Hello Ozan,
looking on the the config script you sent on the list, you just have to move the setflag(1) from the "Record Route Section and Acc section" to route[3]:
if (uri=~"^sip:0[0-9]*@*") { setflag(1); # mark for acc rewritehost("212.XXX.59.XXX"); route(1); break; };
regards, bogdan
Ozan Blotter wrote:
Dear Bogdan,
Thanks for your information about this "setflag" issue but still i have no idea which line i have to remove and move to where ?. For BYE problems i think i can talk with Carrier to setup that for me, they're using MERA VoIP.
Thanks Again, Ozan
ps * : today or tomorrow i will let you to login/test the billing thing since it's still a big mess with codes around..
----- Original Message ----- From: "Bogdan-Andrei Iancu" bogdan@voice-system.ro To: "Ozan Blotter" cosmocid@ispro.net.tr Cc: users@openser.org Sent: Tuesday, July 12, 2005 12:02 PM Subject: Re: [Users] Broken Calls now written to Database
Hi Ozan,
first, if you want not to account SIP2SIP calls, you have just not to set the acc flag for those calls (flag 1 in your script). In other words, if you want to account only calls going to PSTN, set the flag 1 only if destination is PSTN.
Regarding the BYE - this is an well known issue and you can solve it on the GW side (depending what GW you have) - you may enable on your GW two types of timers:
- for received media: if the GW received no more media in X secs,
it will generate the BYE.
- Session Timer: periodically, the GW probes if the UAC is still
on by sending reINVITEs - in case of no answer or negative reply, a BYE is generated. Take a look at your GW specification if it has support for this.
Since the BYEs generated in this case by GW will end by timeout (the client being disconnected), note that you need to enable "failed_transaction" acc param (set it to on) in 0.9.x or set a flag for "failed_transaction_flag" acc param in 0.10.x (see the online doc for more info).
For PostPaid - it's welcome - when you have a working version, please let us know and we can arrange with the upload.
regards, bogdan
Ozan Blotter wrote:
Dear List, I'm making a call to PSTN, it's okay i can talk with the other party via ATA, but suddenly i unplug power from it, and in accounting module it does not write line with BYE message so i cannot understand whether the call is finished or not. only INVITE and ACK are written, no BYE :( This may be a security hole for customers, because they talk for a long time and they may unplug their units or cut power so it does not tell SER it's over. How i can prevent this, also what i need to add as a line for telling ACC module not to write calls from 833 to 833 into database, which are free SIP2SIP calls ? Btw, a Postpaid Billing System is on the way for OpenSER built in PHP, later i will need OpenSER group's help to place it onto hosting site. Shortly it will have features:
- Nothing extra from package, default MySql Database which comes
with OpenSER Release is being used,
- All routing is done via OpenSER's ser.cfg ,
- Initial Rate, Initial Time, Increment Rate, Increment Time will be
user variable ,
- Account Creation/Deletion done in PHP ,
- Invoice Generation and CDR will include many variables ,
- Rating will be in PHP,
- And everything is totaly free :)
If you have ideas and/or suggestion please write back to me so i can work for it too. Reqs: OpenSER & PHP & MySql & Apache Thanks, Ozan
Hi Bogdan,
Instead i found another method to tell SER do not write to database if URI is 833, but i think your method is more effective so will try that one too :)
Thanks, Ozan
this is the changed part from my latest ser.cfg >>>
# ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.XXX.104.XXX:5060;nat=yes"); } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { record_route(); setflag(1); } };
# -----------------------------------------------------------------
----- Original Message ----- From: "Bogdan-Andrei Iancu" bogdan@voice-system.ro To: "Ozan Blotter" cosmocid@ispro.net.tr Cc: "users openser.org" users@openser.org Sent: Wednesday, July 13, 2005 12:01 PM Subject: Re: [Users] Broken Calls now written to Database
Hello Ozan,
looking on the the config script you sent on the list, you just have to move the setflag(1) from the "Record Route Section and Acc section" to route[3]:
if (uri=~"^sip:0[0-9]*@*") { setflag(1); # mark for acc rewritehost("212.XXX.59.XXX"); route(1); break; };regards, bogdan
Ozan Blotter wrote:
Dear Bogdan,
Thanks for your information about this "setflag" issue but still i have no idea which line i have to remove and move to where ?. For BYE problems i think i can talk with Carrier to setup that for me, they're using MERA VoIP.
Thanks Again, Ozan
ps * : today or tomorrow i will let you to login/test the billing thing since it's still a big mess with codes around..
----- Original Message ----- From: "Bogdan-Andrei Iancu" bogdan@voice-system.ro To: "Ozan Blotter" cosmocid@ispro.net.tr Cc: users@openser.org Sent: Tuesday, July 12, 2005 12:02 PM Subject: Re: [Users] Broken Calls now written to Database
Hi Ozan,
first, if you want not to account SIP2SIP calls, you have just not to set the acc flag for those calls (flag 1 in your script). In other words, if you want to account only calls going to PSTN, set the flag 1 only if destination is PSTN.
Regarding the BYE - this is an well known issue and you can solve it on the GW side (depending what GW you have) - you may enable on your GW two types of timers:
- for received media: if the GW received no more media in X secs, it
will generate the BYE.
- Session Timer: periodically, the GW probes if the UAC is still on
by sending reINVITEs - in case of no answer or negative reply, a BYE is generated. Take a look at your GW specification if it has support for this.
Since the BYEs generated in this case by GW will end by timeout (the client being disconnected), note that you need to enable "failed_transaction" acc param (set it to on) in 0.9.x or set a flag for "failed_transaction_flag" acc param in 0.10.x (see the online doc for more info).
For PostPaid - it's welcome - when you have a working version, please let us know and we can arrange with the upload.
regards, bogdan
Ozan Blotter wrote:
Dear List, I'm making a call to PSTN, it's okay i can talk with the other party via ATA, but suddenly i unplug power from it, and in accounting module it does not write line with BYE message so i cannot understand whether the call is finished or not. only INVITE and ACK are written, no BYE :( This may be a security hole for customers, because they talk for a long time and they may unplug their units or cut power so it does not tell SER it's over. How i can prevent this, also what i need to add as a line for telling ACC module not to write calls from 833 to 833 into database, which are free SIP2SIP calls ? Btw, a Postpaid Billing System is on the way for OpenSER built in PHP, later i will need OpenSER group's help to place it onto hosting site. Shortly it will have features:
- Nothing extra from package, default MySql Database which comes with
OpenSER Release is being used,
- All routing is done via OpenSER's ser.cfg ,
- Initial Rate, Initial Time, Increment Rate, Increment Time will be
user variable ,
- Account Creation/Deletion done in PHP ,
- Invoice Generation and CDR will include many variables ,
- Rating will be in PHP,
- And everything is totaly free :)
If you have ideas and/or suggestion please write back to me so i can work for it too. Reqs: OpenSER & PHP & MySql & Apache Thanks, Ozan
-
Bogdan-Andrei Iancu -
Ozan Blotter