Hi Bodgan, Thanks for your reply. I had no idea about ACLs , i saw a document that ACLS are nothing but group checking but i'm not clear abt that.so i modified and tried the code and it's working for DND. The procedure i followed is when i called to internal number the call goes to route[10] as follows: In opensips.cfg: if (is_uri_host_local()) { # -- Inbound to Inbound route(10);
route[10] { if (!does_uri_exist()) { if (uri=~"^sip:[2-9][0-9]{9}@") { if (is_user_in("credentials","local")) { route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for local calls"); exit; }; }; if (uri=~"^sip:1[2-9][0-9]{9}@") { if (is_user_in("credentials","ld")) { route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for long distance"); exit; }; }; if (uri=~"^sip:011[0-9]*@") { if (is_user_in("credentials","int")) { route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for international calls"); }; }; }; if (uri=~"^\*[1-9]+") { # we do provide access to media services only to our # subscribers, who were previously authenticated xlog("****Starting of $ru***\n"); if (!is_from_local()) { send_reply("403","Forbidden access to media service"); exit; } #identify the services and translate to Asterisk extensions if ($rU=="*78") { #ENABLE DND seturi("sip:an_fw...@asterisk IP:5060"); } else if ($rU=="*79") { # DISABLE DND seturi("sip:an_fw...@asterisk IP:5060"); } t_relay(); exit; } if (!lookup("location")) { if (does_uri_exist()) { ## User not registered at this time. revert_uri(); prefix("u"); rewritehostport("localhost"); #Use the IP address of your voicemail server route(6); route(1); } else { sl_send_reply("404", "Not Found-10-1"); exit; } sl_send_reply("404", "Not Found-10-2"); exit; }; route(6); route(1); } Asterisk Server: In the Asterisk(extensions.conf) i wrote a dialplan for DND: extensions.conf: exten => AN_fwdok,1,Answer exten => AN_fwdok,2,Set(DB(SIP/DND/${CALLERID(num)})=1) exten => AN_fwdok,3,Playback(beep) exten => AN_fwdok,4,Wait(2) exten => AN_fwdok,5,Hangup exten => AN_fwdko,1,Answer exten => AN_fwdko,2,NoOp(${DB_DELETE(SIP/DND/${CALLERID(num)})}) exten => AN_fwdko,3,Playback(beep) exten => AN_fwdko,4,Wait(2) exten => AN_fwdko,5,Hangup. The above code i posted is working for me Is the above procedure i followed for activation of DND in opensips is correct? Thanks in Advance! Bogdan-Andrei Iancu wrote: > > Hi Indiver, > > your code is not implementing DND, but simply redirects *78 to a voice > announcement. What you have to do is to actually implement the DND > service first (maybe using an ACL -> if a user receives a call and the > user had the DND ACL on, reject the call) and then to control it > (enable/disable) via the *78 code. > > Regards, > Bogdan > > Indiver wrote: >> Hi Bodgan, >> >> For Virtual service activation codes, as per my observation when ever >> user >> dials *78 u just redirecting to asterisk as follows: >> if ($rU=~"^\*[1-9]+") { >> # we do provide access to media services only to our >> # subscribers, who were previously authenticated >> if (!is_from_local()) { >> send_reply("403","Forbidden access to media service"); >> exit; >> } >> #identify the services and translate to Asterisk extensions >> if ($rU=="*78") { >> # access to own voicemail IVR >> seturi("sip:an_fw...@asterisk_ip:ASTERISK_PORT"); >> t_relay(); >> exit; >> and when the user dialing *79 it was just rewriting the uri >> as >> follows >> >> seturi("sip:an_fw...@asterisk_ip:ASTERISK_PORT"); >> t_relay(); >> exit; >> >> Is my assumption is right?Thanks in advance >> >> >> >> >> Bogdan-Andrei Iancu wrote: > >> >>> Hi Indiver, >>> >>> the Activation Codes service is pure scripting - it is not a module in >>> opensips. Most of the VoIP features you can do with OpenSIPS are not >>> necessarily provided by module (directly by C code), but rather via >>> scripting logic (combination of different ops in the opensips script). >>> >>> Regards, >>> Bogdan >>> >>> Indiver wrote: >>> >>>> Hi Every One, >>>> >>>> I registered in opensips voip services site. I found a tab regarding >>>> dialing >>>> plan. such as *78 for enable dnd,*72 for setting to permanent >>>> redirect,*50 >>>> for voicemail inbox. I found no documentation in opensips regarding >>>> these >>>> services. Is there a way for acheiving this thru opensips. Thanks in >>>> advance. >>>> >>>> >>> -- >>> Bogdan-Andrei Iancu >>> www.voice-system.ro >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> > > > -- > Bogdan-Andrei Iancu > www.voice-system.ro > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- View this message in context: http://n2.nabble.com/Virtual-Service-Activation-Codes-tp4063251p4139290.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users