sr-users December 2004

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184 participants
400 discussions

[Serusers] Ser dead + mediaproxy
by Bui Thai Thinh 30 Dec '04

30 Dec '04

Hi All. I have a problem with SER when i use mediaproxy. I install SER in a computer(Linux Fedora 2) with ip 192.168.1.37. System run properly. I install more mediaproxy into system SER because i want to solve problem NAT.I had configured my config file. I begin test system mediaproxy : I have 2 GranStream BT100: BT1 :192.168.1.33 Phone :111 BT2 :192.168.1.35 Phone :333 When i make a call from 333 to 111 then display following: [root@dhcppc4 root]# tail -f /var/log/messages Dec 30 11:59:55 dhcppc4 proxydispatcher[3292]: command delete 57d2c31487b829d9(a)192.168.1.33 info= Dec 30 11:59:55 dhcppc4 proxydispatcher[3292]: command execution time: 0.77 ms Dec 30 11:59:55 dhcppc4 proxydispatcher[3292]: command request 57d2c31487b829d9(a)192.168.1.33 192.168.1.33:5004:audio 192.168.1.33 192.168.1.37 local 192.168.1.35 remote Grandstream=20BT100=201.0.5.16 info=from:111@192.168.1.37:5060,to:333@192.168.1.37:5060,fromtag:41cbda14a0e4af53,totag: Dec 30 11:59:55 dhcppc4 proxydispatcher[3292]: domain 192.168.1.37 doesn't define any mediaproxy. Dec 30 11:59:55 dhcppc4 proxydispatcher[3292]: will use default mediaproxy for this call. Dec 30 11:59:55 dhcppc4 mediaproxy[3296]: command request 57d2c31487b829d9(a)192.168.1.33 192.168.1.33:5004:audio 192.168.1.33 192.168.1.37 local 192.168.1.35 remote Grandstream=20BT100=201.0.5.16 info=from:111@192.168.1.37:5060,to:333@192.168.1.37:5060,fromtag:41cbda14a0e4af53,totag:,dispatcher Dec 30 11:59:55 dhcppc4 mediaproxy[3296]: session 57d2c31487b829d9(a)192.168.1.33: started. listening on 192.168.1.37:35022 Dec 30 11:59:55 dhcppc4 mediaproxy[3296]: command execution time: 4.85 ms Dec 30 11:59:55 dhcppc4 proxydispatcher[3292]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '192.168.1.37 35022' Dec 30 11:59:55 dhcppc4 proxydispatcher[3292]: command execution time: 246.63 ms Dec 30 12:00:17 dhcppc4 mediaproxy[3296]: command status Dec 30 12:00:17 dhcppc4 mediaproxy[3296]: command execution time: 0.48 ms Dec 30 12:00:55 dhcppc4 mediaproxy[3296]: session 57d2c31487b829d9(a)192.168.1.33: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Dec 30 12:00:55 dhcppc4 mediaproxy[3296]: session 57d2c31487b829d9(a)192.168.1.33: ended (did timeout). I see no packets throught mediaproxy and Ser dead.And when i do statement ./sessions.py then display : Caller Via Called Status Duration Codec Type Traffic --------------------------------------------------------------------------------------- ?.?.?.?:? - 192.168.1.37:35022 - ?.?.?.?:? inactive 0'22" Unknown Audio 0/0/0 I don't know to solve it. Can anybody help me. Thanks a lot. __________________________________ Do you Yahoo!? Send a seasonal email greeting and help others. Do good. http://celebrity.mail.yahoo.com

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[Serusers] How to install SER_FIFO
by SIPhyh 30 Dec '04

30 Dec '04

Hello folks, need help... I have got my ser running. But when I serctl ps, the process 'hangs' saying something to the effect that 'mkfifo: cannot create fifo...'. I took a look at ser.cfg and just to confirm that /tmp/ser_fifo exist, I did an slocate and found that there isn't any ser_fifo. Question: How do I install this ser_fifo? I had actually thought it would have been installed by default together with ser. Thanks in advance. Joe. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 12/28/2004

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[Serusers] Rewrite Contact header
by Robert Brown 30 Dec '04

30 Dec '04

Can someone give me some direction as to how I can get SER to functin as a basic B2BUA?...a document, how-to, etc? If that is not readily available then I wonder if there is some way that I can get SER to statefully rewrite the Contact header when it relays messages? The reason for this is that I want all of my UAs to interact only with/through SER and not directly with each other. Something like this: UA1 SER UA2 IPaddr1 IPaddr2 IPaddr3 | ----------> | | | INVITE | | | contact: name@IPaddr1 | | | | ----------> | | | INVITE | | | contact: name@IPaddr2 | ..... ..... ..... | | <---------- | | | BYE | | | ruri: name@IPaddr2 | | <---------- | | | BYE | | | ruri: name@IPaddr1 | | | | | Without rewriting the contact header I get the following which is undesireable: UA1 SER UA2 IPaddr1 IPaddr2 IPaddr3 | ----------> | | | INVITE | | | contact: name@IPaddr1 | | | | ----------> | | | INVITE | | | contact: name@IPaddr1 | ..... ..... ..... | <---------------------------------- | | BYE | | ruri: name@IPaddr1 | Thanks to all for contributing to a really nice open source system. Regards, Robert Brown

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[Serusers] Ser, Cisco and NAT
by Victor Chan 30 Dec '04

30 Dec '04

I have some problem about Cisco and NAT device. I have SER and rtpproxy running on a server. Call is made from a SIP endpoint device to Cisco 5300 gateway via SER. But voice can't hear at my SIP device. I try point to point call without going to Cisco gateway, both side can hear. Please give advise to solve it. Thanks a lot. The following is my ser.cfg # # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ # # simple quick-start config script including nathelper support # This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------ debug= 7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) # Uncomment these lines to enter debugging mode #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" # ------------------ 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" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- 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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic 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; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request log(1, "loose_route processing\n"); append_hf("P-hint: rr-enforced\r\n"); t_relay(); 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") { save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if ( uri=~"^sip:9[0-9]*@5300_IP" ) { log(1, "SER22: matched - forward to GW\n"); fix_nated_sdp("2"); force_rtp_proxy("5300_IP"); rewritehostport ("5300_IP:5060"); t_relay_to_udp("5300_IP","5060"); forward(5300_IP, 5060); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if(!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } _________________________________________________________________ 「碧咸嫂」點樣學外語? http://go.msnserver.com/HK/46164.asp

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[Serusers] I am newcome , I can not login my sip server that I build?
by Zhaomin 30 Dec '04

30 Dec '04

I am newcome ,I can not login my sip server(ser0.8.14) that I build? I run the commamd serctl ping localhost echo me 400.Where is failt ?Anyones can help me

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[Serusers] Calls bet/ 2 sip phones and forward Asterisk
by Chris HARIGA 29 Dec '04

29 Dec '04

Hi, I install ser and I try to use it with asterisk. I setup two ser accounts and I register 2 ip phones (no problem with registration). The problem is when I try to place calls between ip phones, 105 and 106. In next step I try to forward all calls to my Asterisk box and I get the same result :-( Thank you, Chris HARIGA I attach my ser.cfg bellow: 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" # ------------------ 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/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- 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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # modparam("nathelper", "natping_interval", 10) # # ------------------------- request routing logic ------------------- 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; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); 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") { if (!www_authorize("160.79.172.149", "subscriber")) { www_challenge("160.79.172.149", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; if(method=="INVITE"){ if (uri=~"^sip:1[0-9]{10}@*") { log(1,"Forwarding to Asterisk\n"); rewritehostport("sipny.no-ip.info: 5060"); t_relay(); break; } } }

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[Serusers] Calls bet/ 2 sip phones and forward Asterisk
by Chris HARIGA 29 Dec '04

29 Dec '04

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[Serusers] I do not see de Acct-Stop records ...
by Lucas Aimaretto 29 Dec '04

29 Dec '04

Hi list. I have this problem. There's something that's not working, because, when ever I set up a call, I receive both messages INVITE and BYE instantly. I think the correct way would it be to send a Start messages (Acct-Status-Type = Start) upon INVITEs and a Stop messages (Acct-Status-Type = Stop) upon BYEs. But this is not happening. Look ... radrecv: Accounting Request from host c0a801fd code=4, id=205, length=270 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Invite User-Name = "1992001(a)192.168.1.253" Calling-Station-Id = "sip:1992001@192.168.1.253:5060;user=phone" Called-Station-Id = "sip:1992005@192.168.1.253:5060;user=phone" Sip-Translated-Request-URI = "sip:1992005@192.168.1.178:11005" Acct-Session-Id = "6dfb6640367b1ab7(a)192.168.1.113" Sip-To-Tag = "54061171" Sip-From-Tag = "39eeac148f846cb9" Sip-CSeq = "1544" NAS-IP-Address = 192.168.1.253 NAS-Port-Id = 5060 Acct-Delay-Time = 0 Sending Accounting Ack of id 205 to c0a801fd (nas linux) radrecv: Accounting Request from host c0a801fd code=4, id=206, length=270 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Bye User-Name = "1992001(a)192.168.1.253" Calling-Station-Id = "sip:1992001@192.168.1.253:5060;user=phone" Called-Station-Id = "sip:1992005@192.168.1.253:5060;user=phone" Sip-Translated-Request-URI = "sip:1992005@192.168.1.178:11005" Acct-Session-Id = "6dfb6640367b1ab7(a)192.168.1.113" Sip-To-Tag = "54061171" Sip-From-Tag = "39eeac148f846cb9" Sip-CSeq = "1544" NAS-IP-Address = 192.168.1.253 NAS-Port-Id = 5060 Acct-Delay-Time = 0 Sending Accounting Ack of id 206 to c0a801fd (nas linux) Do you see ? ...Both outputs show an Acct-Status-Type = Start, but first output refers to an INVITE and second output refers to BYE message. I don't think this is correct, isn't it ? Any hint ? here goes my ser.cfg ... [...] # -- RADIUS ACC -- modparam("acc", "radius_config", "/etc/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_level", 1) modparam("acc", "service_type", 15) [...] if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { # Lo siguiente es para auth con RADIUS if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; if (method=="INVITE") { setflag(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; t_relay(); [...] Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004

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[Serusers] Greeting, and Request for help on incoming "Strip" Command
by Ian Bonham 29 Dec '04

29 Dec '04

Hello Everyone, I've just joined the list having started using SER and would first like to say Hi to all the subscribers. I'm having a problem with moving calls between SER servers which I hope someone can offer some advice on. I'm running my own SER system, which is used to make calls between local users no problems. We are also partnered with another company who provide a POTS gateway for making calls externally. At the moment my setup works perfectly placing calls between extensions on the same domain, and I have got the routing working fine to allow users to make outgoing calls onto the POTS network. My problem arises when someone on POTS makes a call into our domain. A phone may be registered on my domain as 123456(a)sip.sft.net, but the call comes in from our partners in the UK. The call is routed to our domain fine, as I have seen running an Ngrep on port 5060. However the calls arrive with a prefix of 197544, then our internal number 123456. They also arrive with the domain www.partner.net on the TO field. Because the TO field now reads 197544123456(a)www.partner.net, as soon as it hits my domain it gets lost as I can't direct it to 123456(a)sip.sft.net. I've tried changing settings in ser.conf but I can't seem to find the right configuration or location to make it strip the first 6 numbers (197544) and change the domain from www.partner.net to sip.sft.net before I try to send the call to the destination on our domain. I am guessing that it is ser.conf I should be changing, but I am unsure as to where I should be making changes and exactly what changes I should be making to ensure it is only incoming calls from the www.partner.net domain that are affected. If anyone can offer some advice on how I can achieve this I'd be really grateful. Am I looking in the right place? I have RTFM, but am still lost on this one as I'm quite new to SER. Many thanks in advance, and wishing everyone a Happy New Year, Ian Bonham

2 1

[Serusers] SMS Callback
by Pavel Siderov 29 Dec '04

29 Dec '04

Hi guys, I'm trying to make sms callback with ser. Somebody tried this scenario? Thanks, Pavel Siderov

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sr-users December 2004