sr-users December 2004

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184 participants
400 discussions

[Serusers] SER+Rtp proxy -> no voice going through

by Tareq Siraj

Hi all, I have found the following ser.cfg file for SER+RTP proxy combination in the mailing list. However, with this, no audio is going through. The call goes through, if i accept the call, no audio. I'm completely new to SER and SIP servers and still going through the manuals for the ser script. Any help will be appreciated. Thanks. - tareq # # $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) listen=192.168.1.21 # hostname matching an alias will satisfy the condition uri==myself". alias=mypc alias=mypc.localdomain # 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" # ------------------ module loading ---------------------------------- 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 modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc|auth_db|acc|group|msilo|uri","db_url","mysql://ser:heslo@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:heslo@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://login:password@localhost/ser") # -- voicemail params -- #modparam("group", "db_url","mysql://login:password@localhost/ser") # -- nathelper params -- modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("tm", "fr_inv_timer", 30 ) #modparam("tm", "fr_inv_timer", 8 ) #Explicitly set the socket used by rtpproxy #modparam("nathelpler", "rtpproxy_sock", "/var/run/rtpproxy.sock") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "-------------------------------------------\n"); log(1, "entering main loop\n"); if (nat_uac_test("2")) { log(1, "src address different than via header->NAT detected\n"); log(1, "force_rport and fix_nated_contact and setflag(5)\n"); #try NAT traversal, works only if the client is symmetrical force_rport(); fix_nated_contact(); append_hf("P-hint: fixed NAT contact for request\r\n"); # flag 5 indicates that incoming request is from NATed client setflag(5); }; if (method=="REGISTER") log(1, "REGISTER message received\n"); if (method=="INVITE") log(1, "INVITE message received\n"); if (method=="ACK") log(1, "ACK message received\n"); if (method=="BYE") log(1, "BYE message received\n"); if (method=="CANCEL") log(1, "CANCEL message received\n"); if (method=="SUBSCRIBE") log(1, "SUBSCRIBE message received\n"); if (method=="NOTIFY") log(1, "NOTIFY message received\n"); if (method=="OPTIONS") log(1, "OPTIONS message received\n"); if (method=="INFO") log(1, "INFO message received\n"); if (method=="MESSAGE") log(1, "MESSAGE message received\n"); if (method=="REFER") log(1, "REFER message received\n"); # 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) { #if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # loose-route processing if (loose_route()) { log(1, "loose_route processing\n"); t_relay(); break; }; # create transaction state; abort if error occured # if ( !t_newtran()) { # sl_reply_error(); # break; # }; #new # now check if it's about PSTN destinations through our gateway; # note that 8.... is exempted for numerical non-gw destinations if (uri=~"^sip:0[0-9]*@.*") { log(1, "PSTN call...\n"); route(3); 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") { log(1, "analyzing REGISTER request\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("mypc", "subscriber")) { www_challenge("mypc", "0"); break; }; if (isflagset(5)) { #register from nated client, save nat_flag=6 #in location table setflag(6); }; if (!save("location")) { log(1, "save location error\n"); sl_reply_error(); }; break; }; lookup("aliases"); #mark transaction for voicemail if (is_user_in("Request-URI", "voicemail\n")) { log(1, "requested user is in voicemail group"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found log(1, "requested user not found\n"); route(4); break; }; }; #add failure route which should be performed if response code >=300 if (method=="INVITE" && isflagset(4)) { log(1, "invite for voicemail user->initiate failureroute[1]\n"); t_on_failure("1"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP route(1); } route[1]{ log(1, "-------------------------------------------\n"); log(1, "entering route[1] - relaying SIP message\n"); if ((isflagset(5)) || (isflagset(6))) { log(1, "at least one of the participants 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, set NATED-INVITE flag(7)"); force_rtp_proxy(); append_hf("P-hint: request forced to rtp proxy\r\n"); setflag(7); }; }; log(1, "relaying message ...\n"); if (!t_relay()) { log(1, "t_relay error occured\n"); sl_reply_error(); }; } # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { log(1, "-------------------------------------------\n"); log(1, "onreply_route[1] entered\n"); if (isflagset(6)) { log(1, "transaction was sent to a NATED client -> fix nated contact\n"); fix_nated_contact(); append_hf("P-hint: fixed NAT contact for response\r\n"); } if ( (status=~"100") ) { log(1, "status 100 received\n"); }; if ( (status=~"180") ) { log(1, "status 180 received\n"); }; if ( (status=~"202") ) { log(1, "status 202 received\n"); }; if ( (status=~"200" || status=~"183") ) { log(1, "status 2xx or 183"); if ( isflagset(7) ) { log(1, "marked(7) as NATED-INVITE -> force_rtp_proxy \n"); force_rtp_proxy(); append_hf("P-hint: response forced to rtp proxy\r\n"); }; }; } #new # logic for calls to the PSTN route[3] { # turn accounting on setflag(1); /* require all who call PSTN to be members of the "int" group; apply ACLs only to INVITEs -- we don't need to protect other requests, as they don't imply charges; also it could cause troubles when a call comes in via PSTN and goes to a party that can't authenticate (voicemail, other domain) -- BYEs would fail then; exempt Cisco gateway from authentication by IP address -- it does not support digest */ if (method=="INVITE" && (!src_ip==192.168.1.21)) { if (!proxy_authorize( "mypc" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "mypc" /* realm */, "0" /* no qop */ ); break; }; # let's check from=id ... avoids accounting confusion if(!is_user_in("credentials", "int")) { sl_send_reply("403", "NO PSTN Privileges..."); break; }; consume_credentials(); }; # INVITE to authorized PSTN # if you have passed through all the checks, let your call go to GW! force_rtp_proxy(); record_route(); t_on_reply("1"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) #t_relay_to_udp("212.17.35.184","5060"); t_relay_to_udp("192.168.1.1","5060"); } route[4]{ log(1, "-------------------------------------------\n"); log(1, "entering route[4] = requested user not online\n"); # 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(); }; }; # forward to voicemail now � rewritehostport("192.168.1.21:5060"); log(1, "forward to voicemail\n"); t_relay_to_udp("192.168.1.21", "5060"); } failure_route[1] { /* XX: note: unsafe if preloaded routes without username used */ log(1, "-------------------------------------------\n"); log(1, "failureroute[1] entered\"); revert_uri(); rewritehostport("192.168.1.21:5060"); append_branch(); t_relay_to_udp("192.168.1.21", "5060"); } __________________________________ Do you Yahoo!? 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19 years, 8 months

[Serusers] Dynamic Invite Timer w/AVPOPS

by Java Rockx

Hi All. I thought I read something a while ago about using avpops to dymanically change the INVITE timer - but I can't seem to find that thread. Can anyone tell me if this is possible? I'd like to have a shorter INVITE timer for SIP->SIP calls and a longer INVITE timer for SIP->PSTN calls. We see some 408 replies because a SIP phone is dialing a cell phone and get transfered to the voicemail of the cell phone user, but to prevent the 408 due to a slow voicemail system we have our INVITE timer currently set at 40 seconds which is rather annoying for SIP->SIP calls. I'm using -dev-21 Regards, Paul __________________________________ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250

19 years, 8 months

[Serusers] SER+Rtp proxy -> no voice going through

by Tareq Siraj

Hi all, I have found the following ser.cfg file for SER+RTP proxy combination in the mailing list. However, with this, no audio is going through. The call goes through, if i accept the call, no audio. I'm completely new to SER and SIP servers and still going through the manuals for the ser script. Any help will be appreciated. Thanks. # # $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) listen=192.168.1.21 # hostname matching an alias will satisfy the condition uri==myself". alias=mypc alias=mypc.localdomain # 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" # ------------------ module loading ---------------------------------- 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 modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc|auth_db|acc|group|msilo|uri","db_url","mysql://ser:heslo@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:heslo@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://login:password@localhost/ser") # -- voicemail params -- #modparam("group", "db_url","mysql://login:password@localhost/ser") # -- nathelper params -- modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("tm", "fr_inv_timer", 30 ) #modparam("tm", "fr_inv_timer", 8 ) #Explicitly set the socket used by rtpproxy #modparam("nathelpler", "rtpproxy_sock", "/var/run/rtpproxy.sock") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "-------------------------------------------\n"); log(1, "entering main loop\n"); if (nat_uac_test("2")) { log(1, "src address different than via header->NAT detected\n"); log(1, "force_rport and fix_nated_contact and setflag(5)\n"); #try NAT traversal, works only if the client is symmetrical force_rport(); fix_nated_contact(); append_hf("P-hint: fixed NAT contact for request\r\n"); # flag 5 indicates that incoming request is from NATed client setflag(5); }; if (method=="REGISTER") log(1, "REGISTER message received\n"); if (method=="INVITE") log(1, "INVITE message received\n"); if (method=="ACK") log(1, "ACK message received\n"); if (method=="BYE") log(1, "BYE message received\n"); if (method=="CANCEL") log(1, "CANCEL message received\n"); if (method=="SUBSCRIBE") log(1, "SUBSCRIBE message received\n"); if (method=="NOTIFY") log(1, "NOTIFY message received\n"); if (method=="OPTIONS") log(1, "OPTIONS message received\n"); if (method=="INFO") log(1, "INFO message received\n"); if (method=="MESSAGE") log(1, "MESSAGE message received\n"); if (method=="REFER") log(1, "REFER message received\n"); # 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) { #if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # loose-route processing if (loose_route()) { log(1, "loose_route processing\n"); t_relay(); break; }; # create transaction state; abort if error occured # if ( !t_newtran()) { # sl_reply_error(); # break; # }; #new # now check if it's about PSTN destinations through our gateway; # note that 8.... is exempted for numerical non-gw destinations if (uri=~"^sip:0[0-9]*@.*") { log(1, "PSTN call...\n"); route(3); 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") { log(1, "analyzing REGISTER request\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("mypc", "subscriber")) { www_challenge("mypc", "0"); break; }; if (isflagset(5)) { #register from nated client, save nat_flag=6 #in location table setflag(6); }; if (!save("location")) { log(1, "save location error\n"); sl_reply_error(); }; break; }; lookup("aliases"); #mark transaction for voicemail if (is_user_in("Request-URI", "voicemail\n")) { log(1, "requested user is in voicemail group"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found log(1, "requested user not found\n"); route(4); break; }; }; #add failure route which should be performed if response code >=300 if (method=="INVITE" && isflagset(4)) { log(1, "invite for voicemail user->initiate failureroute[1]\n"); t_on_failure("1"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP route(1); } route[1]{ log(1, "-------------------------------------------\n"); log(1, "entering route[1] - relaying SIP message\n"); if ((isflagset(5)) || (isflagset(6))) { log(1, "at least one of the participants 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, set NATED-INVITE flag(7)"); force_rtp_proxy(); append_hf("P-hint: request forced to rtp proxy\r\n"); setflag(7); }; }; log(1, "relaying message ...\n"); if (!t_relay()) { log(1, "t_relay error occured\n"); sl_reply_error(); }; } # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { log(1, "-------------------------------------------\n"); log(1, "onreply_route[1] entered\n"); if (isflagset(6)) { log(1, "transaction was sent to a NATED client -> fix nated contact\n"); fix_nated_contact(); append_hf("P-hint: fixed NAT contact for response\r\n"); } if ( (status=~"100") ) { log(1, "status 100 received\n"); }; if ( (status=~"180") ) { log(1, "status 180 received\n"); }; if ( (status=~"202") ) { log(1, "status 202 received\n"); }; if ( (status=~"200" || status=~"183") ) { log(1, "status 2xx or 183"); if ( isflagset(7) ) { log(1, "marked(7) as NATED-INVITE -> force_rtp_proxy \n"); force_rtp_proxy(); append_hf("P-hint: response forced to rtp proxy\r\n"); }; }; } #new # logic for calls to the PSTN route[3] { # turn accounting on setflag(1); /* require all who call PSTN to be members of the "int" group; apply ACLs only to INVITEs -- we don't need to protect other requests, as they don't imply charges; also it could cause troubles when a call comes in via PSTN and goes to a party that can't authenticate (voicemail, other domain) -- BYEs would fail then; exempt Cisco gateway from authentication by IP address -- it does not support digest */ if (method=="INVITE" && (!src_ip==192.168.1.21)) { if (!proxy_authorize( "mypc" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "mypc" /* realm */, "0" /* no qop */ ); break; }; # let's check from=id ... avoids accounting confusion if(!is_user_in("credentials", "int")) { sl_send_reply("403", "NO PSTN Privileges..."); break; }; consume_credentials(); }; # INVITE to authorized PSTN # if you have passed through all the checks, let your call go to GW! force_rtp_proxy(); record_route(); t_on_reply("1"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) #t_relay_to_udp("212.17.35.184","5060"); t_relay_to_udp("192.168.1.1","5060"); } route[4]{ log(1, "-------------------------------------------\n"); log(1, "entering route[4] = requested user not online\n"); # 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(); }; }; # forward to voicemail now � rewritehostport("192.168.1.21:5060"); log(1, "forward to voicemail\n"); t_relay_to_udp("192.168.1.21", "5060"); } failure_route[1] { /* XX: note: unsafe if preloaded routes without username used */ log(1, "-------------------------------------------\n"); log(1, "failureroute[1] entered\"); revert_uri(); rewritehostport("192.168.1.21:5060"); append_branch(); t_relay_to_udp("192.168.1.21", "5060"); } ===== - Tareq Never argue with an idiot: first (s)he will drag you down to his/her level, then (s)he will beat you with experience. __________________________________ Do you Yahoo!? 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19 years, 8 months

[Serusers] re-invite and nat proxy

by Richard

Hi, I ran into some issues with re-invite and nat. Ser uses "lookup" location to load the flag of a NATted callee. Because re-invite already knows the callee's contact, it uses loose-route. It doesn't call "lookup". The problem is that if the callee is behind NAT, ser can't tell. Since the ruri of re-invite is already in the location table, I wonder if we can have another function to find it in the location table and load the flag. Does it make sense? Thanks, Richard

19 years, 8 months

[Serusers] How Can I Rewrite The URI When Dialing 411 Directory Assistance

by Java Rockx

Hi All. When someone dials 411 I want to rewrite the request before sending it out to the PSTN gateway. For example if my SIP phone user name is 4071234567 and I dial 411 I want to send the request out to the PSTN gateway as 4075551212 for directory assistance so that the subscriber's __local__ directory assistance is dialed. How can I do this using -dev23? Regards, Paul __________________________________ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com

19 years, 8 months

[Serusers] Re: External Notify (ser+asterisk)

by Java Rockx

Sorry - I have not. The only thing I've worked on is Message Waiting Indicator (MWI) which is just a SIP NOTIFY message. Regards, Paul --- mohammad mirzaee <agent(a)rasatech.com> wrote: > Hi Paul; > > > > Have you worked on "sms voicemail notification" with Asterisk????????? > > > Appreciate your suggestions > mohammad > > > > ----- Original Message ----- > From: "Java Rockx" <javarockx(a)yahoo.com> > To: "mohammad mirzaee" <agent(a)rasatech.com>; <serusers(a)lists.iptel.org> > Sent: Tuesday, November 30, 2004 4:35 PM > Subject: Re: External Notify (ser+asterisk) > > > > Mohammad, > > > > The only reason to use sipsak an Asterisk server is because I don't know > of any other way to have > > Asterisk send sip messages directly to the SIP phones. > > > > For example, you call me and leave me a voicemail. When you hang up your > phone (after recording > > your message) Asterisk has to somehow now contact my SIP phone and send > the MWI NOTIFY message. > > The problem is that my phone never registered with Asterisk - it only > registered with SER. > > > > So the only way I know to get Asterisk to send a message to an unknown > target phone is to simply > > use sipsak to send the SIP NOTIFY message to SER. Once SER gets the > message it will send it to the > > SIP phone - providing that it can indeed find a contact location for it. > > > > Regards, > > Paul > > > > --- mohammad mirzaee <agent(a)rasatech.com> wrote: > > > > > Hi ALL; > > > > > > Hi Paul; > > > > > > > > > > > > > > > About Paul's email on SEP.19 with the subject " ingecting SIP NOTIFY > INTO SER FIFO" , I come up > > > with the following question: > > > > > > Why we use "sipsak" for sending "NOTIFY" message from Asterisk to Ser?? > Is it because of > > > Asterisk is not a "b2bua" > > > > > > > > > > > > > > > > > > Warnest Regards > > > mohammad > > > > > > > > > > __________________________________ > > Do you Yahoo!? > > All your favorites on one personal page - Try My Yahoo! > > http://my.yahoo.com > > > > > > __________________________________ Do you Yahoo!? All your favorites on one personal page � Try My Yahoo! http://my.yahoo.com

19 years, 8 months

Re: [Serusers] Serious problems with module mysql: 0.8.99-dev19

by msg

Hi Jan, Thanks for your reply. I neglected to state that the library did include the symbol: 'nm' reports: 00111fc4 D module_version Regards, Michael >>2. Compiled and linked against libmysqlclient.so.14.0 (built locally >> with '-g'): 'ser' loads the file, reports the >> following error and then exits (debug=9): >>0(9791) loading module /usr/local/lib/ser/modules/mysql.so.4.1.7.new >>0(9791) ERROR: no version info in module >> </usr/local/lib/ser/modules/mysql.so.4.1.7.new>: Can't open file > > > This is strange, there should be "module_version" symbol in the > library, try: > > nm mysql.so | grep module_version > > Jan. >

19 years, 8 months

[Serusers] FW: [Sip-implementors] Whitespace after value in SIP header field

by Martin Koenig

Hello, I think the white space is not legal in 3261. Please take a look at the message below. Best regards, Martin -----Original Message----- From: Attila Sipos [mailto:Attila.Sipos@vegastream.com] Sent: Tuesday, November 16, 2004 12:01 PM To: Martin Koenig; sip-implementors(a)cs.columbia.edu Subject: RE: [Sip-implementors] Whitespace after value in SIP header field Hi, One of the headers in the posting to the SERusers mailing list is: Max-Forwards: 8 . According to RFC3261 Max-Forwards is defined as: Max-Forwards = "Max-Forwards" HCOLON 1*DIGIT and DIGIT (from RFC2234) is: DIGIT = "0" / "1" / "2" / "3" / "4" / "5" / "6" / "7" / "8" / "9" So, you can have a space between the colon and the first digit but you're not allowed any spaces after or between any digits and you're certainly not allowed a '.' character. Regards, Attila Attila Sipos Software Engineer http://www.vegastream.com/ > -----Original Message----- > From: sip-implementors-bounces(a)cs.columbia.edu > [mailto:sip-implementors-bounces@cs.columbia.edu]On Behalf Of Martin > Koenig > Sent: 16 November 2004 09:56 > To: sip-implementors(a)cs.columbia.edu > Subject: [Sip-implementors] Whitespace after value in SIP header field > > > Hello, > > please check the following posting to the SERusers mailinglist: > > http://lists.iptel.org/pipermail/serusers/2004-September/011580.html > > What do you think? Is the white space after the header field > value according > to 3261 or not? > > Best regards, > Martin > > _______________________________________________ > Sip-implementors mailing list > Sip-implementors(a)cs.columbia.edu > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors >

19 years, 8 months

[Serusers] does IVR will work in the redhat versions of linux

by voip technocrat

hello freinds, this is my second attempt to explore the possibility of useing the IVR with the SER in redhat linux 9.0 version. In the first attempt i could not do the this thing after a series of mails and responses to and from the list. At last the rafhel suggested to use the fedora he suspects that the libraries used by the ivr (i.e sipums) are not available in the redhat versions of the linux where iam compiling. what i want from the list is does any body got the success in doing so with configurations. my version of ser is ser ****************************************************** version: ser 0.8.12-1rc6 (i386/linux) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MM AP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SI ZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.168.4.3 2004/06/28 15:41:21 andrei Exp $ main.c compiled on 13:56:51 Nov 26 2004 with gcc 3.2 ****************************************************** you'r help will be highly appreciated with regards voip technocrat __________________________________ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.com

19 years, 8 months

[Serusers] relaying the public ip also through rtpproxy

by voip technocrat

hello friends, i want to relay all the rtp traffic through my rtpproxy i.e irrespective of is it in public ip or in private ip so what guidelines do i need to follow to achive this this is my current ser.cfg my ultimate aim is that all the rtpptrafic must go from the rtpproxy ip only ******************************************************* # ----------- global configuration parameters ------------------------ debug=8 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) 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 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") { save("location"); break; }; lookup("aliases"); # native SIP destinations are handled using our USRLOC DB 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(); }; } ******************************************************* with regards voip technocrat __________________________________ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250

19 years, 8 months

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sr-users December 2004