Hi,
Anyone know of a pre-paid billing system for SER? I'm willing to pay $80 for a module of some sort?
Barry
Hi
This has been asked b4, SER isnt realyy the best at doing prepaid, simply because it is not designed to work that way, for prepaid you really need to be able to sit in the middle of the audio stream, so u can disconnect it when the money runs out, hence the best way to do it is to use a b2bua, asterisk os one solution, there is one called sippy (not free), and vovida has one also, or you can try to put one together urself, although its not a b2bua, u could look at using session-timers as a few on this list have, it depends on where u want to take the solution.
If u want a simple install use asterisk, run it with one of its prepaid solutions and ur done.
Do you want different routes, lcr, different prices pre hour of day, i.e not just cdr, but a rating engine also, this then requires more work.
I would say go with asterisk....although I used this and felt its a bit heavy, but then my needs were scalability, and i didnt want to deal with non-nat traffic.
Iqbal
On 4/3/2005, "Barry Murphy" barry@unix.co.nz wrote:
Hi,
Anyone know of a pre-paid billing system for SER? I'm willing to pay $80 for a module of some sort?
Barry
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
for prepaid you really need to be able to sit in the middle of the audio stream, so u can disconnect it when the money runs out
U need to sit in the middle of the audio stream or the sip stream? Whats the point of having a separate audio and control stream if u can't do things like this(like disconnecting). I'm a sip newbie so please someone, enlighten me.
you can have a separate stream, but something needs to tell it to drop, hence if you are not in the middle of it, how do you know where to send the disconnect, also how will you monitor the usage in real time, in SER you can record the INVITE and BYE (amongst other things) hence you know the start-stop record, which means you can do a post-paid.
Iqbal
Shidan wrote:
for prepaid you really need to be able to sit in the middle of the audio stream, so u can disconnect it when the money runs out
U need to sit in the middle of the audio stream or the sip stream? Whats the point of having a separate audio and control stream if u can't do things like this(like disconnecting). I'm a sip newbie so please someone, enlighten me.
.
You can have that setup in the DB but in order to kill the media session you have to be in the middle to do that since SER is not monitor the media stream.
On Apr 4, 2005 5:44 PM, Shidan shidan@gmail.com wrote:
if you are not in the middle of it, how do you know where to send the disconnect, also how will you monitor the usage in real time
Sorry, this is probably a silly question but for a prepaid solution don't you just need to set a time limit for when to send a disconnect ?
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
yes, but no counters running in SER for one thing, even if u could start one, where u gonna send the disconnect,
UA1 --- SER --- UA2 for INVITE
then they talk direct UA1---UA2 so ser knows after 5 mins to disconnect, but where to, and how will it know they have stopped talking
Iqbal
On 4/4/2005, "Shidan" shidan@gmail.com wrote:
if you are not in the middle of it, how do you know where to send the disconnect, also how will you monitor the usage in real time
Sorry, this is probably a silly question but for a prepaid solution don't you just need to set a time limit for when to send a disconnect ?
yes, but no counters running in SER for one thing, even if u could start one, where u gonna send the disconnect,
UA1 --- SER --- UA2 for INVITE
then they talk direct UA1---UA2 so ser knows after 5 mins to disconnect, but where to, and how will it know they have stopped talking
Thanks, i guess i should change setting in asterisk to canreinvite=no for my prepaid app.! by the way does anyone know a good description of sip with state and sequence diagrams without having to read the rfc or a book?
http://www.iptel.org/sip/ is always a good place to start! This is also a classic tutorial: http://www.cs.tufts.edu/comp/150MMC/SIPTutorial_SinnreichJohnston.pdf And here are many resources: http://www.cs.columbia.edu/sip/papers.html g-)
Shidan wrote:
yes, but no counters running in SER for one thing, even if u could start one, where u gonna send the disconnect,
UA1 --- SER --- UA2 for INVITE
then they talk direct UA1---UA2 so ser knows after 5 mins to disconnect, but where to, and how will it know they have stopped talking
Thanks, i guess i should change setting in asterisk to canreinvite=no for my prepaid app.! by the way does anyone know a good description of sip with state and sequence diagrams without having to read the rfc or a book?
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
Aster -
Barry Murphy -
Greger V. Teigre -
Iqbal -
Shidan