sr-users May 2004

sr-users@lists.kamailio.org

154 participants
333 discussions

Re: [Serusers] asymmetric
by Adrian Georgescu 28 May '04

28 May '04

It is a regular expression matching either User-Agent or Server field An example is supplied in the sample file. >>Hello list, >>Does somebody know the format of the address that you can put in rtp-asymmetric-clients and clients.xt that comes with mediaproxy to >>identify your asymmetric endpoint? Thanks, Tjapko.

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[Serusers] Call transfer
by Gregory D. Burns 28 May '04

28 May '04

Hello Group, I have a problem I hope I can solve by using SER. Here it is. 1. Person A calls person B Person A ---[PBX]-->[SER]--> VoIP --> [SIP Gateway] --> Telco --> Person B 2. Person A calls person C and second line. ( Person B is on Hold) Person A ---[PBX]-->[SER]--> VoIP --> [SIP Gateway] --> Telco --> Person C 3. Now we want to be about to Connect person B and person C, with out Person A needing to be on the line. Person B --> Telco --> [SIP Gateway] --> VoIP --> [SER] -->VoIP --> [SIP Gateway] --> Telco --> Person C It does not sound that hard, but I could not think of a way to make it work. Does any one have any ideas? -Greg VoIpin

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[Serusers] Using AS5300 as PSTN Gateway
by gc 28 May '04

28 May '04

I have a AS5300 setup as PSTN Gateway. It works fine with VOCAL. Now I'd like to connect SER to this PSTN gateway. I added following line to the default ser.cfg file: if (uri=~"^sip:1") { log(1, "Forwarding to PSTN\n"j); forward(189.101.110.132, 5060); break; }; This will allow any dialed number starting with 1 being forward to PSTN gateway. But it always give me busy signal. Can anybody tell me what's wrong? Here is my whole ser.cfg file: # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # 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 debug=3 */ 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" # 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" # ----------------- 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) # ------------------------- 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; }; # 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") { # Uncomment this if you want to use digest authentication if (!www_authorize("vocal0", "subscriber")) { www_challenge("vocal0", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; #Handle PSTN calls. if (uri=~"^sip:1") { log(1,"Forwarding to PSTN\n"); forward(189.101.110.132, 5060); 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(); }; }

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[Serusers] SEMS & mediaproxy
by Ezequiel Colombo 28 May '04

28 May '04

I have problems with sems and SER+mediaproxy When i send a call to sems for voicemail the SDP content received by sems have the c= line with the private address of caller. Before send the call to sems my SER use mediaproxy to fix nat problems, but sems dont receive the SDP changed and the RTP generated by SEMS is sent to the wrong ip address. -- on my ser.cfg -- if (method=="INVITE") { log(1,"ROUTE_0: INVITE METHOD\n"); if (!search("^Record-Route:")) { log(1,"ROUTE_0: Use Mediaproxy"); use_media_proxy(); }; ... log(1,"ROUTE_1: ARCOTEL-TO-ARCOTEL "); lookup("aliases"); if (!lookup("location")) { log(1, "ROUTE_1: USER OFFLINE OR INVALID\n"); route(8); break; }; ... route[8] { log(1,"ROUTE_8: VOICEMAIL"); if (!t_newtran()) { sl_reply_error(); break; }; use_media_proxy(); t_reply("183","Conectandose"); if(!vm("/tmp/am_fifo","voicemail")){ log(1,"could not contact voicemail\n"); t_reply("500","could not contact voicemail"); }; break; } -- on my ser.cfg -- -- on syslog file -- May 26 10:13:17 billing /usr/sbin/ser[9814]: ROUTE_1: ARCOTEL-TO-ARCOTEL May 26 10:13:17 billing /usr/sbin/ser[9814]: ROUTE_1: USER OFFLINE OR INVALID May 26 10:13:17 billing /usr/sbin/ser[9814]: ROUTE_8: VOICEMAIL May 26 10:13:17 billing proxydispatcher[9919]: command request 3B48F4D2-CB2B-42B2-80AD-F5064BC02868(a)10.10.0.29 10.10.0.29:800 0 200.80.35.6 arcotel.net local arcotel.net local X-Lite=20release=201103a flags= May 26 10:13:17 billing proxydispatcher[9919]: domain arcotel.net doesn't define any mediaproxy. May 26 10:13:17 billing proxydispatcher[9919]: will use default mediaproxy for this call. May 26 10:13:17 billing mediaproxy[25721]: command request 3B48F4D2-CB2B-42B2-80AD-F5064BC02868(a)10.10.0.29 10.10.0.29:8000 20 0.80.35.6 arcotel.net local arcotel.net local X-Lite=20release=201103a flags= May 26 10:13:17 billing proxydispatcher[9919]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'public.ip.proxy 35062 May 26 10:13:17 billing /usr/sbin/ser[9812]: ROUTE_0: Fixing Contact May 26 10:13:17 billing /usr/sbin/ser[9812]: ACK May 26 10:13:17 billing /usr/sbin/ser[9812]: END ROUTE_0 - RELAY May 26 10:13:27 billing /usr/sbin/ser[9839]: ROUTE_0: Fixing Contact May 26 10:13:27 billing /usr/sbin/ser[9839]: BYE -- on syslog file -- -- on SEMS trace -- ... (9780) DEBUG: execute (AmServer.cpp:277): body: `v=0 o=ecolombo 2205701 2205771 IN IP4 10.10.0.29 s=X-Lite c=IN IP4 10.10.0.29 t=0 0 m=audio 8000 RTP/AVP 3 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ... (9920) DEBUG: setRAddr (AmRtpStream.cpp:333): RTP remote address set to 10.10.0.29:8000 (9920) DEBUG: negotiate (AmSession.cpp:150): Sending Rtp data to 10.10.0.29/8000 ... -- on SEMS trace -- Thanks Ezequiel Colombo

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[Serusers] asymmetric
by Tjapko Smits 28 May '04

28 May '04

Hello list, Does somebody know the format of the address that you can put in rtp-asymmetric-clients and clients.xt that comes with mediaproxy to identify your asymmetric endpoint? Thanks, Tjapko. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.691 / Virus Database: 452 - Release Date: 26/05/2004

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[Serusers] Announcements to PSTN network, without charging.
by Atle Samuelsen 28 May '04

28 May '04

Hi guys (And Hopfully some girls..) My senario today is that I need a announcement (404) when a PSTN user dials in to me. Right now, Im running a strait forward setup with sems annoucements module etc, Tho the thing I dont know how to do is to make the annoucements free. If I should follow the cisco site, it says to send messages without "PI" (reff : http://www.cisco.com/en/US/products/sw/iosswrel/ps5013/products_feature_gui… ) Is this possible with a ser/sems setup? And if so, how can I do this / what do I need to do this. Thanks in advance for the help ! - Atle

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Re: [Serusers] public ip - private ip communication problem
by Jan Janak 28 May '04

28 May '04

Please always CC the mailing list. You can use any RFC3261 compliant user agent, for example Cisco, Grandstream, Sipura, X-lite, and others. Regarding your configuration file, I did not review that, so I do not know. Jan. On 28-05 08:42, varala ramakanth wrote: > hai janak, > > according to you the softphone is giveng the error > > so which is the best softphone to use with our ser > > software . > > does my ser.cfg is enough to establish the session > > between private to public ips > > if possible please guide me with the sample config > file > > > with regards > rama kanth > > > --- Jan Janak <jan(a)iptel.org> wrote: > > There is an parse error in the logs you sent us, > > some Route header field > > is missing < character, unfortunately the dumps do > > not show the message. > > Locate the source of error and report it to the > > manufacturer. > > > > Jan. > > > > On 26-05 22:49, varala ramakanth wrote: > > > 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 > > > > > > === message truncated === > > > > > > __________________________________ > Do you Yahoo!? > Friends. Fun. Try the all-new Yahoo! Messenger. > http://messenger.yahoo.com/

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[Serusers] Question using serctl add user
by gc 28 May '04

28 May '04

When I use serctl to add new user, it prompt for password. When I supply the root password for mysql, it tell me access denied for user:ser@localhost. What password should I use? Do I have to first create an mysql user as ser@localhost? Gary

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[Serusers] SER & Asterisk in a non-routed environment
by Lars 28 May '04

28 May '04

Hi serusers, after spending 4 days trying to figure out how to set up things using SER I am now hoping for help. The problem is as follows: i have a core network (say 192.168.0.0/24) in which the asterisk (192.168.0.99) resides. i have a users network (say 192.168.10.0/24) in which I (the user, x-lite) reside. Theres a gw between those to networks with addresses 192.168.0.10 and 192.168.10.1. The big problem: This gateway is not allowed to forward packets. It does usermode port-forwarding for required ports, but it has no default route and /proc/sys/net/ipv4/ip_forward is set to 0. The asterisk is working well and i now wanted to be able to place calls to other users (currently one directly connected grandstream) through the asterisk. First i check out siproxd which almost immediately worked as desired, but i realized, that as soon as the 192.168.10.0 network will be populated with more users, i don't want the inter-user calls to appear on the asterisk. That's where SER comes in. I want it to sit on the gw-box and handle request in the users network by itself, but forward requests it cannot handle (e.g. pstn) to the asterisk by pretending to be the user himself, as siproxd does. Especially i think therefor a user must register at the asterisk server through SER which also should notice where to find him using usrloc. I played around with nethelper/rtpproxy but could not even establish a sip session, not to mention rtp. I somehow don't understand the way ser works, and should handle meet this kind of requirement, so my question would be: Is 'ser' the tool I'm looking for? And if 'yes', how would it basically have to be configured to do what i want. For example one problem seems to be, that it forwards packets to the * server from it's 192.168.10.1 address which the * box will never know..... Thanks a lot Lars

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[Serusers] problem in testing with radius server
by varala ramakanth 28 May '04

28 May '04

hello friends, i have configured the freeradiusserver as instructed in radius how to but when performing the 3.2 step in that guide it gives the error as radclient:No token read where we expected an attribute name so what might be the error ser and radisus server and radclient are running in the same host so what changes i need to follow in configuration file with regards rama kanth __________________________________ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/

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sr-users May 2004