sr-users September 2005

sr-users@lists.kamailio.org

231 participants
508 discussions

[Serusers] accounting and radius/sql
by Bjorn Asmul 25 Sep '05

25 Sep '05

The best solution is to use two different tables for start and stop accounting. Have a look in the sql.conf file in FreeRADIUS Problem is that SER replaces the Username in the stop-message, with whoever hung up... Therefore the default queries in FreeRADIUS won't find the corresponding record. To get the CDR just join the two tables on AcctSessionId. I have experienced problems joining on the AcctSessionId in MySQL versions prior to 4.1, for some reason. Bjorn

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[Serusers] More random avpops questions....
by sip 25 Sep '05

25 Sep '05

Okay... I think with Steve's help, I've been getting somewhere with my avpops stuff, but I have a couple of questions. 1) What is the format of the $from and $to headers as far as avpops is concerned? How can I find that out? 2) How does avp_print() work (i.e. to where does it print) ? Is there any way I can get intermediate values of things so I can see just what it is my little scripts are doing? N.

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[Serusers] [about SER]networking among three SIP servers.....
by 송 미진 25 Sep '05

25 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear........ I want to networking among three SIP servers. But now I am trying using two servers likes three servers. That is, - source : computer A ip address ; 192.168.178.194 sipuri : sip:mj@192.168.178.194 - via : computer B ip address : 192.168.178.193 sipuri : sip:hj@192.168.178.193 - destination : computer A ip address :192.168.178.194 sipuri : sip:mj@192.168.178.194 And I am using sipp to test (I modified the "via" part based on original uac module). ===================================================== << computer B's test.cfg >> # # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias="kims.myserver.sampro.cse" alias="myserver.sampro.cse" alias="192.168.178.193" # debugging log_stderror=yes # (cmd line: -E) mhomed=yes 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/exec.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "\n\n"); log(1, "START!!!\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>=204822) { 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 # if (method=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # }; log(1, "record-route processing completed\n"); # 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(uri=="sip:mijin@sam.cse"){ # forward(192.168.178.194, 5060); # }; if (uri=~"sip:192.168.178.193"){ if (method=="REGISTER") { log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save ì„±ê³µ\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3ìœ¼ë¡œ ì�´ë�™ \n"); route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DBì—�ì„œ ì°¾ ì�„ ìˆ˜ ì—†ì�Œ \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:hjkim@myserver.sampro.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.193" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); } log(1, "-------route3 ë�� \n"); } As you see, I added modparam("rr", "enable_full_lr", no) if(uri=="sip:mijin@192.168.178.194"){ forward(192.168.178.194, 5060); append_branch("sip:hjkim@192.168.178.193:5060"); log(1, "next hop...\n"); break; }; but, I received error message in computer B(via): 8(12253) ERROR: forward_reply: no 2nd via found in reply What does it mean? What can I do for doing what I want ??? Is there any problem in my test.cfg file ????? I would like to know in detail . please tell me what to do............................... sipp module can't support that???? And Please tell me about forward funtion. I don't know whether I understand forward function usage well....... sorry ............ Song _________________________________________________________________ 고.. 감.. 도.. 사.. 랑.. 만.. 들.. 기.. MSN 러브 http://www.msn.co.kr/love/

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[Serusers] Radius Authorization and AVP Retrieval
by Tavis P 24 Sep '05

24 Sep '05

I'm having some trouble trying to integrate both Radius authentication and Radius AVP storage. I'm using freeradius and there doesn't seem to be much documentation regarding configuration semantics and such. Could anyone share some information on how they were able to use a Radius server to authenticate requests and (with avp_radius) as an AVP retrieval mechanism? I'm uncertain as to how I can stack these two uses of data on the freeradius server tavis

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[Serusers] [about SER]networking among three SIP servers.....
by 송 미진 24 Sep '05

24 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear........ I want to networking among three SIP servers. But now I am trying using two servers likes three servers. That is, - source : computer A ip address ; 192.168.178.194 sipuri : sip:mj@192.168.178.194 - via : computer B ip address : 192.168.178.193 sipuri : sip:hj@192.168.178.193 - destination : computer A ip address :192.168.178.194 sipuri : sip:mj@192.168.178.194 And I am using sipp to test (I modified the "via" part based on original uac module). ===================================================== << computer B's test.cfg >> # # $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $ # # welcome message for new subscribers; based on exec # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) # must be yes since REGISTER processing causes an INVITE to be sent, # which needs to be processed by another process fork=yes children=4 alias="kims.myserver.sampro.cse" alias="myserver.sampro.cse" alias="192.168.178.193" # debugging log_stderror=yes # (cmd line: -E) mhomed=yes 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/exec.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" ) modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ log(1, "\n\n"); log(1, "START!!!\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>=204822) { 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 # if (method=="INVITE") record_route(); # 1=loose routing if(method=="INVITE"){ log(1, "receive INVITE\n"); record_route(); sl_send_reply("200", "ok"); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # if(method=="ACK"){ # log(1, "receive ACK\n"); # }; log(1, "record-route processing completed\n"); # 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(uri=="sip:mijin@sam.cse"){ # forward(192.168.178.194, 5060); # }; if (uri=~"sip:192.168.178.193"){ if (method=="REGISTER") { log(1, "receive REGISTER\n"); # save location before initiating welcome save("location"); log(1, "save ì„±ê³µ\n"); record_route(); sl_send_reply("200", "ok"); # welcome message log(1, "route3ìœ¼ë¡œ ì´ë™ \n"); route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1, "Usrloc DBì—ì„œ ì°¾ ì„ ìˆ˜ ì—†ìŒ \n"); break; }; }; if(method=="BYE"){ log(1, "receive BYE\n"); record_route(); sl_send_reply("200", "ok"); break; }; t_relay(); } # welcome message -- if a REGISTER succeeded, look if it is the first-time; # if so, initiate click-to-dial transaction to set up call to an announcement # server; edit the config values first to make it work route[3] { if (!exec_msg(' # config: # --announcement server URI ANS="sip:hjkim@myserver.sampro.cse" # --SIP domain # DOMAIN="192.168.178.193" DOMAIN="192.168.178.193" # ctd CTD=${HOME}/sip_router/examples/ctd.sh # ------------------------------------ # check if first time ... SIP_UID=`echo $SIP_HF_TO | sed -e "s/^.*sip:$[a-zA-Z0-9_\.]*$@.*$/\1/g"` QUERY="select flag from subscriber where username=\"$SIP_UID\"; update subscriber set flag=\"x\" where username=\"$SIP_UID\" "; mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null # ... if so, c-t-d to announcement server if [ "$?" -ne 0 ] ; then # flag was not set to x yet -- first-time registration; # initiate a call from telephone of the user to an announcement server $CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1 fi ')) { log(1, "welcome exec failed\n"); } log(1, "-------route3 ë \n"); } As you see, I added modparam("rr", "enable_full_lr", no) if(uri=="sip:mijin@192.168.178.194"){ forward(192.168.178.194, 5060); append_branch("sip:hjkim@192.168.178.193:5060"); log(1, "next hop...\n"); break; }; but, I received error message in computer B(via): 8(12253) ERROR: forward_reply: no 2nd via found in reply What does it mean? What can I do for doing what I want ??? Is there any problem in my test.cfg file ????? I would like to know in detail . please tell me what to do............................... Song _________________________________________________________________ 싸게 싸게 MSN공동구매 http://www.waawaa.com/cobuy/cobuy_default.asp?siteid=10160

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[Serusers] about SER
by 송 미진 24 Sep '05

24 Sep '05

confirm 83b7fb1612811c64f85f47a6046dd84dea286d15 Dear. All Hi~ I was adviced many times by "seruser" but my problems has not been solved. I,m afraid that I have many questions. So I intend to specify my present situation. Operating System: Linux Fedora 3 ================================================================================ [root@cse3 ~]# serctl start Starting SER : started pid(8370) [root@cse3 ~]# serctl ps 0 8370 attendant 1 8375 fifo server 2 8377 receiver child=0 sock= 127.0.0.1:5060 3 8379 receiver child=1 sock= 127.0.0.1:5060 4 8384 receiver child=2 sock= 127.0.0.1:5060 5 8386 receiver child=3 sock= 127.0.0.1:5060 6 8387 receiver child=0 sock= 192.168.178.193:5060 7 8390 receiver child=1 sock= 192.168.178.193:5060 8 8393 receiver child=2 sock= 192.168.178.193:5060 9 8397 receiver child=3 sock= 192.168.178.193:5060 10 8401 timer 11 8403 tcp receiver 12 8406 tcp receiver 13 8410 tcp receiver 14 8412 tcp receiver 15 8415 tcp main process ============================================================================== my_ip: 192.168.178.193 sip_uri: sip:hjkim@192.168.178.193 I added my hostname and my ip address in /etc/host file. And I added alias="hostname.myserver.sampro.cse" alias="192.168.178.193" in /usr/local/etc/ser/ser.cfg file. ==================================== 1 ============================================ [root@cse3 ~]# serctl ping sip:hjkim@192.168.178.193 [1]+ Done cat <$path | filter_fl 483 =========================================================================================== ==================================== 2 =========================================== I received "SIP/2.0 483 Too Many Hops" . but I don't know the reason. Well...My computer is connected other computers including computer has 192.168.178.194 as ip address. And my /etc/host file has 127.0.0.1 cse4 localhost.localdomain localhost 192.168.178.194 kims myserver.sampro.cse Is there reason in here??? I don't know . =========================================================================================== [root@cse3 ~]# sipsak -T -s sip:hjkim@192.168.178.193 warning: IP extract from warning activated to be more informational 0: 192.168.178.193 (0.355 ms) SIP/2.0 483 Too Many Hops 1: 192.168.178.193 (0.982 ms) SIP/2.0 483 Too Many Hops 2: 192.168.178.193 (0.459 ms) SIP/2.0 483 Too Many Hops 3: 192.168.178.193 (0.520 ms) SIP/2.0 483 Too Many Hops .................... 177: timeout after 500 ms 177: timeout after 1000 ms 177: timeout after 2000 ms 177: 192.168.178.193 (3306.067 ms) SIP/2.0 483 Too Many Hops 178: 192.168.178.193 (3.073 ms) SIP/2.0 483 Too Many Hops 179: 192.168.178.193 (3.104 ms) SIP/2.0 483 Too Many Hops .................. 313: 192.168.178.193 (3.345 ms) SIP/2.0 483 Too Many Hops 314: 192.168.178.193 (3.494 ms) SIP/2.0 483 Too Many Hops 315: 192.168.178.193 (3.038 ms) SIP/2.0 483 Too Many Hops 316: 192.168.178.193 (3.518 ms) SIP/2.0 483 Too Many Hops 317: 192.168.178.193 (5.048 ms) SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) without Contact header ============================================================================================ ======================================== 3 ====================================== In response part, I found the message "SIP/2.0 401 Unauthorized" and "authorization failed request already contains (Proxy-) Authorization, but received 40[1|7], see above" What happened?? ============================================================================================ [root@cse3 ~]# sipsak -U -vv -s sip:hjkim@192.168.178.193 warning: ignoring -i option when in usrloc mode registering user hjkim... authorizing registering user hjkim... request: REGISTER sip:192.168.178.193 SIP/2.0 Authorization: Digest username="hjkim", uri="sip:192.168.178.193", algorithm=MD5, realm="myserver.sampro.cse", nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87", response="f58e11aff01e5123bcf5010405e91278" Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport From: sip:hjkim@192.168.178.193;tag=4de59540 To: sip:hjkim@192.168.178.193 Call-ID: 1306891584(a)127.0.0.1 CSeq: 2 REGISTER Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.1 Expires: 15 Contact: sip:hjkim@127.0.0.1:32860 response: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport=32860;received=192.168.178.193 From: sip:hjkim@192.168.178.193;tag=4de59540 To: sip:hjkim@192.168.178.193;tag=b27e1a1d33761e85846fc98f5f3a7e58.ad3a Call-ID: 1306891584(a)127.0.0.1 CSeq: 2 REGISTER WWW-Authenticate: Digest realm="myserver.sampro.cse", nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87" Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 192.168.178.193:5060 "Noisy feedback tells: pid=8397 req_src_ip=192.168.178.193 req_src_port=32860 in_uri=sip:192.168.178.193 out_uri=sip:192.168.178.193 via_cnt==1" error: authorization failed request already contains (Proxy-) Authorization, but received 40[1|7], see above ============================================================================================ [root@cse3 ~]# serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0xf50b44c8 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0xf50b2218 d_ll { n : 2 first: 0xf50b4220 last : 0xf50b4348 } ...Record(0xf50b4220)... domain: 'location' aor : 'kim' ~~~Contact(0xf50b4260)~~~ domain : 'location' aor : 'kim' Contact : 'sip:hjkim@myserver.sampro.cse' Expires : Permanent q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50b4348)... domain: 'location' aor : 'mj' ~~~Contact(0xf50b4388)~~~ domain : 'location' aor : 'mj' Contact : 'sip:mj@myserver.sampro.cse' Expires : Permanent q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ===/Domain list=== ==================================================================================================================================== 4 ===================================== I received the message "received ICMP packet (type: 3, code: 10): from 192.168.178.194" What does means this message?? ============================================================================================ [root@cse3 ~]# sipsak -T -vv -s sip:iptel@192.168.178.194 warning: IP extract from warning activated to be more informational Max-Forwards set to 0 received ICMP packet (type: 3, code: 10): from 192.168.178.194 ======================================================= ======================================================= << my ser.cfg >> # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 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) alias="192.168.178.193" alias="myserver.sampro.cse" alias="kims.myserver.sampro.cse" /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- 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 >= 2000048 ) { 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 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=="sip:192.168.178.193") { # 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=="sip:192.168.178.193") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("myserver.sampro.cse", "subscriber")) { www_challenge("myserver.sampro.cse", "0"); break; }; save("location"); log(1, "save \n"); break; }; lookup("aliases"); if (!uri=="sip:192.168.178.193") { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # 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] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ###################################################################################### I’d appreciate it if you would give me some solutions. Thank you in advance..... mi-jin Song _________________________________________________________________ 행운의 주인공이 이번엔 나일꺼야, 진짜루... 인터넷 복권 http://www.msn.co.kr/money/interlotto/

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[Serusers] cannot receive
by rajesh volluru 24 Sep '05

24 Sep '05

hi I am using Sip communicator. I cant receive any voice from the other calling party but the calling party can hear whatever I talk. The ReceiveStreamEvent is not getting generated. Would anyone help me reg this please? Thanks Rajesh --------------------------------- Yahoo! India Matrimony: Find your partner now.

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RE: [Serusers] Permanent aliases
by Igor Balk 23 Sep '05

23 Sep '05

Thank you guys for the clarification. If you do not mind I would ask one more question. Let say I have aliases 1234 of the account user@server and I'm getting call to the 1234 alias but there is no one registered at the moment with URI name@server. I also have SER configured in a way that if call will come to name@server URI and no user is availeble the call will be forwarded to voice mail box. Will same scenario work in case of call to the alias if I do lookup("aliases") before forwarding or I should do something more sofisticated to make it work? Thank you in advance, Igor -----Original Message----- From: Atle Samuelsen [mailto:clona@cyberhouse.no] Sent: Friday, September 23, 2005 1:14 PM To: Igor Balk Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Permanent aliases Hi Igor, I think your wrong there. AS far as you register with your box to your sipserver, and have a alias pointing to the username@yoursipserver then you should'nt have a problem. -Atle * Igor Balk <IBalk(a)intermedia.net> [050923 19:01]: > > Sorry if I'm asking stupid question but I'm trying to set up alias for > example 1234<->john.doe and make it permanent. As far as I understand > if I use serctl add alias command and then contact for the person will > change (dynamic ip, login from other coputer etc) my alias would not > work anymore. Please tell me if I'm worng and if there is a way to fix > this problem. > > Thank you, > Igor > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Question regarding UDP TCP translation
by Weiler, Frank 23 Sep '05

23 Sep '05

Hi all, I'm new to SER and I started to play around with it as we need a solution to connect UPD UA's connected to Asterisk to a Microsoft LCS and we tough to use SER for this purpose. I've read so far through the Admin guide and it seems that the TCP support is natively embedded into SER. However I've added to the ser.cfg file the following lines (within the route[0] block just before branching to route [1]): #Forward all traffic with the uri*(a)172.16.0.201 <mailto:uri*@172.16.0.201> If (uri=~"sip:*@172.16.0.201) { forward_tcp(172.16.0.201,5060); Break; }; However while sniffing the packets it still appears that the INVITE requests are sent to the LCS using UDP and I receive an ICMP port unreachable. The original request for the UA is sent in UDP. I assume that I configured something wrong and I've checked the mailing lists and I didn't find any related documents. I'm using the latest version of SER available. Needless to say that UDP SIP calls between ASTERISK and SER work fine whatever clients I use. Any clue or help would be highly appreciated. Thank you very much in forecast for your help. Cheers, Frank

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[Serusers] Getting starting with ser...
by IS2 - Florent BARBIER 23 Sep '05

23 Sep '05

Hello all, Not really easy ton understand how to setup... If somebody can help, I'll really appreciate. My config : redhat 7.3 (SME E/Smith edition) Asterisk 1.0.9 ser serweb IP phone 1 IP phone 2 IP public 1 IP public 2 ! ! ! Internet ! --------------------------------------------- ! ! ! IP public 3 ! ! Lan Asterisk ! ------------------------------------------------- ! ! IP Phone 3 IP Phone 4 IP Local 3 IP Local 4 Every IP Phone are registered in the AsterisK IP Phone 3 comunique with IP Phone 4 whithout PB IP Phone 1 comunique with IP Phone 3 or 4 whithout PB IP Phone 1 comunique with IP Phone 2 whith delay ! So audio stream doesn't go directly from IP Public 1 to IP Public 2. As I understand S.E.R. should help me to redirect audio stream... Well I setup S.E.R. and SERWEB under the Asterisk box (wich is the gateway & the firewall) I add some user whith serweb I add register in my sip.conf to register my asterisk to ser I try to do an echo test and nothing append. Well, how to setup SER to build this config... I'm sure this question certainly basic for most of you, but I've good Knowledge of AsterisK and I start with SIP proxies. I'm really not find F.A.Q. or userguide to build this... Thank you for your help, Florent.

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sr-users September 2005