sr-users August 2004

sr-users@lists.kamailio.org

181 participants
424 discussions

by Terry Wilson

We are trying to use SER as a NAT traversal solution in front of our Asterisk PSTN gateways and our Asterisk Feature Server. Currently, the feature server is doing all of the dialplan processing. Because of this we can have user configurable dial times, unavailability numbers, forward on busy/unavailable numbers, auto-attendants, 7-digit dialing, etc. Of course, my life would be simple if Asterisk acted more like a SIP proxy server and the dialplan could be executed somewhat like an SER routing script. Unfortunately, any call that comes from a UA to asterisk creates one call, then asterisk creates another call to the other end of the conversation (other SIP device, PSTN number, whatever), then issues reinvites to the two ends of the conversation to tell them to send RTP to each other. So what I need, is an SER script that will intelligently handle NAT traversal for UA to UA calls, UA to PSTN calls, and PSTN to UA calls (all going through the Asterisk Dialplan so we can keep all of the end user features) including the damn re-invites. Anybody had to do this before? We are quite willing to pay you for your knowledge if you want to do some design/consulting/coding work for a solution to our problem. Thanks.

19 years, 10 months

[Serusers] trying to USE SER for a contact centre application

by Neill Wilkinson

All, just started using SER (great experience so far!). I'm trying to us it for a contact centre application where I have a specific requirement that calls can only be terminated in one direction, from a Cisco AS53xx gateway, not a Cisco phone. Here's the set-up and desired result: User logs into the contact centre systems which initiates a call on the PSTN side of the AS53xx. This then creates a VOIP (SIP) call leg from a dial-peer on the gateway, via the SER proxy. The agents all have Cisco 7960 IP phones which register with the SER registrar. The INVITE is proxied via a simple location lookup in the SER route[0] config file to the appropriate handset. The agent answers the call and is "logged" in to the system. Now's the fun part. I want to prevent the SIP leg being torn-down by the agent either accidentally or on purpose hanging up the call from the handset. Here's the logic (pseudo code) I have though out: Trap the BYE from the handset using a statement something like: if (method=="BYE" and uri=~"sip:[0-9]+@*") { # Agent phone initiated the BYE so do something route(1); } else { #do something else - pass on the message I guess }; route[1] { # use exec to call an external function exec_msg ('script.file'); } The external script needs to OK the BYE message - I assume use the FIFO function to construct the OK message. Now the tricky part: how do I stop the called phone from hanging up, (I tried a simple 403 Forbidden response) but that didn't work. So I thought about the REFER method with the URL of the phone sending the BYE message- can I send a refer to the Gateway (AS5300) that still thinks the dialogue is open to the phone and "hopefully" get it to send a new INVITE dialogue to the phone? Or would I have to generate a new INVITE sequence from the SER proxy exec code? If its the latter, then I need access to the RTP port number on the AS53xx from the original INVITE message - and this implies access to the SDP payload - which I can't see from the SER script (unless anyone can tell me how?) Here's a picture of what I hope to achieve! [AS5300} [SER Proxy} [Cisco 7960] ----------------------------------------------->INVITE <----------------------------------------------TRYING INVITE -------------------------------> <---------------------------------TRYING <---------------------------------RINGING <----------------------------------------------RINGING <-----------------------------------OK <------------------------------------------------OK <==================RTP exchange =================> <-------------------------------- BYE OK ----------------------------------> <----------------------------------------------REFER (with URI of Cisco 7960) -------------------------------------------------->INVITE <-------------------------------------------------TRYING INVITE -----------------------------> <-------------------------------TRYING <-------------------------------RINGING <--------------------------------------------------RINGING <-------------------------------OK <-------------------------------------------------- OK <=================RTP Exchange ===================> Thanks in advance!!! All ideas Gratefully received! Neill....:o) Neill Wilkinson Senior Consultant Quortex Consultants Ltd. e: neill.wilkinson(a)quortex.com

19 years, 10 months

[Serusers] Extract "from" header

by Bob Bob

Hi, I'd like to extract the From URI from a request and use it to add a Call-Info header: From: <sip:test@test.com>;tag=fd29df3df47 -> Call-Info: <;purpose=icon">http://myserver.com/get.php?id=sip:test@test.com>;purpose=icon Can it be done with ser? There are a lot of functions but none of them seem to be able to extract headers and use the information in another header. Best regards, Chris --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages!

19 years, 10 months

[Serusers] VoiceMail problem

by Veselin Iordanov

I found some example configuration someware here i the mailing lisy but throw some strange error .. here is the configuration files 1. Conf file for the main ser server # # $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=no # (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) listen=80.72.85.178 port=5060 children=4 fifo="/tmp/ser_fifo" alias=testinfo.intersoftpro.com 127.0.0.1 80.72.85.178 # ------------------ 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 modparam("usrloc", "db_mode", 2) modparam("usrloc", "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://ser:heslo@localhost/ser") # -- voicemail params -- modparam("group", "db_url","mysql://serro:heslo@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 ) # ------------------------- 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; # }; # 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("80.72.85.178", "subscriber")) { www_challenge("80.72.85.178", "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"); # check if number beginns with 00, then convert it into a + if (uri=~"^sip:359[0-9]*@") { log(1, "00 International number detected\n"); # strip booth leading "0" strip(2); prefix("+"); }; # check if request uri begins with an internation phone number, if yes, try enum to resolve if (uri=~"^sip:\+[0-9]*@") { log(1, "+ International number detected\n"); if (!enum_query("voice")) { log(1, "ENUM query failed, try enum without voice prefix\n"); if (enum_query("")) { log(1, "ENUM without voice prefix also failed\n"); }; }; if (uri != myself) { log(1, "ENUM lookup revealed an external address, relaying..."); route(1); break; }; }; #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"); }; }; } 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("machtnix.ict.tuwien.ac.at:5060"); log(1, "forward to voicemail\n"); t_relay_to_udp("80.72.85.178", "5090"); #klaus #forward(machtnix.ict.tuwien.ac.at, 5060) } failure_route[1] { /* XX: note: unsafe if preloaded routes without username used */ log(1, "-------------------------------------------\n"); log(1, "failureroute[1] entered\"); revert_uri(); #rewritehostport("machtnix.ict.tuwien.ac.at:5060"); # append_branch(); t_relay_to_udp("machtnix.ict.tuwien.ac.at", "5060"); } 2. Configuration for the second instance of the ser server (for voice mail ( # # $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=no # (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) listen=80.72.85.178 port=5090 children=4 fifo="/tmp/ser_fifo" alias=testinfo.intersoftpro.com 127.0.0.1 80.72.85.178 # ------------------ 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" # load the voicemail module loadmodule "/usr/local/lib/ser/modules/vm.so" # ----------------- setting module-specific parameters --------------- modparam("voicemail", "db_url","mysql://ser:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "entering main loop\n"); 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"); # lookup("aliases"); # 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; }; # check changed from 0.8.11 to 0.8.12 # if (len_gt( max_len )) { if (msg:len > max_len) { sl_send_reply("513", "Message too big"); break; }; # if (!uri==myself) { # sl_send_reply("404", "not reponsible for host in request-uri"); # break; # }; # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE" || method=="REFER" || method=="CANCEL"){ log(1, "Entering voicemail processing\n"); log(1, "check if new transaction ... \n"); if(t_newtran()){ log(1, "... yes, new transaction\n"); t_reply("100","voicemail ser - Trying -- just wait a minute !"); t_reply("100","voicemail ser - Trying 2 -- just wait a minute !"); if(method=="INVITE" || method=="REFER"){ log("**************** vm start - begin ******************\n"); if( uri =~ "conference" ){ log(1, "incoming conference call\n"); if(!vm("/tmp/am_fifo","conference")){ log("could not contact conference server\n"); t_reply("500","could not contact conference server"); }; } else if( uri =~ "echo" ){ log(1, "incoming echo call\n"); if(!vm("/tmp/am_fifo","echo")){ log("could not contact echo\n"); t_reply("500","could not contact echo"); }; } else { log(1, "incoming voicemail call\n"); if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact voicemail\n"); t_reply("500","could not contact voicemail"); }; }; log("**************** vm start - end ******************\n"); break; }; if(method=="BYE" || method=="CANCEL"){ log("**************** vm end/refer - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the media server\n"); t_reply("500","could not contact the media server"); }; log("**************** vm end/refer - end ********************\n"); break; }; } else { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; }; # Voicemail specific configuration - end sl_send_reply("501", "method not understood here"); log(1, "method not understood here\n"); } 3. Configuration for sems server # $Id: sems.conf.sample,v 1.11.2.1 2004/01/27 15:08:28 rco Exp $ # # sems.conf.sample # # Sip Express Media Server (sems) # # sample configuration file # # # whitespaces (spaces and tabs) are ignored # comments start with a "#" and may be used inline # # example: option=value1, value2 # i like this option # ################################## # global parameters # ################################## # optional parameter: fork={yes|no} # # - specifies if sems should run in daemon mode (background) # (fork=no is the same as -E) fork=yes # optional parameter: stderr={yes|no} # # - debug mode: do not fork and log to stderr # (stderr=yes is the same as -E) stderr=no # optional parameter: loglevel={0|1|2|3} # # - sets log level (error=0, warning=1, info=2, debug=3) # (same as -D) loglevel=1 # optional parameter: fifo_name=<filename> # # - path and file name of our fifo file (same as -i) fifo_name=/tmp/am_fifo # optional parameter: ser_fifo_name=<filename> # # - path and file name of Ser's fifo file (same as -o) ser_fifo_name=/tmp/ser_fifo # optional parameter: plugin_path=<path> # # - sets the path to the plug-ins # - may be absolute or relative to CWD plugin_path=/usr/local/lib/sems/plug-in/ # optional parameter: smtp_server=<hostname> # # - sets address of smtp server smtp_server=localhost # optional parameter: smtp_port=<port> # # - sets port of smtp server smtp_port=25 ################################## # module specific parameters # ################################## # sample voicemail configuration (inline) config.voicemail=inline # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for # - the file to be played is determined the following way: # <announce_path>/<domainname>/<username>.wav # if this file is not available <announce_path>/<default_anounce> is used announce_path=/usr/local/lib/sems/audio/ # parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # parameter: max_record_time=<seconds> # # - maximum record time max_record_time=30 # parameter: accept_delay=<x> # # - delays accepting of the call for x seconds # - default value is 0 accept_delay=20 # end of configuration section # for voicemail module config.voicemail=end # sample announcement configuration (inline) config.announcement=inline # optional parameter: announce_path=<path> # # - sets the path where announce files are searched for announce_path=/usr/local/lib/sems/audio/ # parameter: default_announce=<filename> # # - sets the name of the default announce WAV file default_announce=default_en.wav # end of configuration section for announcement module config.announcement=end # sample isdngw module configuration (external file) # config.isdngw=/etc/isdngw.conf # sample ivr module configuration (inline) config.ivr=inline #parameter: python_script_path=<full path> python_script_path=/etc/ivr #parameter: python_script_file=<filename> python_script_file=example.py # end of configuration section for ivr module config.ivr=end # sample conference configuration (inline) config.conference=inline # parameter: default_announce=<filename> # # - sets the full pathed name of the default announce WAV file. # Will be played to lonely users. default_announce=/usr/local/lib/sems/audio/first_participant.wav # end of configuration section for conference module config.conference=end # example configuration for number reader config.number_reader=inline number_path=/usr/local/lib/sems/audio/ prolog_file=welcome_to_number_reader.wav epilog_file=thanks_calling_number_reader.wav # end of number_reader configuration config.number_reader=end # add more module configurations here (inline or external): # # config.mymodule=<filename> # or # config.mymodule=inline # ... # config.mymodule=end Error is in second instance of the ser server 0(19586) **************** vm start - end ****************** 0(19586) DEBUG:destroy_avp_list: destroing list (nil) 0(19586) receive_msg: cleaning up 2(19588) ERROR: fifo_server: command t_reply is not available 2(19588) ERROR: fifo_server: command must begin with :: ringing 2(19588) ERROR: fifo_server: command must begin with :: 00004C89728793F0 2(19588) ERROR: fifo_server: command must have at least 3 chars 2(19588) INFO: fifo_server: command empty 2(19588) ERROR: fifo_server: command t_reply is not available 2(19588) ERROR: fifo_server: command must have at least 3 chars 2(19588) ERROR: fifo_server: command must begin with :: 00004C89728793F0 2(19588) ERROR: fifo_server: command must begin with :: Content-Type: application/sdp 2(19588) ERROR: fifo_server: command must begin with :: v=0 2(19588) ERROR: fifo_server: command must begin with :: s=session 2(19588) ERROR: fifo_server: command must begin with :: t=0 0 2(19588) ERROR: fifo_server: command must begin with :: a=rtpmap:0 / 2(19588) INFO: fifo_server: command empty 2(19588) ERROR: fifo_server: command t_reply is not available 2(19588) ERROR: fifo_server: command must begin with :: could not send response. 2(19588) ERROR: fifo_server: command must begin with :: 00004C89728793F0 2(19588) ERROR: fifo_server: command must have at least 3 chars 2(19588) INFO: fifo_server: command empty 1(19587) DEBUG: timer routine:0,tl=0x402ecf9c next=(nil) 1(19587) DEBUG: FR_handler:stop retr. and send CANCEL (0x402ece60) 1(19587) ->>>>>>>>> T_code=100, new_code=408 actualy i think that i miss some modiule to load but i am not sure and dont have any fogest idea what to do .. 10x in advance Ps: sry about my bad english

19 years, 10 months

[Serusers] Voicemail wierdness

by Dave Bath

Hey all, Another wierdness for a sunny tuesday afternoon.... Have been using ser and * quite successfully so far, and am not into the stage of testing voicemail. Currently I am using * with mysql vm storage. I have a failure route which affter the fr_timer expires, does: failure_route[4] { # If we've got here, it means that a call timer has expired, and the user is in the # voicemail group. Therefore we're going to try and hand off the call to the Asterisk # voicemail system. revert_uri(); rewritehostport("asterisk.dev.inmarsat.com:5060"); log (1, "VOICEMAIL: VM-Divert --> Handing off to Asterisk\n"); append_branch(); append_urihf("CC-Diversion: ", "\r\n"); t_relay(); break; } This works great for normal users who are registered but not answering their phone (or indeed are busy it seems - i havent' worked out how to configure this behaviour -- any suggestions?). However, i have two problems: (1) If the call is from Nonnat UA --> SER --> Mediaproxy --> Nat UA | \/ Asterisk VM then the Nonnat UA doesn't get audio from asterisk vm. I think this has somethign to do with mediaproxy trying to proxy the vm request when it shouldnt and not knwoing where to send it - again, any suggestions? (2) The most annoying issue at the moment is for offline users. It seems logical that if you have an offline user (one that has been found in the subscriber db but just isnt' registered at the moment) you might want their address to divert to vm. However, no amount of fiddling has enabled be to use a failure route for this, and i'm not sure why. In desperation i just created a new route block to handle it - the same as above, but called route[4] instead of failure_route. This at least forwards the request correctly. However, all i get is "The p...." and then the media stream cuts out, and * console shows the channel has been hung up. Below is the ngrep. For some reason two INVITE reuqests are sent from ser(.136) to *(.137) and i'm not sure why. Notice also that a CANCEL is generated, which corresponds to the media stream dying almost as soon as it starts, but again i'm not sure why.... If anyone has anyideas... if you need to look at the ser.cfg then let me know. I'm sure there must be a mor elegant way of handling vm but im not sure what it is at the moment! Dave U 161.30.94.151:2378 -> 161.30.94.136:5060 INVITE sip:1000@sip.dev.inmarsat.com SIP/2.0..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..T o: sip:1000@sip.dev.inmarsat.com..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 1 INVITE..Cont act: <sip:test3@161.30.94.151>..Content-Type: application/sdp..Content-Length: 308..Accept-Langua ge: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/UDP 161.30.94.151....v=0..o=Pingtel 5 5 IN IP4 161.30.94.151..s=phone- call..c=IN IP4 161.30.94.151..t=0 0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap:96 eg711u/800 0/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=rtpmap:18 g729/ 8000/1..a=fmtp:18 annexb=no..a=rtpmap:98 telephone-event/8000/1.. # U 161.30.94.136:5060 -> 161.30.94.151:5060 SIP/2.0 407 Proxy Authentication Required..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag=9d060f1cca80def62c99c3e616c718b0.21f2..Call-Id: call-1093359137 -28@161.30.94.151..Cseq: 1 INVITE..Via: SIP/2.0/UDP 161.30.94.151..Proxy-Authenticate: Digest rea lm="sip.dev.inmarsat.com", nonce="412b574f5e5a864751e3e96fb32f3f7a4929a17a"..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 161.30.94.136:5060 "Noisy feedback tells: pid=25819 req_src_ip=161.30.94.151 req_src_port=2378 in_uri=sip:1000@sip.dev.inmarsat.com out_uri=sip:admin@sip.dev.inmarsat.com via_cnt==1".... # U 161.30.94.151:2378 -> 161.30.94.136:5060 ACK sip:1000@sip.dev.inmarsat.com SIP/2.0..Contact: sip:test3@161.30.94.151..From: sip:test3@sip. dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag=9d060f1cca80def62c99c3e616c71 8b0.21f2..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 1 ACK..Accept-Language: en..User-Agent : Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/UDP 161.30.94.151.. Content-Length: 0.... # U 161.30.94.151:2378 -> 161.30.94.136:5060 INVITE sip:1000@sip.dev.inmarsat.com SIP/2.0..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..T o: sip:1000@sip.dev.inmarsat.com..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 2 INVITE..Cont act: <sip:test3@161.30.94.151>..Content-Type: application/sdp..Content-Length: 308..Accept-Langua ge: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, sip-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Proxy-Authorization: DIGEST USERNAME="test3", REALM="sip.dev.inmarsat.com", NONCE=" 412b574f5e5a864751e3e96fb32f3f7a4929a17a", URI="sip:1000@sip.dev.inmarsat.com", RESPONSE="520ee04 3cedf1c3d2ccbc0fa05101d2c"..Via: SIP/2.0/UDP 161.30.94.151....v=0..o=Pingtel 5 5 IN IP4 161.30.94 .151..s=phone-call..c=IN IP4 161.30.94.151..t=0 0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap :96 eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=r tpmap:18 g729/8000/1..a=fmtp:18 annexb=no..a=rtpmap:98 telephone-event/8000/1.. # U 161.30.94.136:5060 -> 161.30.94.151:5060 SIP/2.0 100 trying -- your call is important to us..From: sip:test3@sip.dev.inmarsat.com;tag=2c24 948..To: sip:1000@sip.dev.inmarsat.com..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 2 INVITE ..Via: SIP/2.0/UDP 161.30.94.151..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Lengt h: 0..Warning: 392 161.30.94.136:5060 "Noisy feedback tells: pid=25827 req_src_ip=161.30.94.151 req_src_port=2378 in_uri=sip:1000@sip.dev.inmarsat.com out_uri=sip:1000@asterisk.dev.inmarsat.com :5060 via_cnt==1".... # U 161.30.94.136:5060 -> 161.30.94.137:5060 INVITE sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:1000 @161.30.94.136;ftag=2c24948;lr=on>..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:100 0@sip.dev.inmarsat.com..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 2 INVITE..Contact: <sip: test3(a)161.30.94.151>..Content-Type: application/sdp..Content-Length: 308..Accept-Language: en..Al low: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, si p-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 G MT..Proxy-Authorization: DIGEST USERNAME="test3", REALM="sip.dev.inmarsat.com", NONCE="412b574f5e 5a864751e3e96fb32f3f7a4929a17a", URI="sip:1000@sip.dev.inmarsat.com", RESPONSE="520ee043cedf1c3d2 ccbc0fa05101d2c"..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UDP 161.30.94.151..CC-Diversion: sip:1000@sip.dev.inmarsat.com....v=0..o=Pingtel 5 5 IN IP4 161.30.94 .151..s=phone-call..c=IN IP4 161.30.94.151..t=0 0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap :96 eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=r tpmap:18 g729/8000/1..a=fmtp:18 annexb=no..a=rtpmap:98 telephone-event/8000/1.. # U 161.30.94.136:5060 -> 161.30.94.137:5060 INVITE sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:1000 @161.30.94.136;ftag=2c24948;lr=on>..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:100 0@sip.dev.inmarsat.com..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 2 INVITE..Contact: <sip: test3(a)161.30.94.151>..Content-Type: application/sdp..Content-Length: 308..Accept-Language: en..Al low: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE..Supported: sip-cc, si p-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 G MT..Proxy-Authorization: DIGEST USERNAME="test3", REALM="sip.dev.inmarsat.com", NONCE="412b574f5e 5a864751e3e96fb32f3f7a4929a17a", URI="sip:1000@sip.dev.inmarsat.com", RESPONSE="520ee043cedf1c3d2 ccbc0fa05101d2c"..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.1..Via: SIP/2.0/UDP 161.30.94.151..CC-Diversion: sip:1000@sip.dev.inmarsat.com....v=0..o=Pingtel 5 5 IN IP4 161.30.94 .151..s=phone-call..c=IN IP4 161.30.94.151..t=0 0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap :96 eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=r tpmap:18 g729/8000/1..a=fmtp:18 annexb=no..a=rtpmap:98 telephone-event/8000/1.. # U 161.30.94.137:5060 -> 161.30.94.136:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UD P 161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat. com;tag=as78a37589..Call-ID: call-1093359137-28@161.30.94.151..CSeq: 2 INVITE..User-Agent: Asteri sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Conte nt-Length: 0.... # U 161.30.94.137:5060 -> 161.30.94.136:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.1..Via: SIP/2.0/UD P 161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat. com;tag=as78a37589..Call-ID: call-1093359137-28@161.30.94.151..CSeq: 2 INVITE..User-Agent: Asteri sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Conte nt-Length: 0.... # U 161.30.94.137:5060 -> 161.30.94.136:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UDP 16 1.30.94.151..Record-Route: <sip:1000@161.30.94.136;ftag=2c24948;lr=on>..From: sip:test3@sip.dev.i nmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-ID: call-10933591 37-28@161.30.94.151..CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTION S, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Content-Type: application/sdp..Content-Length: 288....v=0..o=root 2063 2063 IN IP4 161.30.94.137..s=session..c=IN IP4 161.30.94.137..t=0 0..m=au dio 10248 RTP/AVP 18 0 97 3 98..a=rtpmap:18 G729/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:97 iLBC/800 0..a=rtpmap:3 GSM/8000..a=rtpmap:98 telephone-event/8000..a=fmtp:98 0-16..a=silenceSupp:off - - - -.. # U 161.30.94.136:5060 -> 161.30.94.151:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 161.30.94.151..Record-Route: <sip:1000@161.30.94.136;ftag=2c2494 8;lr=on>..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag =as78a37589..Call-ID: call-1093359137-28@161.30.94.151..CSeq: 2 INVITE..User-Agent: Asterisk PBX. .Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Content-Type : application/sdp..Content-Length: 288....v=0..o=root 2063 2063 IN IP4 161.30.94.137..s=session.. c=IN IP4 161.30.94.137..t=0 0..m=audio 10248 RTP/AVP 18 0 97 3 98..a=rtpmap:18 G729/8000..a=rtpma p:0 PCMU/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:3 GSM/8000..a=rtpmap:98 telephone-event/8000..a=fm tp:98 0-16..a=silenceSupp:off - - - -.. # U 161.30.94.136:5060 -> 161.30.94.137:5060 CANCEL sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Via: SIP/2.0/UDP 161.30.94.136;branch=z9h G4bK4773.683f9521.1..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..Call-Id: call-1093359137-2 8@161.30.94.151..To: sip:1000@sip.dev.inmarsat.com..Cseq: 2 CANCEL..User-Agent: Sip EXpress route r(0.8.14 (i386/linux))..Content-Length: 0.... # U 161.30.94.137:5060 -> 161.30.94.136:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.1..From: sip:test3@sip .dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-ID: call-10 93359137-28@161.30.94.151..CSeq: 2 CANCEL..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Content-Length: 0.... # U 161.30.94.151:2394 -> 161.30.94.136:5060 ACK sip:1000@161.30.94.136;ftag=2c24948;lr=on SIP/2.0..Route: <sip:1000@161.30.94.137>..Contact: sip:test3@161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.i nmarsat.com;tag=as78a37589..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 2 ACK..Accept-Langua ge: en..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/U DP 161.30.94.151..Content-Length: 0.... # U 161.30.94.136:5060 -> 161.30.94.137:5060 ACK sip:1000@161.30.94.137 SIP/2.0..Max-Forwards: 10..Contact: sip:test3@161.30.94.151..From: sip :test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-I d: call-1093359137-28@161.30.94.151..Cseq: 2 ACK..Accept-Language: en..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/UDP 161.30.94.136;branch=0..Via: SI P/2.0/UDP 161.30.94.151..Content-Length: 0.... exit

19 years, 10 months

[Serusers] The AA Application suite in Python - how can I get it?

by John Draper

Hi, In the IPTel web site, it mentions an AA Application suite in Python, but I couldn't find a link to it. Has anyone written a Python interface module to "ser" yet? Is this what the AA Application suite is? John

19 years, 10 months

[Serusers] Problem with Asterisk VM + MySQL Database

by TSS Support

We are trying to use the MySQL database for voicemail logins. I can see that * connects correctly to the database and is executing queries, so I'm pretty sure I got the config right. However, it doesn't seem to be picking the password up properly. Found a couple of old mailing list messages about this but no solution. Anyone seen this before and can help me figure out what I'm missing? Here's the info: Relevent debug statements from * vm:/etc/mysql# asterisk -vvvvvvvvvgrc == Parsing '/etc/asterisk/asterisk.conf': Found Asterisk CVS-05/31/04-22:00:51, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster(a)digium.com> ========================================================================= Connected to Asterisk CVS-05/31/04-22:00:51 currently running on vm (pid = 30519) -- Remote UNIX connection -- Executing Wait("SIP/voicemail-6e98", "1") in new stack -- Executing VoiceMailMain("SIP/voicemail-6e98", "") in new stack -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Incorrect password '1234' for user '3609152000' (context = <any>) -- Playing 'vm-incorrect' (language 'en') vm*CLI> A snippet from the startup debug messages for *: vm:~# grep database asterisk.txt -- Logging into database with user asterisk, password (redacted), and database asterisk And from the MySQL log: 040823 13:36:10 3 Connect asterisk@localhost on asterisk 040823 13:36:43 3 Query SELECT password,fullname,email,pager,options FROM users WHERE context='default' AND mailbox='3609152000' And finally, the table in MySQL: mysql> select * from users; +---------+------------+----------+----------+-----------------+-------+---- -----+----------------+ | context | mailbox | password | fullname | email | pager | options | stamp | +---------+------------+----------+----------+-----------------+-------+---- -----+----------------+ | default | 3609152000 | 1234 | | support(a)tss.net | | | 20040713164014 | | default | 3609152050 | 1234 | | support(a)tss.net | | | 20040713164025 | +---------+------------+----------+----------+-----------------+-------+---- -----+----------------+ 2 rows in set (0.00 sec) So what did I miss? It looks like the right data is there, it's selecting it, and successfully getting it back, but not letting us log in. -Keith, TSS

19 years, 10 months

[Serusers] trying to USE SER for a contact centre application

by Neill Wilkinson

19 years, 10 months

[Serusers] Cant' start SER with Radius Accounting support

by jawad bokhari

Hi, I re-compiled ser on a red hat linux machine with radius-accounting support. I added following lines in ser.cfg file: modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "service_type", 16) modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 2) but when I start SER, I get following: 0(9526) set_mod_param_regex: parameter <radius_config> not found in module <acc> 0(9526) parse error (39,81-82): Can't set module parameter 0(9526) set_mod_param_regex: acc matches module acc 0(9526) set_mod_param_regex: parameter <service_type> not found in module <acc> 0(9526) parse error (40,35-36): Can't set module parameter 0(9526) set_mod_param_regex: acc matches module acc 0(9526) set_mod_param_regex: parameter <radius_flag> not found in module <acc> 0(9526) parse error (41,33-34): Can't set module parameter 0(9526) set_mod_param_regex: acc matches module acc 0(9526) set_mod_param_regex: parameter <radius_missed_flag> not found in module <acc> 0(9526) parse error (42,40-41): Can't set module parameter and SER can't start itself. Any help greatly appreciated. Regards, Jawad Bokhari __________________________________ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out! http://promotions.yahoo.com/new_mail

19 years, 10 months

[Serusers] SEMS issue

by Steve Blair

I just updated and rebuilt my SEMS source from cvs. SEMS starts ok but when an inbound calls is relayed to SEMS I get a reorder tone and the following log messages. I didn't see any mention of this in the list archives. Can someone explain what is happening? Thanks,Steve 8396@128.91.50.49:5060>> Aug 23 14:20:15 voicesrvr ser[28966]: [SER_vm]: URI match. Lookup location: Time: [Mon Aug 23 14:20:15 2004] Method: <INVITE> R-uri: <68001@voicesrvr:5090> Call-id:<EFC9B8FE-175011CC-80D0C946-43B09BE9(a)128.91.56.38> Contact Header: <<sip:2155738396@128.91.50.50:5060>> From uri <sip:2155738396@128.91.50.50> To < sip:68001@net.isc.upenn.edu> IP source address <128.91.50.50> Aug 23 14:20:15 voicesrvr ser[28970]: ERROR: fifo_server: command t_reply is not available Aug 23 14:20:15 voicesrvr ser[28970]: ERROR: fifo_server: command must begin with :: ringing Aug 23 14:20:15 voicesrvr ser[28970]: ERROR: fifo_server: command must begin with :: 0000714060F672D8 Aug 23 14:20:15 voicesrvr ser[28970]: ERROR: fifo_server: command must have at least 3 chars Aug 23 14:20:16 voicesrvr ser[14729]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 -- 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

19 years, 10 months

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sr-users August 2004