in the pool 1(10316) DEBUG: init_mod_child (-1): domain 1(10316) DEBUG: init_mod_child (-1): nathelper 1(10316) rtpp_test: RTP proxy found, support for it enabled 0(10313) DEBUG: init_mod_child (1): domain 0(10313) get_connection(): Connection found in the pool 0(10313) DEBUG: init_mod_child (1): nathelper 0(10313) rtpp_test: RTP proxy found, support for it enabled 2(10317) DEBUG: init_mod_child (-2): auth_db 2(10317) get_connection(): Connection not found in the pool 2(10317) DEBUG: init_mod_child (-2): domain 2(10317) DEBUG: init_mod_child (-2): nathelper 2(10317) rtpp_test: RTP proxy found, support for it enabled 2(10317) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(10317) DEBUG: register_fifo_cmd: new command (print) registered 2(10317) DEBUG: register_fifo_cmd: new command (uptime) registered 2(10317) DEBUG: register_fifo_cmd: new command (version) registered 2(10317) DEBUG: register_fifo_cmd: new command (pwd) registered 2(10317) DEBUG: register_fifo_cmd: new command (arg) registered 2(10317) DEBUG: register_fifo_cmd: new command (which) registered 2(10317) DEBUG: register_fifo_cmd: new command (ps) registered 2(10317) DEBUG: register_fifo_cmd: new command (kill) registered 2(10317) WARNING: no fifo_db_url given - fifo DB commands disabled! Can anyone please have a look at above information and let me know what is wrong with ser? Thanks Rajesh --------------------------------- Enjoy this Diwali with Y! India Click here --------------------------------- Enjoy this Diwali with Y! India Click here

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[Serusers] SRC Port Change
by Ritesh Jalan 03 Nov '05

03 Nov '05

Hi Pls. help us in resolving the problem that the SIP proxy server changes the Source and Destination port while replying for message. Pls. find the below message for the Reference In below message A cisco ata device (which is behind NAT with IP 192.168.10.52) is trying to register it self with SIP proxy server IP 172.16.101.10 with SRC port 5060 and DST port 5060, but when SIP proxy server replies back to the message the port number gets changed i.e SRC port 53492 and DST port 5060. Is there any way by which we can fix this port num. No. Time Source Destination Protocol Info 532 152.672945 172.16.101.251 172.16.101.10 SIP Request: REGISTER sip:172.16.101.10 Frame 532 (469 bytes on wire, 469 bytes captured) Arrival Time: Nov 3, 2005 11:32:02.280623000 Time delta from previous packet: 0.007492000 seconds Time since reference or first frame: 152.672945000 seconds Frame Number: 532 Packet Length: 469 bytes Capture Length: 469 bytes Protocols in frame: eth:ip:udp:sip Ethernet II, Src: Cisco_b5:15:f0 (00:04:c1:b5:15:f0), Dst: Dell_17:ea:82 (00:14:22:17:ea:82) Destination: Dell_17:ea:82 (00:14:22:17:ea:82) Source: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Type: IP (0x0800) Internet Protocol, Src: 172.16.101.251 (172.16.101.251), Dst: 172.16.101.10 (172.16.101.10) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x68 (DSCP 0x1a: Assured Forwarding 31; ECN: 0x00) 0110 10.. = Differentiated Services Codepoint: Assured Forwarding 31 (0x1a) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 455 Identification: 0x01f8 (504) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 249 Protocol: UDP (0x11) Header checksum: 0x1e63 [correct] Source: 172.16.101.251 (172.16.101.251) Destination: 172.16.101.10 (172.16.101.10) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 435 Checksum: 0x6c94 [correct] Session Initiation Protocol Request-Line: REGISTER sip:172.16.101.10 SIP/2.0 Method: REGISTER Resent Packet: True Suspected resend of frame: 34 Message Header Via: SIP/2.0/UDP 192.168.10.52:5060;branch=z9hG4bK9749dfca14c0ffb2 From: 23456 <sip:23456@172.16.101.10;user=phone>;tag=2971075767 SIP Display info: 23456 SIP from address: sip:23456@172.16.101.10 SIP tag: 2971075767 To: 23456 <sip:23456@172.16.101.10;user=phone> SIP Display info: 23456 SIP to address: sip:23456@172.16.101.10 Call-ID: 3431569068(a)192.168.10.52 CSeq: 1 REGISTER Contact: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp>;expires=300 Contact Binding: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp>;expires=300 URI: 23456 <sip:23456@192.168.10.52:5060;user=phone;transport=udp> SIP Display info: 23456 SIP contact address: sip:23456@192.168.10.52:5060 User-Agent: Cisco ATA 186 v3.2.1 atasip (050616A) Content-Length: 0 No. Time Source Destination Protocol Info 533 152.674292 172.16.101.10 172.16.101.251 SIP Status: 401 Unauthorized (0 bindings) Frame 533 (432 bytes on wire, 432 bytes captured) Arrival Time: Nov 3, 2005 11:32:02.281970000 Time delta from previous packet: 0.001347000 seconds Time since reference or first frame: 152.674292000 seconds Frame Number: 533 Packet Length: 432 bytes Capture Length: 432 bytes Protocols in frame: eth:ip:udp:sip Ethernet II, Src: Dell_17:ea:82 (00:14:22:17:ea:82), Dst: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Destination: Cisco_b5:15:f0 (00:04:c1:b5:15:f0) Source: Dell_17:ea:82 (00:14:22:17:ea:82) Type: IP (0x0800) Internet Protocol, Src: 172.16.101.10 (172.16.101.10), Dst: 172.16.101.251 (172.16.101.251) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x60 (DSCP 0x18: Class Selector 3; ECN: 0x00) 0110 00.. = Differentiated Services Codepoint: Class Selector 3 (0x18) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 418 Identification: 0x0000 (0) Flags: 0x04 (Don't Fragment) 0... = Reserved bit: Not set .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x9988 [correct] Source: 172.16.101.10 (172.16.101.10) Destination: 172.16.101.251 (172.16.101.251) User Datagram Protocol, Src Port: 53492 (53492), Dst Port: 5060 (5060) Source port: 53492 (53492) Destination port: 5060 (5060) Length: 398 Checksum: 0x0ad1 [correct] Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 Resent Packet: True Suspected resend of frame: 480 Message Header Via: SIP/2.0/UDP 192.168.10.52:5060;received=172.16.101.251;branch=z9hG4bK9749dfca14c0ffb2 Call-ID: 3431569068(a)192.168.10.52 From: 23456 <sip:23456@172.16.101.10;user=phone>;tag=2971075767 SIP Display info: 23456 SIP from address: sip:23456@172.16.101.10 SIP tag: 2971075767 To: 23456 <sip:23456@172.16.101.10;user=phone> SIP Display info: 23456 SIP to address: sip:23456@172.16.101.10 CSeq: 1 REGISTER WWW-Authenticate: DIGEST realm="CISCO", nonce="4369a82d", qop="auth", algorithm=MD5 Content-Length: 0 Ritesh Jalan

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[Serusers] Anyone can help me with sems
by Antonio Membrives 03 Nov '05

03 Nov '05

The scenario are ser 0.9.3 + sems 0.9.0 + mediaproxy + mysql, al instances are in the same machine The code in my ser.cfg if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ) { acc_db_request("Llamada perdida","missed_calls"); if(!t_write_req("/tmp/am_fifo","voicemail")){ t_reply("500","Error contacting sems voicemail"); }; #t_reply("404", "Usuario no conectado"); break; }; } else if(method=="BYE" || method == "CANCEL"){ if(!t_write_req("/tmp/am_fifo","bye")){ t_reply("500","Error contacting sems"); }; }; sl_send_reply("404", "Usuario no encontrado"); break; setflag(3); when i call a offline user, the softphone make a call but I don't ear any message or received any notification, anyone can help me? this is /var/log/messages Nov 3 11:04:56 dns2 ser[23838]: Maxfwd module- initializing Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is initially 110592 Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is finally 262142 Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is initially 110592 Nov 3 11:04:56 dns2 ser[23838]: INFO: udp_init: SO_RCVBUF is finally 262142 Nov 3 11:04:56 dns2 ser[23847]: INFO: fifo process starting: 23847 Nov 3 11:04:56 dns2 ser[23847]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... Nov 3 11:05:07 dns2 ser[23848]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:08 dns2 ser[23849]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:10 dns2 ser[23851]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:14 dns2 ser[23849]: WARNING: t_reply: ACKs are not replied Nov 3 11:05:23 dns2 mediaproxy[3158]: command delete 3924182bdc181a37 info= Nov 3 11:05:23 dns2 mediaproxy[3158]: command execution time: 0.20 ms

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[Serusers] Accounting
by harry gaillac 03 Nov '05

03 Nov '05

Hello, Does acc module allow these requests to a radius server according to rfc2866 ? A Talk Start information is sent when SIP Server receives 200 OK for a INVITE request. A TAlK Stop information when SIP Server receives BYE. How may i configure it ? if acc module can't calculate acct-session-time why does this module provide radius support ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] SEMS + SER performance question
by Ryan Pagquil 03 Nov '05

03 Nov '05

Hi, How many simultaneous connection can SER + SEMS can handle? i deployed SER + SEMS in Pentium 2 300MHz with 128MB memory, how many connections can this handle? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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[Serusers] Ser+Media proxy
by KM 03 Nov '05

03 Nov '05

I have installed SER 0.9.3.2 and media proxy 1.2.1 on the same machine. The machine is on public network. Calls work ok except when I have a UA on a LAN behind NAT calling to a UA that is in the same LAN. In this case I get : +----------+-------------------------------+--------------------------+| username | contact | received |+----------+-------------------------------+--------------------------+| 22000001 | sip:22000001@192.168.1.7:5060 | sip:xxx.xxx.xxx.xxx:5060 || 22000002 | sip:22000002@192.168.1.6:5060 | sip:xxx.xxx.xxx.xxx:2557 |+----------+-------------------------------+--------------------------+The number that takes up port 5060 on the NAT server,which is 22000001 can call the other UA but not the other way round.22000002 can only receive calls.Please help me,KM

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[Serusers] about rtp mixing
by Jorge C 03 Nov '05

03 Nov '05

Hello everybody, I have a question related with conference servers. Do you now how is done the mixing in a Conference? Specifically, how input RTP streams are mixed into output string? Should I need to mix in a linear way, interleaving samples of different users? Thanks in advance!!

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sr-users November 2005