sr-users June 2006

sr-users@lists.kamailio.org

203 participants
495 discussions

[Serusers] SIP registration with openser

by Shenen Shenen

I've set the SIP_DOMAIN environment variable, (the IP of the pc where I'm running the openser and then the same command but with the IP where I've a sopftphone on a XP pc) and then I've run this: openserctl add 650 1234 650(a)192.16.25.12 (650 is a softphone registered on 192.16.25.10...a XP pc)(I've made also with 650@192.16.25.10..but doesn't change...) Mysql password: (empty I think) MySql password for user 'openser@localhost': (empty) Enter password: (root password ) ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES) introducing the new user to the database failed Where is the problem? Which are the right passwords?....I'm using openser on an asterisk at home box..

18 years, 5 months

[Serusers] About Change in Listening port of SER

by sumit dengri

sir Can we change default listening port (5060) of SER to any other port by changing the port number in ser.cfg file. Or by any other means? regards Send instant messages to your online friends http://in.messenger.yahoo.com Stay connected with your friends even when away from PC. Link: http://in.mobile.yahoo.com/new/messenger/

18 years, 5 months

[Serusers] Database in openser and SIP registration

by Shenen Shenen

Hi!I've run the command to create the database in this way; openser_mysql.sh create I've entered a blank password for root, I entered ser as the domain: I got a creating database ... message, then got another "Enter password:" request, entered blank and then returned to the command line Domain (realm) for the default user 'admin' : ser I've put; If I run in mysql with mysql -p and entered in it,I run connect,I've this; mysql> connect Connection id: 115 Current database: *** NONE *** If I use mysql>connect ser ERROR 1049 (42000): Unknown database 'ser' Where is the problem? I use the following command to register my openser; openserctl add 650 1234 650(a)192.168.251.12 [mailto:650@192.168.251.12<650(a)192.168.251.12> ] /usr/local/sbin/openserctl 1.1 - $Revision: 1.6.2.2 $ parameter usage: * subscribers * add <username> <password> <email> .. add a new subscriber (*) passwd <username> <passwd> ......... change user's password (*) rm <username> ...................... delete a user (*) mail <username> .................... send an email to a user alias show [<alias>] ............... show aliases alias rm <alias> ................... remove an alias alias add <alias> <uri> ............ add an aliases rpid add <username> <rpid> ......... add rpid for a user (*) rpid rm <username> ................. set rpid to NULL for a user (*) rpid show <username> ............... show rpid of a user alias_db show <alias> .............. show alias details ............. etc etc.... Where is the problem?

18 years, 5 months

[Serusers] DB Connectivity Error in SerWeb

by sumit dengri

sir When I try to login in serweb it showed the following error message: DB Error: Connect Failed Bad User Name/Password PLZ tell me the possible reasons for this error message. regards. Send instant messages to your online friends http://in.messenger.yahoo.com Stay connected with your friends even when away from PC. Link: http://in.mobile.yahoo.com/new/messenger/

18 years, 5 months

[Serusers] sip to h323 gateway ...

by Cesc

Hi, I know, a bit off topic ... but i am sure people here have the expertise, so :) I am facing a request to bridge a sip network and an h323 network. I would like to operate the sip with ser as the proxy and some gatekeeper on the h323 side (not required though). The question is ... what should i use to bridge the two networks? - asterisk - yate - vovida -> vocal Any others? comments? Ideally, the bridge would be only signalling-wise (rtp to be direct end-to-end). But, if someone had bad experience with this and would recommend to use a B2BUA approach, please, tell me. I don't know if it makes a difference, but most of the calls would go from the H323 side to the SIP side ... but i don't really want to restrict SIP->H323. Cesc

18 years, 5 months

[Serusers] ser.cfg for VRRP protocol => using t_replicate for 2 proxies

by Shenen Shenen

Hi!I've 2 asterisk at home and I use openser becouse I want use more proxies for my softphones;my problem is this: I'm using vrrp protocol; 192.168.25.1 is the master (asterisk1) 192.168.25.2 is the slave (asterisk2) 192.168.25.200 is the virtual IP. then I have my softphones on a Xp pc with IP 192.168.25.10 I set my softphones with 192.168.25.200 and they run, only that, the problem with asterisk is that softphones and IP phones(some like wi-fi) can register only one proxy, and asterisk can't make a option for more proxies than one;so the proxy is the 192.168.25.200, but at real it is 192.168.25.1..andsoftphones or wi-fi are registered only in the first asterisk with the 192.168.25.1..if the master falls down the slave hasn't the authentication SIP registration...only the master..So I must use openser or ser to have 2 = box my problem is resolved only if I use the redirect option to have 2 = asterisk box so the database can be opened and read from one asterisk to the other;if the master falls down the slave has all the configuration and the SIP registration, so it can work without problems. A) I've set this file on asterisk 192.168.25.1 the master # # $Id: replicate.cfg,v 1.2 2005/07/25 16:27:33 miconda Exp $ # # demo script showing how to set-up usrloc replication # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) # ------------------ module loading ---------------------------------- loadmodule "modules/mysql/mysql.so" loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/usrloc/usrloc.so" loadmodule "modules/registrar/registrar.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" # ----------------- setting module-specific parameters --------------- # digest generation secret; use the same in backup server; # also, make sure that the backup server has sync'ed time modparam("auth", "secret", "alsdkhglaksdhfkloiwr") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); return; }; # 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") { # verify credentials if (!www_authorize("", "subscriber")) { www_challenge("", "0"); return; }; # if ok, update contacts and ... save("location"); # ... if this REGISTER is not a replica from our # peer server, replicate to the peer server if (!src_ip== 192.168.25.2) { t_replicate("192.168.25.2", "5060"); }; return; }; # do whatever else appropriate for your domain log("non-REGISTER\n"); }; } B) I've set this file on asterisk 192.168.25.2 the slave # # $Id: replicate.cfg,v 1.2 2005/07/25 16:27:33 miconda Exp $ # # demo script showing how to set-up usrloc replication # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) # ------------------ module loading ---------------------------------- loadmodule "modules/mysql/mysql.so" loadmodule "modules/sl/sl.so" loadmodule "modules/tm/tm.so" loadmodule "modules/maxfwd/maxfwd.so" loadmodule "modules/usrloc/usrloc.so" loadmodule "modules/registrar/registrar.so" loadmodule "modules/auth/auth.so" loadmodule "modules/auth_db/auth_db.so" # ----------------- setting module-specific parameters --------------- # digest generation secret; use the same in backup server; # also, make sure that the backup server has sync'ed time modparam("auth", "secret", "alsdkhglaksdhfkloiwr") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); return; }; # 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") { # verify credentials if (!www_authorize("", "subscriber")) { www_challenge("", "0"); return; }; # if ok, update contacts and ... save("location"); # ... if this REGISTER is not a replica from our # peer server, replicate to the peer server if (!src_ip== 192.168.25.1) { t_replicate("192.168.25.1", "5060"); }; return; }; # do whatever else appropriate for your domain log("non-REGISTER\n"); }; } Must I call this file ser.cfg?Right?Must I set the on the master listen=192.168.25.1 #on the master port=5060 on the slave listen=192.168.25.2 #on the slave port=5060 Then I use on all the 2 linux; #chkconfig -add ser #chkconfig -add mediaproxy And ser and mediaproxy works like deamon every time I start-up the 2 linux. Is it right to add this 2 sip number in openser or ser in this way?(they're registered in asterisk and 650 is a number from a X-lite, 651 from Phoner) serctl add 650 1234 650(a)192.168.25.200 [mailto: 650(a)192.168.25.200] serctl add 651 1234 651(a)192.168.25.200 [mailto: 651(a)192.168.25.200] Is it right this configuration?If the master falls down the slave has all the Sip registration authentificated and run without any problem? 100000 thanks!

18 years, 5 months

[Serusers] serctl errors;adding new users

by Shenen Shenen

Hi!I've a lot of softphones to one pc with IP 192.1.168.10 (XP pc) and ser in a asterik_at_home box with IP 192.1.168.12, if I make : serctl add 650(a)192.1.168.10 1234 mymail(a)mymail.com Mysql password: Password?is helso or pass0rd ...no one of this,it returns me always this error ERROR 1045 (28000): Access denied for user 'serro'@'localhost' (using password: YES) ERROR 1045 (28000): Access denied for user 'serro'@'localhost' (using password: YES) ERROR 1045 (28000): Access denied for user 'ser'@'localhost' (using password: YES) introducing the new user to the database failed which is the password? With serctl ps I'vent the IP of the XP... I must add it in ser.cfg?And like?If I have other softphones or IP phones on other machine,must I set in ser.cfg the IP of every on pc or IP phone? And how?

18 years, 5 months

[Serusers] alias table

by olivier.taylor

Hi all, Is there a way to reload alias table without restarting Ser? When I add an alias, I need to restart Ser, otherwise the alias will not be found. Via serct? Olivier

18 years, 5 months

[Serusers] SER with nathelper & rtprpoxy - how to disable rtpproxy when two phones are behind the SAME NAT

by Arek Bekiersz

Hi, Not sure if such issue was discussed before. Is this possible to detect that two phones are behind the same NAT, to avoid forcing rtpproxy in SER? This way we let the phones to send media directly to each other in private network behind NAT. Imagine two SIP phones connected to WAN using router with NAT. Both phones are in the same private network. Both phones have accounts on SER proxy, that in turn runs nathelper and rtpproxy. Phone_1 <--> NAT <--> SER,nathelper,rtpproxy Phone_2 <--> Normally when we make call between Phone_1 and Phone_2 using SER, RTP media will flow from Phone_1 to rtpproxy and back to Phone_2. Well, this is basically what we expect from rtpproxy & nathelper :-) Now somehow we could be able to detect in SER that both phones are behind the SAME NAT. When such situation is confirmed, we do not use nathelper and rtpproxy. As a result the phones send RTP media directly between them in private network, not using (and probably saturating) the poor internet link. I was thinking about: 1) Standard detection of NAT-ed situation, and then 2) comparing source IP of INVITE from Phone_1 with Contact IP stored for Phone_2 in "location" table. If they match, both phones MAY be in the same private network and we do not use nathelper functions. Of course it would work only if there is no other NAT before the primary NAT (i.e. we do not have the double NAT situation or similar) -- Regards, Arek Bekiersz

18 years, 5 months

[Serusers] Nedd Help Badly

by Nhadie

Hi All, I'm having problem that the rtpproxy is using 54% of my CPU. Unfortunately i don't have enough money to buy a newer server. Can I make it such a way that I will force only rtpproxy when the call is from a private top another private, but when a private goes to a public IP (like my PSTN gateway) I won't use rtpproxy. another prob of mine is that some phones don't have STUN configured on it, but I have a STUN server. The big clients Ihave pu STUN already, but still they're going to the rtpproxy, I tried this config if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || (uri=~"^(sip:)?00[0-9]*@202.202.202.203")) { if (is_user_in("From", "world-pstn")) { if (isflagset(8)) {xlog("L_INFO", "Call to anywhere in the Worldn");}; if ((is_user_in("From", "onstun"))) { # Strip first to digits (00) strip(2); prefix("25050#"); rewritehostport("202.202.202.202:5060"); consume_credentials(); t_relay_to_udp("202.202.202.202", "5060"); break; } else { if (isflagset(6)) route(1); strip(2); prefix("25050#"); rewritehostport("202.202.202.202:5060"); #rewritehostport("202.202.202.202:5060"); consume_credentials(); t_relay_to_udp("202.202.202.202", "5060"); #t_relay_to_udp("202.202.202.202", "5060"); break; }; } else { if (isflagset(8)) {xlog("L_INFO", "Your not allowed to call PSTNn");}; sl_send_reply("404", "DENIED: USER NOT IN WORLD-PSTN"); break; }; which would detect if an extension is a member of the onstun it will not use rtpproxy, but the caller (using IP phone) cannot hear the called party but the call is connecting, hope anyone can help me, badly need it. Thank You Regards, Nhadie ________________________________________________ Message sent using UebiMiau 2.7

18 years, 5 months

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sr-users June 2006